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Subject: [dispatch] draft-mdolly-dispatch-oma-push
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The area directors for RAI asked me to review 
draft-mdolly-dispatch-oma-push, specifically as they relate to the 
ongoing revisions to RFCs 3265 and 3427.

While there are some relatively minor issues with some of the 3265 
concepts used in this document, there is an overarching structural issue 
that needs to be addressed.

By my reading, what this document seeks to do is open a door for OMA to 
create new event packages without requiring any IETF review (and, even 
if such is not the intent, it would certainly be the outcome). This is 
achieved by using an event package type of "oma-push," and then using a 
new "Event" header field parameter of "event-app-id" to indicate the 
actual event package. In the language of object-oriented programming, 
this draft seeks to define an event package that is little more than a 
façade to wrap RFC 3265, with the only substantive difference being 
whether IETF review is required.

The nature of this proposed "event package" as a framework (as opposed 
to an actual event package as RFC 3265 defines that term) is revealed 
substantially by the contents of two of its sections: section 10 and 
section 12. In section 10, the draft tacitly indicates that it is to be 
applied to a broad range of events:

    The durations of push event subscriptions, and in general the
    conditions under which they are terminated, vary widely with the
    nature of the applications as they are assumed to be application-
    specific criteria.


Even more tellingly, section 12 simply indicates that the notification 
bodies can be carried either inline or via indirection, and gives no 
actual definition of body type or of body syntax. By contrast, consider 
the body of published event packages:

    * RFC4575 defines a new application/conference-info+xml for NOTIFY
    * RFC5362 defines a new application/resource-lists+xml for NOTIFY
    * RFC4235 defines a new application/dialog-info+xml for NOTIFY
    * RFC4730 defined a new application/kpml-request+xml for NOTIFY
    * RFC3842 defines a new appcliation/simple-message-summary for NOTIFY
    * RFC4354 defines a new application/poc-settings+xml for NOTIFY
    * RFC3856 uses application/pidf+xml (which was designed for this
      purpose) for NOTIFY
    * RFC3680 defines a new application/reginfo+xml for NOTIFY
    * RFC3515 uses message/sipfrag to convey the state of the indicated
      resource in a NOTIFY
    * RFC3910 defines a new application/spirits-event+xml for NOTIFY
    * RFC3857 defines a new application/watcherinfo+xml for NOTIFY


A legitimate event package would indicate specifically and completely 
what state is to be conveyed. It will also define or normatively 
reference a formally-defined syntax for conveying the state. The fact 
that the draft cannot provide such a description of state and a syntax 
for conveying the state is a good indication that it is not an event 
package, but actually an attempt to wrap RFC3265 in a parallel framework.

If the overall mechanism proposed by the current document is approved as 
an IETF-sanctioned event package, the net result will be a loophole that 
effectively eliminates section 4.1 of rfc3427bis entirely.

By my reading of subject matter covered by this document -- and with the 
caveat that I am not familiar with any of the OMA applications that the 
mechanism is intended to enable -- the only path forward that both 
achieves the goals stated in this document's problem statement and 
conforms to the restrictions put forth in RFC 3427 (and its draft 
successor) is to define each of the applications as a new event package. 
Unless I misunderstand the mechanism defined in OMA-TS-SIP_PUSH, there 
appear to be approximately 23 such applications presently defined [1]. 
For the sake of expediency, it may be sensible to have a single document 
that defines the RFC3265 event packages for all of them -- that is, a 
single draft with 23 distinct copies of the sections required by section 
4.4 of RFC3265.

/a

[1] A (partial?) list of defined applications appears to be published 
here: http://www.wapforum.org/wina/push-app-id.htm

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<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
The area directors for RAI asked me to review
draft-mdolly-dispatch-oma-push, specifically as they relate to the
ongoing revisions to RFCs 3265 and 3427.<br>
<br>
While there are some relatively minor issues with some of the 3265
concepts used in this document, there is an overarching structural
issue that needs to be addressed.<br>
<br>
By my reading, what this document seeks to do is open a door for OMA to
create new event packages without requiring any IETF review (and, even
if such is not the intent, it would certainly be the outcome). This is
achieved by using an event package type of "oma-push," and then using a
new "Event" header field parameter of "event-app-id" to indicate the
actual event package. In the language of object-oriented programming,
this draft seeks to define an event package that is little more than a
fa&ccedil;ade to wrap RFC 3265, with the only substantive difference being
whether IETF review is required.<br>
<br>
The nature of this proposed "event package" as a framework (as opposed
to an actual event package as RFC 3265 defines that term) is revealed
substantially by the contents of two of its sections: section 10 and
section 12. In section 10, the draft tacitly indicates that it is to be
applied to a broad range of events:<br>
<br>
<pre>   The durations of push event subscriptions, and in general the
   conditions under which they are terminated, vary widely with the
   nature of the applications as they are assumed to be application-
   specific criteria.
</pre>
<br>
Even more tellingly, section 12 simply indicates that the notification
bodies can be carried either inline or via indirection, and gives no
actual definition of body type or of body syntax. By contrast, consider
the body of published event packages:<br>
<ul>
  <li>RFC4575 defines a new application/conference-info+xml for NOTIFY</li>
  <li>RFC5362 defines a new application/resource-lists+xml for NOTIFY</li>
  <li>RFC4235 defines a new application/dialog-info+xml for NOTIFY</li>
  <li>RFC4730 defined a new application/kpml-request+xml for NOTIFY</li>
  <li>RFC3842 defines a new appcliation/simple-message-summary for
NOTIFY</li>
  <li>RFC4354 defines a new application/poc-settings+xml for NOTIFY</li>
  <li>RFC3856 uses application/pidf+xml (which was designed for this
purpose) for NOTIFY</li>
  <li>RFC3680 defines a new application/reginfo+xml for NOTIFY</li>
  <li>RFC3515 uses message/sipfrag to convey the state of the indicated
resource in a NOTIFY<br>
  </li>
  <li>RFC3910 defines a new application/spirits-event+xml for NOTIFY</li>
  <li>RFC3857 defines a new application/watcherinfo+xml for NOTIFY</li>
</ul>
<br>
A legitimate event package would indicate specifically and completely
what state is to be conveyed. It will also define or normatively
reference a formally-defined syntax for conveying the state. The fact
that the draft cannot provide such a description of state and a syntax
for conveying the state is a good indication that it is not an event
package, but actually an attempt to wrap RFC3265 in a parallel
framework.<br>
<br>
If the overall mechanism proposed by the current document is approved
as an IETF-sanctioned event package, the net result will be a loophole
that effectively eliminates section 4.1 of rfc3427bis entirely.<br>
<br>
By my reading of subject matter covered by this document -- and with
the caveat that I am not familiar with any of the OMA applications that
the mechanism is intended to enable -- the only path forward that both
achieves the goals stated in this document's problem statement and
conforms to the restrictions put forth in RFC 3427 (and its draft
successor) is to define each of the applications as a new event
package. Unless I misunderstand the mechanism defined in
OMA-TS-SIP_PUSH, there appear to be approximately 23 such applications
presently defined [1]. For the sake of expediency, it may be sensible
to have a single document that defines the RFC3265 event packages for
all of them -- that is, a single draft with 23 distinct copies of the
sections required by section 4.4 of RFC3265.<br>
<br>
/a<br>
<br>
[1] A (partial?) list of defined applications appears to be published
here: <a class="moz-txt-link-freetext"
 href="http://www.wapforum.org/wina/push-app-id.htm">http://www.wapforum.org/wina/push-app-id.htm</a>
</body>
</html>

--------------000502060803010708050500--

From Henry.Lum@alcatel-lucent.com  Mon Feb  1 09:33:44 2010
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From: "Henry Lum" <Henry.Lum@alcatel-lucent.com>
To: "Leon Portman" <Leon.Portman@nice.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] Comments on draft-rehor-dispatch-session-recording-req-00
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Hi Leon,

=20

Thanks for the quick response, I have a few more follow ups marked
[hlum] below.

=20

Regards,

Henry=20

=20

From: Leon Portman [mailto:Leon.Portman@nice.com]=20
Sent: Sunday, January 31, 2010 6:03 AM
To: Henry Lum; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: Comments on draft-rehor-dispatch-session-recording-req-00

=20

Hi, Henry.

=20

I will try to answer some of the questions. Please see below.

=20

Leon

=20

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Henry Lum
Sent: Sunday, January 31, 2010 6:11 AM
To: dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: [dispatch] Comments on
draft-rehor-dispatch-session-recording-req-00

=20

Hi Ken,

=20

I have a bunch of questions and comments on the requirements for SRP
requirements.

=20

-          Section 3 - Recording Client - If the SRP requirement does
not mandate to use SIP as the actual recording protocol, why does the
recording client have to be a SIP user agent or B2BUA?

[LeonP] Actually there is a difference between Recorded session (the
actual call to be recorded) that can be not SIP based and Recording
Session (part of SRP) that is used for triggering recording and it is
SIP based. So both RC and RS are SIP UA.

[[hlum]] I now understand that the recorded session does not have to be
SIP based, but the wording in the definition of SRP itself mentions that
it may be SIP; then why does RC and RS have to be a SIP UA if SRP does
not have to be SIP?

-          Section 3 - Recorded session - Is a recorded session has to
be a SIP session?

[LeonP] No, my comment above

-          Section 3 - Persistent Recording - does recorded session
exist in this context since it seems like the recording session is
persistent from the endpoint perspective.

[LeonP] No, recorded session are not required to be existing during
initiation of persistent session. Persistent recording session can be
initiated on agent logon for example.

-          Section 4 - use case 5 - if the recording can include
recording for an IVR application (ie. VoiceXML application), does that
mean the call metadata must include application contexts as well?

[LeonP] Yes, like specific script or input results

[[hlum]] I assume what kind of call metadata and how it is delivered is
outside of the scope of this document?

-          Section 4, use case 7 - the first sentence has a strange word
"&mdash". This use case somehow implies certain architectural design for
a PBX with regards to media forking. Is that necessary from a
requirement perspective?

[LeonP] XML convert mistake. We don't want assume any specific
configurations.

-          Section 4, use case 8 - shall we use a more general
terminology such as multi-party call? Depending on the situation, the
supervisor may be conferenced with the agent and customer so the
recording session needs to define how a multi-party session should be
recorded.

[LeonP] Yes, probably we want to use more generic term then Complex Call

-          Section 4, use case 11, does a recorder "must" support some
sort of media processing such as real-time analytics? This is more of a
business application on the recording server side, which is independent
of the session recording protocol itself.

[LeonP] The main point there was that Recording Protocol must support
real time streaming for real time analytics

-          Section 6:

o   REQ-001 - what does full-time Recording session mean?

[LeonP] Record all calls

o   REQ-003 - it's unclear what is the target of the recording session
here since this not directly related to a single recorded session
(perhaps many)

[LeonP] To minimize media clipping associated with recording initiation

[[hlum]] You actually answered my question above about what a persistent
recording mean and why it's useful, but what is unclear to me that is
whether there is something called a persistent recording session that
targets something like an agent device.

o   REQ-006 - I mentioned above that IVR sessions may contain additional
IVR application metadata; should this requirement clarify that the
mechanism should include application metadata?

[LeonP] Yes, we should

o   REQ-007 - this wording of the requirement to me is pointing towards
a specific implementation of high-availability or fault-tolerance of a
recording session.

o   REQ-009 - the wording here is a bit unclear here. Do you mean the
recorded session represents a multi-party session where there are
multiple media streams from multiple recorded sessions, and the media
streams may be mixed by a conference mixer?

[LeonP]  It is more for trading floor environments where multiple
concurrent calls on the same device (turret) recorded together. It is
not related to summation for conference participants at the same call.

[[hlum]] I'm not familiar with trading floor environments so I may not
understand the intent here. If you have multiple concurrent calls then
wouldn't you have multiple recording sessions to describe each call
individually? Does the endpoint device perform mixing function for
multiple calls so that's why you need to club them together in a single
recording session?

o   REQ-013 - this is only one way of notifying the customer that the
call is being recorded but there are others. I think this requires a bit
more explanation of why this is needed. To me this looks like a
duplicate of REQ-022

o   REQ-015 - what does non-repudiation mean in this context? Isn't this
the same as REQ-017?

[LeonP] REQ-015 talk regarding media and REQ-017 regarding SRP SIP
Session

o   REQ-021 - what happens with SRTP if recording client simply forks
the media? The recording server would not be able to decrypt the media
unless it gets the private keys from the recorded session.

[LeonP] It is out of scope of this document.=20

o   REQ-028 - the wording redirected looks out of context here; the
requirement only applies when recording client initiates a recording
session to an available recorder server?

[LeonP] We want to support both directions

[[hlum]] Okay, then I think we should re-word to mention something about
availability in general.  (eg. "A request for a new recording session
must reach an available recording client or server").

o   REQ-029/030 - They look very generic and I don't think these should
be requirements for SRP, but rather a product requirement on how OA&M
works for the recording client/server.

=20

Thanks

Henry

=20

=20


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'color:blue'>Hi Leon,<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'color:blue'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:blue'>Thanks for the quick resp=
onse, I
have a few more follow ups marked [hlum] below.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:blue'><o:p>&nbsp;</o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:blue'>Regards,<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'color:blue'>Henry <o:p></o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:blue'><o:p>&nbsp;</o:p></span><=
/p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Taho=
ma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Leon Portma=
n
[mailto:Leon.Portman@nice.com] <br>
<b>Sent:</b> Sunday, January 31, 2010 6:03 AM<br>
<b>To:</b> Henry Lum; dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> RE: Comments on draft-rehor-dispatch-session-recording-re=
q-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>Hi,
Henry.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>I
will try to answer some of the questions. Please see below.<o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>Leon<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Taho=
ma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> dispatch-bo=
unces@ietf.org
[mailto:dispatch-bounces@ietf.org] <b>On Behalf Of </b>Henry Lum<br>
<b>Sent:</b> Sunday, January 31, 2010 6:11 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> [dispatch] Comments on
draft-rehor-dispatch-session-recording-req-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hi Ken,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I have a bunch of questions and comments on the requ=
irements
for SRP requirements.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 &#8211; Recording Client &#8211; If the=
 SRP
requirement does not mandate to use SIP as the actual recording protocol,=
 why
does the recording client have to be a SIP user agent or B2BUA?<o:p></o:p=
></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Actually there is a difference between Recorded se=
ssion
(the actual call to be recorded) that can be not SIP based and Recording
Session (part of SRP) that is used for triggering recording and it is SIP=

based. So both RC and RS are SIP UA.<o:p></o:p></span></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I now understand=
 that the
recorded session does not have to be SIP based, but the wording in the de=
finition
of SRP itself mentions that it may be SIP; then why does RC and RS have t=
o be a
SIP UA if SRP does not have to be SIP?<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 - Recorded session &#8211; Is a recorde=
d
session has to be a SIP session?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] No, my comment above</span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 - Persistent Recording &#8211; does rec=
orded
session exist in this context since it seems like the recording session i=
s
persistent from the endpoint perspective.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] No, recorded session are not required to be existi=
ng
during initiation of persistent session. Persistent recording session can=
 be
initiated on agent logon for example.</span></i></b><span style=3D'font-f=
amily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4 &#8211; use case 5 &#8211; if the recor=
ding
can include recording for an IVR application (ie. VoiceXML application), =
does
that mean the call metadata must include application contexts as well?<o:=
p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, like specific script or input results<o:p></o=
:p></span></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I assume what ki=
nd of call
metadata and how it is delivered is outside of the scope of this document=
?<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 7 &#8211; the first sentence =
has a
strange word &#8220;&amp;mdash&#8221;. This use case somehow implies cert=
ain
architectural design for a PBX with regards to media forking. Is that nec=
essary
from a requirement perspective?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] XML convert mistake. We don&#8217;t want assume an=
y
specific configurations.</span></i></b><span style=3D'font-family:"Arial"=
,"sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 8 &#8211; shall we use a more=

general terminology such as multi-party call? Depending on the situation,=
 the
supervisor may be conferenced with the agent and customer so the recordin=
g
session needs to define how a multi-party session should be recorded.<o:p=
></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, probably we want to use more generic term the=
n
Complex Call</span></i></b><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 11, does a recorder
&#8220;must&#8221; support some sort of media processing such as real-tim=
e
analytics? This is more of a business application on the recording server=
 side,
which is independent of the session recording protocol itself.<o:p></o:p>=
</p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] The main point there was that Recording Protocol m=
ust
support real time streaming for real time analytics</span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 6:<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-001 &#8211; what does full-time Record=
ing
session mean?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Record all calls</span></i></b><span style=3D'font=
-family:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-003 &#8211; it&#8217;s unclear what is=
 the
target of the recording session here since this not directly related to a=

single recorded session (perhaps many)<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] To minimize media clipping associated with recordi=
ng
initiation<o:p></o:p></span></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] You actually ans=
wered my
question above about what a persistent recording mean and why it&#8217;s =
useful,
but what is unclear to me that is whether there is something called a per=
sistent
recording session that targets something like an agent device.<o:p></o:p>=
</span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-006 &#8211; I mentioned above that IVR=
 sessions
may contain additional IVR application metadata; should this requirement
clarify that the mechanism should include application metadata?<o:p></o:p=
></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, we should</span></i></b><span style=3D'font-f=
amily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-007 &#8211; this wording of the requir=
ement
to me is pointing towards a specific implementation of high-availability =
or
fault-tolerance of a recording session.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-009 &#8211; the wording here is a bit
unclear here. Do you mean the recorded session represents a multi-party s=
ession
where there are multiple media streams from multiple recorded sessions, a=
nd the
media streams may be mixed by a conference mixer?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>&nbsp;</span>=
</i></b><span
style=3D'color:#1F497D'>It is more for trading floor environments where m=
ultiple
concurrent calls on the same device (turret) recorded together. It is not=

related to summation for conference participants at the same call.</span>=
<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I&#8217;m not fa=
miliar
with trading floor environments so I may not understand the intent here. =
If you
have multiple concurrent calls then wouldn&#8217;t you have multiple reco=
rding
sessions to describe each call individually? Does the endpoint device per=
form
mixing function for multiple calls so that&#8217;s why you need to club t=
hem
together in a single recording session?<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-013 &#8211; this is only one way of
notifying the customer that the call is being recorded but there are othe=
rs. I
think this requires a bit more explanation of why this is needed. To me t=
his
looks like a duplicate of REQ-022<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-015 &#8211; what does non-repudiation =
mean
in this context? Isn&#8217;t this the same as REQ-017?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>REQ-015 talk
regarding media and REQ-017 regarding SRP SIP Session</span></i></b><o:p>=
</o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-021 &#8211; what happens with SRTP if
recording client simply forks the media? The recording server would not b=
e able
to decrypt the media unless it gets the private keys from the recorded se=
ssion.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] It is out of scope of this document. </span></i></=
b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-028 &#8211; the wording redirected loo=
ks out
of context here; the requirement only applies when recording client initi=
ates a
recording session to an available recorder server?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] We want to support both directions<o:p></o:p></spa=
n></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] Okay, then I thi=
nk we
should re-word to mention something about availability in general.&nbsp; =
(eg. &#8220;A
request for a new recording session must reach an available recording cli=
ent or
server&#8221;).<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-029/030 &#8211; They look very generic=
 and I
don&#8217;t think these should be requirements for SRP, but rather a prod=
uct
requirement on how OA&amp;M works for the recording client/server.<o:p></=
o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Henry<o:p></o:p></p>

<div>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'>&nbsp;<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
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<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
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To: Henry Lum <Henry.Lum@alcatel-lucent.com>, Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 1 Feb 2010 18:28:23 -0500
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Henry,

Thanks for your comments. Inline:


-        Section 3 - Recording Client - If the SRP requirement does not man=
date to use SIP as the actual recording protocol, why does the recording cl=
ient have to be a SIP user agent or B2BUA?
[LeonP] Actually there is a difference between Recorded session (the actual=
 call to be recorded) that can be not SIP based and Recording Session (part=
 of SRP) that is used for triggering recording and it is SIP based. So both=
 RC and RS are SIP UA.
[[hlum]] I now understand that the recorded session does not have to be SIP=
 based, but the wording in the definition of SRP itself mentions that it ma=
y be SIP; then why does RC and RS have to be a SIP UA if SRP does not have =
to be SIP?

[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).



-        Section 4 - use case 5 - if the recording can include recording fo=
r an IVR application (ie. VoiceXML application), does that mean the call me=
tadata must include application contexts as well?
[LeonP] Yes, like specific script or input results
[[hlum]] I assume what kind of call metadata and how it is delivered is out=
side of the scope of this document?

[rj] Correct.


o   REQ-003 - it's unclear what is the target of the recording session here=
 since this not directly related to a single recorded session (perhaps many=
)
[LeonP] To minimize media clipping associated with recording initiation
[[hlum]] You actually answered my question above about what a persistent re=
cording mean and why it's useful, but what is unclear to me that is whether=
 there is something called a persistent recording session that targets some=
thing like an agent device.

[rj] I guess, Persistent Recording is a bit confusing term (we've received =
other feedback on this as well). If I can use the term always-on recording =
from the wing-srtp draft then these sessions get established when the agent=
 logs on and torn off when the agent logs off. Codecs such as G.729 and G.7=
11 with VAD are used to ensure that silence is not recorded (for instance, =
when the agent isn't on a call).


o   REQ-009 - the wording here is a bit unclear here. Do you mean the recor=
ded session represents a multi-party session where there are multiple media=
 streams from multiple recorded sessions, and the media streams may be mixe=
d by a conference mixer?
[LeonP]  It is more for trading floor environments where multiple concurren=
t calls on the same device (turret) recorded together. It is not related to=
 summation for conference participants at the same call.
[[hlum]] I'm not familiar with trading floor environments so I may not unde=
rstand the intent here. If you have multiple concurrent calls then wouldn't=
 you have multiple recording sessions to describe each call individually? D=
oes the endpoint device perform mixing function for multiple calls so that'=
s why you need to club them together in a single recording session?

[rj] Due to bandwidth and other cost burdens, the customer demand is that y=
ou mix multiple _recorded sessions_ in a single _recording session_.



o   REQ-028 - the wording redirected looks out of context here; the require=
ment only applies when recording client initiates a recording session to an=
 available recorder server?
[LeonP] We want to support both directions
[[hlum]] Okay, then I think we should re-word to mention something about av=
ailability in general.  (eg. "A request for a new recording session must re=
ach an available recording client or server").

[rj] The example test you quoted doesn't reflect the actual intent here. Th=
e reason for supporting both directions is dependent on where the policy to=
 record or not resides - RC or RS. The INVITE always comes from either of t=
hese two places.

Thanks,
Raj Jain


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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Henry,<o:p></o:p></spa=
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks for your commen=
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
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4.0pt">
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;
</span></span><![endif]>Section 3 &#8211; Recording Client &#8211; If the S=
RP requirement does not mandate to use SIP as the actual recording protocol=
, why does the recording client have to be a SIP user agent or B2BUA?<o:p><=
/o:p></p>
<p class=3D"MsoNormal"><b><i>[LeonP] Actually there is a difference between=
 Recorded session (the actual call to be recorded) that can be not SIP base=
d and Recording Session (part of SRP) that is used for triggering recording=
 and it is SIP based. So both RC and
 RS are SIP UA.<o:p></o:p></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:blue">[[hlum]] I now understand=
 that the recorded session does not have to be SIP based, but the wording i=
n the definition of SRP itself mentions that it may be SIP; then why does R=
C and RS have to be a SIP UA if SRP
 does not have to be SIP?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">[rj] The SRP, which ma=
nages how the recording media gets delivered to the recorders, is based on =
SIP (SRP is a set of extensions to the SIP core protocol). The sessions tha=
t SRP manages are called &#8220;Recording
 Sessions&#8221; and they are SIP sessions. Therefore, RC and RS are, by de=
finition, SIP UAs. The sessions that get actually recorded (e.g. communicat=
ion between a call center agent and an external caller) may or may not be S=
IP (this depends on the kind of the PBX
 phone the agent). <o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">-<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;
</span></span><![endif]>Section 4 &#8211; use case 5 &#8211; if the recordi=
ng can include recording for an IVR application (ie. VoiceXML application),=
 does that mean the call metadata must include application contexts as well=
?<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i>[LeonP] Yes, like specific script or input res=
ults<o:p></o:p></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:blue">[[hlum]] I assume what ki=
nd of call metadata and how it is delivered is outside of the scope of this=
 document?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">[rj] Correct.<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2">
<![if !supportLists]><span style=3D"font-family:&quot;Courier New&quot;"><s=
pan style=3D"mso-list:Ignore">o<span style=3D"font:7.0pt &quot;Times New Ro=
man&quot;">&nbsp;&nbsp;
</span></span></span><![endif]>REQ-003 &#8211; it&#8217;s unclear what is t=
he target of the recording session here since this not directly related to =
a single recorded session (perhaps many)<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i>[LeonP] To minimize media clipping associated =
with recording initiation<o:p></o:p></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:blue">[[hlum]] You actually ans=
wered my question above about what a persistent recording mean and why it&#=
8217;s useful, but what is unclear to me that is whether there is something=
 called a persistent recording session that
 targets something like an agent device.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">[rj] I guess, Persiste=
nt Recording is a bit confusing term (we&#8217;ve received other feedback o=
n this as well). If I can use the term always-on recording from the wing-sr=
tp draft then these sessions get established
 when the agent logs on and torn off when the agent logs off. Codecs such a=
s G.729 and G.711 with VAD are used to ensure that silence is not recorded =
(for instance, when the agent isn&#8217;t on a call).<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2">
<![if !supportLists]><span style=3D"font-family:&quot;Courier New&quot;"><s=
pan style=3D"mso-list:Ignore">o<span style=3D"font:7.0pt &quot;Times New Ro=
man&quot;">&nbsp;&nbsp;
</span></span></span><![endif]>REQ-009 &#8211; the wording here is a bit un=
clear here. Do you mean the recorded session represents a multi-party sessi=
on where there are multiple media streams from multiple recorded sessions, =
and the media streams may be mixed by
 a conference mixer?<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i>[LeonP] <span style=3D"color:#1F497D">&nbsp;</=
span></i></b><b><i>It is more for trading floor environments where multiple=
 concurrent calls on the same device (turret) recorded together. It is not =
related to summation for conference participants
 at the same call.<o:p></o:p></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:blue">[[hlum]] I&#8217;m not fa=
miliar with trading floor environments so I may not understand the intent h=
ere. If you have multiple concurrent calls then wouldn&#8217;t you have mul=
tiple recording sessions to describe each call individually?
 Does the endpoint device perform mixing function for multiple calls so tha=
t&#8217;s why you need to club them together in a single recording session?=
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">[rj] Due to bandwidth =
and other cost burdens, the customer demand is that you mix multiple _<i>re=
corded sessions</i>_ in a single _<i>recording session</i>_.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoListParagraph" style=3D"margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2">
<![if !supportLists]><span style=3D"font-family:&quot;Courier New&quot;"><s=
pan style=3D"mso-list:Ignore">o<span style=3D"font:7.0pt &quot;Times New Ro=
man&quot;">&nbsp;&nbsp;
</span></span></span><![endif]>REQ-028 &#8211; the wording redirected looks=
 out of context here; the requirement only applies when recording client in=
itiates a recording session to an available recorder server?<o:p></o:p></p>
<p class=3D"MsoNormal"><b><i>[LeonP] We want to support both directions<o:p=
></o:p></i></b></p>
<p class=3D"MsoNormal"><span style=3D"color:blue">[[hlum]] Okay, then I thi=
nk we should re-word to mention something about availability in general.&nb=
sp; (eg. &#8220;A request for a new recording session must reach an availab=
le recording client or server&#8221;).</span><span style=3D"color:#1F497D">
</span><o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">[rj] The example test =
you quoted doesn&#8217;t reflect the actual intent here. The reason for sup=
porting both directions is dependent on where the policy to record or not r=
esides &#8211; RC or RS. The INVITE always comes
 from either of these two places.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Raj Jain<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
</div>
</div>
<br>
<hr>
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From Leon.Portman@nice.com  Mon Feb  1 23:40:13 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: Dave Smith <Dave.Smith@genesyslab.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 2 Feb 2010 09:40:33 +0200
Thread-Topic: Comments on draft-rehor-dispatch-session-recording-req-00
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Dave, Hi

Thank you very much for the comments.

Please see below.

We can also discuss it on our call today.

Leon
From: Dave Smith [mailto:Dave.Smith@genesyslab.com]
Sent: Tuesday, February 02, 2010 7:03 AM
To: dispatch@ietf.org
Cc: krehor@cisco.com; Leon Portman; Henry Lum
Subject: RE: Comments on draft-rehor-dispatch-session-recording-req-00

Hi Leon, Henry,
I put together some inputs on top of yours for consideration as well....pls=
 see my inputs below.

Regards,
Dave

From: Leon Portman [mailto:Leon.Portman@nice.com]
Sent: Sunday, January 31, 2010 3:03 AM
To: Henry Lum; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: Comments on draft-rehor-dispatch-session-recording-req-00

Hi, Henry.

I will try to answer some of the questions. Please see below.

Leon

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Henry Lum
Sent: Sunday, January 31, 2010 6:11 AM
To: dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: [dispatch] Comments on draft-rehor-dispatch-session-recording-req-=
00

Hi Ken,

I have a bunch of questions and comments on the requirements for SRP requir=
ements.


-          Section 3 - Recording Client - If the SRP requirement does not m=
andate to use SIP as the actual recording protocol, why does the recording =
client have to be a SIP user agent or B2BUA?
[LeonP] Actually there is a difference between Recorded session (the actual=
 call to be recorded[DS]  Is this just the voice - the RTP stream (the repl=
icated voice stream that is sent to a Logger)?  ) that can be not SIP based=
 and Recording Session (part of SRP) that is used for triggering recording =
and it is SIP based[DS] Is this the control messaging...for starting/stoppi=
ng/pause/resume...the messaging (SIP) for set up of the replicated voice st=
ream/RTP transfer? . So both RC and RS are SIP UA.
[LeonP] Yes, SRP is the control session for replicated media stream (any RT=
P type of media)

-          Section 3 - Recorded session - Is a recorded session has to be a=
 SIP session?
[LeonP] No, my comment above[DS] I recommend better Terms and/or better def=
initions of the Terms "Recorded Session" and "Recording Session"in the Draf=
t - is it possible to change one or both terms since they sound so similar.=
  That is, the IETF Draft definitions are not clear to me...and the differe=
nce between these is not clear (for example, do not use the terms themselve=
s in the definitions...provide examples.  Recommend also to differentiate t=
hese from the Metadata messaging (if they are or could be different), and t=
o define that as well (as rec'd by Martin).
[LeonP] Let's talk about it

-          Section 3 - Persistent Recording - does recorded session exist i=
n this context since it seems like the recording session is persistent from=
 the endpoint perspective.
[LeonP] No, recorded session are not required to be existing during initiat=
ion of persistent session. Persistent recording session can be initiated on=
 agent logon for example.

-          Section 4 - use case 5 - if the recording can include recording =
for an IVR application (ie. VoiceXML application), does that mean the call =
metadata must include application contexts as well?
[LeonP] Yes, like specific script or input results

-          Section 4, use case 7 - the first sentence has a strange word "&=
mdash". This use case somehow implies certain architectural design for a PB=
X with regards to media forking. Is that necessary from a requirement persp=
ective?
[LeonP] XML convert mistake. We don't want assume any specific configuratio=
ns.

-          Section 4, use case 8 - shall we use a more general terminology =
such as multi-party call? Depending on the situation, the supervisor may be=
 conferenced with the agent and customer so the recording session needs to =
define how a multi-party session should be recorded.
[LeonP] Yes, probably we want to use more generic term then Complex Call[DS=
] The term "Multi-Party Call" seems to imply a call/interaction with more t=
han two users...that is, a conference.  Recommend to keep "Complex Calls" t=
erminology but define several different cases/examples - for example, Confe=
rences (Henry's example), Consults (as in use case 8), also could include a=
 Whisper Agent/Coaching example, Barge-In example, other...  Alternatively,=
 change Use Case 8 to "Consult Call" and add additional use cases for these=
 other complex call examples.

-          Section 4, use case 11, does a recorder "must" support some sort=
 of media processing such as real-time analytics? This is more of a busines=
s application on the recording server side, which is independent of the ses=
sion recording protocol itself.
[LeonP] The main point there was that Recording Protocol must support real =
time streaming for real time analytics[DS]  This Use Case reads like a requ=
irement and should read more like an example of how it is used.
Is the ability to support "real-time" analytics dependent on the recording =
protocol?  Is there also a need separate streams for the two channels (what=
 the agent said, what the agent heard) - And do we already have a req't for=
 that.....i.e., req't 10 and/or req't 11?

-          Section 6:

o   REQ-001 - what does full-time Recording session mean?
[LeonP] Record all calls[DS] Recommend we explain that in the req't; also, =
is that by DN, or by Agent, or by Customer, other, or all of the above...?
[LeonP] Yes, the requirement is to initiate recording session not by call i=
d but by some fixed identifier as above

o   REQ-003 - it's unclear what is the target of the recording session here=
 since this not directly related to a single recorded session (perhaps many=
)
[LeonP] To minimize media clipping associated with recording initiation

o   REQ-006 - I mentioned above that IVR sessions may contain additional IV=
R application metadata; should this requirement clarify that the mechanism =
should include application metadata?
[LeonP] Yes, we should

o   REQ-007 - this wording of the requirement to me is pointing towards a s=
pecific implementation of high-availability or fault-tolerance of a recordi=
ng session.

o   REQ-009 - the wording here is a bit unclear here. Do you mean the recor=
ded session represents a multi-party session where there are multiple media=
 streams from multiple recorded sessions, and the media streams may be mixe=
d by a conference mixer?
[LeonP]  It is more for trading floor environments where multiple concurren=
t calls on the same device (turret) recorded together. It is not related to=
 summation for conference participants at the same call.     [DS] Recommend=
 to state this info in the req'ts and provide this example.

o   REQ-013 - this is only one way of notifying the customer that the call =
is being recorded but there are others. I think this requires a bit more ex=
planation of why this is needed. To me this looks like a duplicate of REQ-0=
22    [DS] Recommend to change this req't from "Must" to "May"

o   REQ-015 - what does non-repudiation mean in this context? Isn't this th=
e same as REQ-017?
[LeonP] REQ-015 talk regarding media and REQ-017 regarding SRP SIP Session =
[DS] Recommend to define "non-repudiation" and/or provide an example to avo=
id any confusion.

o   REQ-017 - [DS]  Why is non-repudiation of both the SRP and the Media ne=
eded?

o   Req - 022  - [DS]  Recommend to add "This MAY be performed by injecting=
 tones into the Recorded Session either from the RS or from the RC."  (but =
could also be done by informing via speech script upon first answering call=
)

o   REQ-021 - what happens with SRTP if recording client simply forks the m=
edia? The recording server would not be able to decrypt the media unless it=
 gets the private keys from the recorded session.
[LeonP] It is out of scope of this document.

o   REQ-028 - the wording redirected looks out of context here; the require=
ment only applies when recording client initiates a recording session to an=
 available recorder server?
[LeonP] We want to support both directions

o   REQ-029/030 - They look very generic and I don't think these should be =
requirements for SRP, but rather a product requirement on how OA&M works fo=
r the recording client/server.

Thanks
Henry



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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Dave,
Hi<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Thank
you very much for the comments.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Please
see below.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>We
can also discuss it on our call today.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Leon<o:p></o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Dave Smith
[mailto:Dave.Smith@genesyslab.com] <br>
<b>Sent:</b> Tuesday, February 02, 2010 7:03 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Cc:</b> krehor@cisco.com; Leon Portman; Henry Lum<br>
<b>Subject:</b> RE: Comments on draft-rehor-dispatch-session-recording-req-=
00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'>Hi
Leon, Henry,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'>I
put together some inputs on top of yours for consideration as well&#8230;.p=
ls
see my inputs below.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'>Regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'>Dave<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#993366'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Leon Portman =
[mailto:Leon.Portman@nice.com]
<br>
<b>Sent:</b> Sunday, January 31, 2010 3:03 AM<br>
<b>To:</b> Henry Lum; dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> RE: Comments on draft-rehor-dispatch-session-recording-req-=
00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Hi,
Henry.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>I
will try to answer some of the questions. Please see below.<o:p></o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'>Leon<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";color:=
#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On Behalf O=
f </b>Henry
Lum<br>
<b>Sent:</b> Sunday, January 31, 2010 6:11 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> [dispatch] Comments on draft-rehor-dispatch-session-recordi=
ng-req-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hi Ken,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I have a bunch of questions and comments on the requir=
ements
for SRP requirements.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 3 &#8211; Recording =
Client
&#8211; If the SRP requirement does not mandate to use SIP as the actual
recording protocol, why does the recording client have to be a SIP user age=
nt
or B2BUA?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Actually there is a difference between Recorded sess=
ion
(the actual call to be recorded</span></i></b><b><i><span style=3D'font-fam=
ily:
"Arial","sans-serif";color:#993366'>[DS]&nbsp; Is this just the voice - the=
 RTP
stream (the replicated voice stream that is sent to a Logger)?&nbsp; </span=
></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'>) that can be not =
SIP
based and Recording Session (part of SRP) that is used for triggering recor=
ding
and it is SIP based</span></i></b><b><i><span style=3D'font-family:"Arial",=
"sans-serif";
color:#993366'>[DS] Is this the control messaging&#8230;for
starting/stopping/pause/resume&#8230;the messaging (SIP) for set up of the
replicated voice stream/RTP transfer? </span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'>. So both RC and R=
S are
SIP UA.</span></i></b><b><i><span style=3D'font-family:"Arial","sans-serif"=
;
color:#993366'>&nbsp; </span></i></b><b><i><span style=3D'font-family:"Aria=
l","sans-serif";
color:#1F497D'><o:p></o:p></span></i></b></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Yes, SRP is the control session for replicated media
stream (any RTP type of media)</span></i></b><span style=3D'font-family:"Ar=
ial","sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 3 - Recorded session
&#8211; Is a recorded session has to be a SIP session?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] No, my comment above</span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] I recommend b=
etter
Terms and/or better definitions of the Terms &#8220;Record<u>ed</u>
Session&#8221; and &#8220;Record<u>ing</u> Session&#8221;in the Draft &#821=
1;
is it possible to change one or both terms since they sound so similar.&nbs=
p;
That is, the IETF Draft definitions are not clear to me&#8230;and the
difference between these is not clear (for example, do not use the terms
themselves in the definitions&#8230;provide examples.&nbsp; Recommend also =
to
differentiate these from the Metadata messaging (if they are or could be
different), and to define that as well (as rec&#8217;d by Martin).&nbsp; </=
span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</i></b></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Let&#8217;s talk about it</span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 3 - Persistent Recor=
ding
&#8211; does recorded session exist in this context since it seems like the
recording session is persistent from the endpoint perspective.<o:p></o:p></=
p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] No, recorded session are not required to be existing
during initiation of persistent session. Persistent recording session can b=
e
initiated on agent logon for example.</span></i></b><span style=3D'font-fam=
ily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 4 &#8211; use case 5
&#8211; if the recording can include recording for an IVR application (ie.
VoiceXML application), does that mean the call metadata must include
application contexts as well?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Yes, like specific script or input results</span></i=
></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 4, use case 7 &#8211=
; the
first sentence has a strange word &#8220;&amp;mdash&#8221;. This use case
somehow implies certain architectural design for a PBX with regards to medi=
a
forking. Is that necessary from a requirement perspective?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] XML convert mistake. We don&#8217;t want assume any
specific configurations.</span></i></b><span style=3D'font-family:"Arial","=
sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 4, use case 8 &#8211=
;
shall we use a more general terminology such as multi-party call? Depending=
 on
the situation, the supervisor may be conferenced with the agent and custome=
r so
the recording session needs to define how a multi-party session should be
recorded.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Yes, probably we want to use more generic term then
Complex Call</span></i></b><b><i><span style=3D'font-family:"Arial","sans-s=
erif";
color:#993366'>[DS] The term &#8220;Multi-Party Call&#8221; seems to imply =
a
call/interaction with more than two users&#8230;that is, a conference.&nbsp=
;
Recommend to keep &#8220;Complex Calls&#8221; terminology but define severa=
l
different cases/examples &#8211; for example, Conferences (Henry&#8217;s
example), Consults (as in use case 8), also could include a Whisper
Agent/Coaching example, Barge-In example, other&#8230;&nbsp; Alternatively,
change Use Case 8 to &#8220;Consult Call&#8221; and add additional use case=
s
for these other complex call examples.&nbsp; </span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 4, use case 11, does=
 a
recorder &#8220;must&#8221; support some sort of media processing such as
real-time analytics? This is more of a business application on the recordin=
g
server side, which is independent of the session recording protocol itself.=
<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] The main point there was that Recording Protocol mus=
t
support real time streaming for real time analytics</span></i></b><b><i><sp=
an
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS]&nbsp; This Us=
e Case
reads like a requirement and should read more like an example of how it is
used.&nbsp; <o:p></o:p></span></i></b></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#993366'>Is the ability to support &#8220;real-time&#8221; analytics
dependent on the recording protocol?&nbsp; Is there also a need separate
streams for the two channels (what the agent said, what the agent heard) - =
And
do we already have a req&#8217;t for that&#8230;..i.e., req&#8217;t 10 and/=
or
req&#8217;t 11?<o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'text-indent:-18.0pt;mso-list:l0 level1=
 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>Section 6:<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-001 &#8211; what =
does
full-time Recording session mean?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Record all calls</span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend we
explain that in the req&#8217;t; also, is that by DN, or by Agent, or by
Customer, other, or all of the above&#8230;?&nbsp;&nbsp; </span></i></b><b>=
<i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</i></b></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Yes, the requirement is to initiate recording sessio=
n
not by call id but by some fixed identifier as above</span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-003 &#8211; it&#8=
217;s
unclear what is the target of the recording session here since this not
directly related to a single recorded session (perhaps many)<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] To minimize media clipping associated with recording
initiation</span></i></b><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-006 &#8211; I men=
tioned
above that IVR sessions may contain additional IVR application metadata; sh=
ould
this requirement clarify that the mechanism should include application
metadata?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] Yes, we should</span></i></b><span style=3D'font-fam=
ily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-007 &#8211; this
wording of the requirement to me is pointing towards a specific implementat=
ion
of high-availability or fault-tolerance of a recording session.<o:p></o:p><=
/p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-009 &#8211; the w=
ording
here is a bit unclear here. Do you mean the recorded session represents a
multi-party session where there are multiple media streams from multiple
recorded sessions, and the media streams may be mixed by a conference mixer=
?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>&nbsp;</span></=
i></b><span
style=3D'color:#1F497D'>It is more for trading floor environments where mul=
tiple
concurrent calls on the same device (turret) recorded together. It is not
related to summation for conference participants at the same call.&nbsp;&nb=
sp;
&nbsp;&nbsp;</span><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#993366'>[DS] Recommend to state this info in the req&#8217;ts and
provide this example.</span></i></b><o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-013 &#8211; this =
is
only one way of notifying the customer that the call is being recorded but
there are others. I think this requires a bit more explanation of why this =
is
needed. To me this looks like a duplicate of REQ-022&nbsp;&nbsp;&nbsp; <b><=
i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend to =
change
this req&#8217;t from &#8220;Must&#8221; to &#8220;May&#8221;</span></i></b=
>&nbsp;&nbsp;&nbsp;
<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-015 &#8211; what =
does
non-repudiation mean in this context? Isn&#8217;t this the same as REQ-017?=
<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>REQ-015 talk re=
garding
media and REQ-017 regarding SRP SIP Session </span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend to =
define
&#8220;non-repudiation&#8221; and/or provide an example to avoid any
confusion.&nbsp; <o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-017 &#8211; <b><i=
><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS]&nbsp; Why is
non-repudiation of both the SRP and the Media needed?</span></i></b><o:p></=
o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New";
color:#C00000'><span style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "=
Times New Roman"'>&nbsp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>Req &#8211; 022 &nbsp=
;- <b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS]&nbsp; Recomme=
nd to
add &#8220;This MAY be performed by </span><span style=3D'color:#C00000'>in=
jecting
tones into the Recorded Session either from the RS or from the RC.&#8221;
&nbsp;(but could also be done by informing via speech script upon first
answering call) &nbsp;<o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-021 &#8211; what
happens with SRTP if recording client simply forks the media? The recording
server would not be able to decrypt the media unless it gets the private ke=
ys
from the recorded session.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] It is out of scope of this document. </span></i></b>=
<span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-028 &#8211; the w=
ording
redirected looks out of context here; the requirement only applies when
recording client initiates a recording session to an available recorder ser=
ver?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif";
color:#1F497D'>[LeonP] We want to support both directions</span></i></b><sp=
an
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:72.0pt;text-indent:-18.0pt=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]><span dir=3DLTR></span>REQ-029/030 &#8211; T=
hey
look very generic and I don&#8217;t think these should be requirements for =
SRP,
but rather a product requirement on how OA&amp;M works for the recording
client/server.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Henry<o:p></o:p></p>

<div>

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

</div>

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

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New=
 Roman","serif"'><br>
CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
confidential and proprietary information of Alcatel-Lucent and/or its
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ny
liability arising from any party acting, or refraining from acting, on any
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<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New=
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<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New=
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<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New=
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CONFIDENTIALITY NOTICE: This e-mail and any files attached may contain
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affiliated entities. Access by the intended recipient only is authorized. A=
ny
liability arising from any party acting, or refraining from acting, on any
information contained in this e-mail is hereby excluded. If you are not the
intended recipient, please notify the sender immediately, destroy the origi=
nal
transmission and its attachments and do not disclose the contents to any ot=
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From Dave.Smith@genesyslab.com  Mon Feb  1 21:02:23 2010
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Hi Leon, Henry,

I put together some inputs on top of yours for consideration as
well....pls see my inputs below.

=20

Regards,

Dave

=20

From: Leon Portman [mailto:Leon.Portman@nice.com]=20
Sent: Sunday, January 31, 2010 3:03 AM
To: Henry Lum; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: Comments on draft-rehor-dispatch-session-recording-req-00

=20

Hi, Henry.

=20

I will try to answer some of the questions. Please see below.

=20

Leon

=20

From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Henry Lum
Sent: Sunday, January 31, 2010 6:11 AM
To: dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: [dispatch] Comments on
draft-rehor-dispatch-session-recording-req-00

=20

Hi Ken,

=20

I have a bunch of questions and comments on the requirements for SRP
requirements.

=20

-          Section 3 - Recording Client - If the SRP requirement does
not mandate to use SIP as the actual recording protocol, why does the
recording client have to be a SIP user agent or B2BUA?

[LeonP] Actually there is a difference between Recorded session (the
actual call to be recorded[DS]  Is this just the voice - the RTP stream
(the replicated voice stream that is sent to a Logger)?  ) that can be
not SIP based and Recording Session (part of SRP) that is used for
triggering recording and it is SIP based[DS] Is this the control
messaging...for starting/stopping/pause/resume...the messaging (SIP) for
set up of the replicated voice stream/RTP transfer? . So both RC and RS
are SIP UA. =20

-          Section 3 - Recorded session - Is a recorded session has to
be a SIP session?

[LeonP] No, my comment above[DS] I recommend better Terms and/or better
definitions of the Terms "Recorded Session" and "Recording Session"in
the Draft - is it possible to change one or both terms since they sound
so similar.  That is, the IETF Draft definitions are not clear to
me...and the difference between these is not clear (for example, do not
use the terms themselves in the definitions...provide examples.
Recommend also to differentiate these from the Metadata messaging (if
they are or could be different), and to define that as well (as rec'd by
Martin). =20

-          Section 3 - Persistent Recording - does recorded session
exist in this context since it seems like the recording session is
persistent from the endpoint perspective.

[LeonP] No, recorded session are not required to be existing during
initiation of persistent session. Persistent recording session can be
initiated on agent logon for example.

-          Section 4 - use case 5 - if the recording can include
recording for an IVR application (ie. VoiceXML application), does that
mean the call metadata must include application contexts as well?

[LeonP] Yes, like specific script or input results

-          Section 4, use case 7 - the first sentence has a strange word
"&mdash". This use case somehow implies certain architectural design for
a PBX with regards to media forking. Is that necessary from a
requirement perspective?

[LeonP] XML convert mistake. We don't want assume any specific
configurations.

-          Section 4, use case 8 - shall we use a more general
terminology such as multi-party call? Depending on the situation, the
supervisor may be conferenced with the agent and customer so the
recording session needs to define how a multi-party session should be
recorded.

[LeonP] Yes, probably we want to use more generic term then Complex
Call[DS] The term "Multi-Party Call" seems to imply a call/interaction
with more than two users...that is, a conference.  Recommend to keep
"Complex Calls" terminology but define several different cases/examples
- for example, Conferences (Henry's example), Consults (as in use case
8), also could include a Whisper Agent/Coaching example, Barge-In
example, other...  Alternatively, change Use Case 8 to "Consult Call"
and add additional use cases for these other complex call examples. =20

-          Section 4, use case 11, does a recorder "must" support some
sort of media processing such as real-time analytics? This is more of a
business application on the recording server side, which is independent
of the session recording protocol itself.

[LeonP] The main point there was that Recording Protocol must support
real time streaming for real time analytics[DS]  This Use Case reads
like a requirement and should read more like an example of how it is
used. =20

Is the ability to support "real-time" analytics dependent on the
recording protocol?  Is there also a need separate streams for the two
channels (what the agent said, what the agent heard) - And do we already
have a req't for that.....i.e., req't 10 and/or req't 11?

-          Section 6:

o   REQ-001 - what does full-time Recording session mean?

[LeonP] Record all calls[DS] Recommend we explain that in the req't;
also, is that by DN, or by Agent, or by Customer, other, or all of the
above...?  =20

o   REQ-003 - it's unclear what is the target of the recording session
here since this not directly related to a single recorded session
(perhaps many)

[LeonP] To minimize media clipping associated with recording initiation

o   REQ-006 - I mentioned above that IVR sessions may contain additional
IVR application metadata; should this requirement clarify that the
mechanism should include application metadata?

[LeonP] Yes, we should

o   REQ-007 - this wording of the requirement to me is pointing towards
a specific implementation of high-availability or fault-tolerance of a
recording session.

o   REQ-009 - the wording here is a bit unclear here. Do you mean the
recorded session represents a multi-party session where there are
multiple media streams from multiple recorded sessions, and the media
streams may be mixed by a conference mixer?

[LeonP]  It is more for trading floor environments where multiple
concurrent calls on the same device (turret) recorded together. It is
not related to summation for conference participants at the same call.
[DS] Recommend to state this info in the req'ts and provide this
example.

o   REQ-013 - this is only one way of notifying the customer that the
call is being recorded but there are others. I think this requires a bit
more explanation of why this is needed. To me this looks like a
duplicate of REQ-022    [DS] Recommend to change this req't from "Must"
to "May"   =20

o   REQ-015 - what does non-repudiation mean in this context? Isn't this
the same as REQ-017?

[LeonP] REQ-015 talk regarding media and REQ-017 regarding SRP SIP
Session [DS] Recommend to define "non-repudiation" and/or provide an
example to avoid any confusion. =20

o   REQ-017 - [DS]  Why is non-repudiation of both the SRP and the Media
needed?

o   Req - 022  - [DS]  Recommend to add "This MAY be performed by
injecting tones into the Recorded Session either from the RS or from the
RC."  (but could also be done by informing via speech script upon first
answering call) =20

o   REQ-021 - what happens with SRTP if recording client simply forks
the media? The recording server would not be able to decrypt the media
unless it gets the private keys from the recorded session.

[LeonP] It is out of scope of this document.=20

o   REQ-028 - the wording redirected looks out of context here; the
requirement only applies when recording client initiates a recording
session to an available recorder server?

[LeonP] We want to support both directions

o   REQ-029/030 - They look very generic and I don't think these should
be requirements for SRP, but rather a product requirement on how OA&M
works for the recording client/server.

=20

Thanks

Henry

=20

=20


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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'>Hi
Leon, Henry,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'>I
put together some inputs on top of yours for consideration as well&#8230;=
=2Epls
see my inputs below.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'>Regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'>Dave<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Taho=
ma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Leon Portma=
n
[mailto:Leon.Portman@nice.com] <br>
<b>Sent:</b> Sunday, January 31, 2010 3:03 AM<br>
<b>To:</b> Henry Lum; dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> RE: Comments on draft-rehor-dispatch-session-recording-re=
q-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>Hi,
Henry.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>I
will try to answer some of the questions. Please see below.<o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'>Leon<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Taho=
ma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <b>On Behalf=
 Of </b>Henry
Lum<br>
<b>Sent:</b> Sunday, January 31, 2010 6:11 AM<br>
<b>To:</b> dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> [dispatch] Comments on
draft-rehor-dispatch-session-recording-req-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hi Ken,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I have a bunch of questions and comments on the requ=
irements
for SRP requirements.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 &#8211; Recording Client &#8211; If the=
 SRP
requirement does not mandate to use SIP as the actual recording protocol,=
 why
does the recording client have to be a SIP user agent or B2BUA?<o:p></o:p=
></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Actually there is a difference between Recorded se=
ssion
(the actual call to be recorded</span></i></b><b><i><span style=3D'font-f=
amily:
"Arial","sans-serif";color:#993366'>[DS]&nbsp; Is this just the voice - t=
he RTP
stream (the replicated voice stream that is sent to a Logger)?&nbsp; </sp=
an></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'>) that can be no=
t SIP
based and Recording Session (part of SRP) that is used for triggering rec=
ording
and it is SIP based</span></i></b><b><i><span style=3D'font-family:"Arial=
","sans-serif";
color:#993366'>[DS] Is this the control messaging&#8230;for
starting/stopping/pause/resume&#8230;the messaging (SIP) for set up of th=
e
replicated voice stream/RTP transfer? </span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'>. So both RC and=
 RS are
SIP UA.</span></i></b><b><i><span style=3D'font-family:"Arial","sans-seri=
f";
color:#993366'>&nbsp; </span></i></b><span style=3D'font-family:"Arial","=
sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 - Recorded session &#8211; Is a recorde=
d
session has to be a SIP session?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] No, my comment above</span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] I recommend=
 better
Terms and/or better definitions of the Terms &#8220;Record<u>ed</u>
Session&#8221; and &#8220;Record<u>ing</u> Session&#8221;in the Draft &#8=
211; is
it possible to change one or both terms since they sound so similar.&nbsp=
; That
is, the IETF Draft definitions are not clear to me&#8230;and the differen=
ce
between these is not clear (for example, do not use the terms themselves =
in the
definitions&#8230;provide examples.&nbsp; Recommend also to differentiate=
 these
from the Metadata messaging (if they are or could be different), and to d=
efine that
as well (as rec&#8217;d by Martin).&nbsp; </span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 - Persistent Recording &#8211; does rec=
orded
session exist in this context since it seems like the recording session i=
s
persistent from the endpoint perspective.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] No, recorded session are not required to be existi=
ng
during initiation of persistent session. Persistent recording session can=
 be
initiated on agent logon for example.</span></i></b><span style=3D'font-f=
amily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4 &#8211; use case 5 &#8211; if the recor=
ding
can include recording for an IVR application (ie. VoiceXML application), =
does
that mean the call metadata must include application contexts as well?<o:=
p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, like specific script or input results</span><=
/i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 7 &#8211; the first sentence =
has a
strange word &#8220;&amp;mdash&#8221;. This use case somehow implies cert=
ain
architectural design for a PBX with regards to media forking. Is that nec=
essary
from a requirement perspective?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] XML convert mistake. We don&#8217;t want assume an=
y
specific configurations.</span></i></b><span style=3D'font-family:"Arial"=
,"sans-serif";
color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 8 &#8211; shall we use a more=

general terminology such as multi-party call? Depending on the situation,=
 the
supervisor may be conferenced with the agent and customer so the recordin=
g
session needs to define how a multi-party session should be recorded.<o:p=
></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, probably we want to use more generic term the=
n
Complex Call</span></i></b><b><i><span style=3D'font-family:"Arial","sans=
-serif";
color:#993366'>[DS] The term &#8220;Multi-Party Call&#8221; seems to impl=
y a
call/interaction with more than two users&#8230;that is, a conference.&nb=
sp;
Recommend to keep &#8220;Complex Calls&#8221; terminology but define seve=
ral
different cases/examples &#8211; for example, Conferences (Henry&#8217;s
example), Consults (as in use case 8), also could include a Whisper
Agent/Coaching example, Barge-In example, other&#8230;&nbsp; Alternativel=
y,
change Use Case 8 to &#8220;Consult Call&#8221; and add additional use ca=
ses
for these other complex call examples.&nbsp; </span></i></b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4, use case 11, does a recorder
&#8220;must&#8221; support some sort of media processing such as real-tim=
e
analytics? This is more of a business application on the recording server=
 side,
which is independent of the session recording protocol itself.<o:p></o:p>=
</p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] The main point there was that Recording Protocol m=
ust
support real time streaming for real time analytics</span></i></b><b><i><=
span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS]&nbsp; This =
Use Case
reads like a requirement and should read more like an example of how it i=
s used.&nbsp;
<o:p></o:p></span></i></b></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#993366'>Is the ability to support &#8220;real-time&#8221; analytic=
s
dependent on the recording protocol?&nbsp; Is there also a need separate
streams for the two channels (what the agent said, what the agent heard) =
- And
do we already have a req&#8217;t for that&#8230;..i.e., req&#8217;t 10 an=
d/or
req&#8217;t 11?<o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 6:<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-001 &#8211; what does full-time Record=
ing
session mean?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Record all calls</span></i></b><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend w=
e
explain that in the req&#8217;t; also, is that by DN, or by Agent, or by
Customer, other, or all of the above&#8230;?&nbsp;&nbsp; </span></i></b><=
span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-003 &#8211; it&#8217;s unclear what is=
 the
target of the recording session here since this not directly related to a=

single recorded session (perhaps many)<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] To minimize media clipping associated with recordi=
ng
initiation</span></i></b><span style=3D'font-family:"Arial","sans-serif";=

color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-006 &#8211; I mentioned above that IVR=

sessions may contain additional IVR application metadata; should this req=
uirement
clarify that the mechanism should include application metadata?<o:p></o:p=
></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] Yes, we should</span></i></b><span style=3D'font-f=
amily:
"Arial","sans-serif";color:#1F497D'><o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-007 &#8211; this wording of the requir=
ement
to me is pointing towards a specific implementation of high-availability =
or
fault-tolerance of a recording session.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-009 &#8211; the wording here is a bit
unclear here. Do you mean the recorded session represents a multi-party s=
ession
where there are multiple media streams from multiple recorded sessions, a=
nd the
media streams may be mixed by a conference mixer?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>&nbsp;</span>=
</i></b><span
style=3D'color:#1F497D'>It is more for trading floor environments where m=
ultiple
concurrent calls on the same device (turret) recorded together. It is not=

related to summation for conference participants at the same call.&nbsp;&=
nbsp; &nbsp;&nbsp;</span><b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend t=
o state
this info in the req&#8217;ts and provide this example.</span></i></b><o:=
p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-013 &#8211; this is only one way of
notifying the customer that the call is being recorded but there are othe=
rs. I
think this requires a bit more explanation of why this is needed. To me t=
his
looks like a duplicate of REQ-022&nbsp;&nbsp;&nbsp; <b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend t=
o change
this req&#8217;t from &#8220;Must&#8221; to &#8220;May&#8221;</span></i><=
/b>&nbsp;&nbsp;&nbsp;
<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-015 &#8211; what does non-repudiation =
mean
in this context? Isn&#8217;t this the same as REQ-017?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] </span><span style=3D'color:#1F497D'>REQ-015 talk
regarding media and REQ-017 regarding SRP SIP Session </span></i></b><b><=
i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS] Recommend t=
o define
&#8220;non-repudiation&#8221; and/or provide an example to avoid any conf=
usion.&nbsp;
<o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-017 &#8211; <b><i><span style=3D'font-=
family:
"Arial","sans-serif";color:#993366'>[DS]&nbsp; Why is non-repudiation of =
both
the SRP and the Media needed?</span></i></b><o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New";
color:#C00000'><span style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt=
 "Times New Roman"'>&nbsp;&nbsp;
</span></span></span><![endif]>Req &#8211; 022 &nbsp;- <b><i><span
style=3D'font-family:"Arial","sans-serif";color:#993366'>[DS]&nbsp; Recom=
mend to add
&#8220;This MAY be performed by </span></i></b><b><i><span style=3D'color=
:#C00000'>injecting</span></i></b><b><i><span
style=3D'color:#C00000'> tones into the Recorded Session either from the =
RS or
from the RC.&#8221; &nbsp;(but could also be done by informing via speech=

script upon first answering call) &nbsp;<o:p></o:p></span></i></b></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-021 &#8211; what happens with SRTP if
recording client simply forks the media? The recording server would not b=
e able
to decrypt the media unless it gets the private keys from the recorded se=
ssion.<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] It is out of scope of this document. </span></i></=
b><span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-028 &#8211; the wording redirected loo=
ks out
of context here; the requirement only applies when recording client initi=
ates a
recording session to an available recorder server?<o:p></o:p></p>

<p class=3DMsoNormal><b><i><span style=3D'font-family:"Arial","sans-serif=
";
color:#1F497D'>[LeonP] We want to support both directions</span></i></b><=
span
style=3D'font-family:"Arial","sans-serif";color:#1F497D'><o:p></o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-029/030 &#8211; They look very generic=
 and I
don&#8217;t think these should be requirements for SRP, but rather a prod=
uct
requirement on how OA&amp;M works for the recording client/server.<o:p></=
o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Henry<o:p></o:p></p>

<div>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'>&nbsp;<o:p></o:p></span></p>

</div>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'><br>
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From Dave.Smith@genesyslab.com  Tue Feb  2 11:41:55 2010
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From: "Dave Smith" <Dave.Smith@genesyslab.com>
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Cc: Henry Lum <Henry.Lum@genesyslab.com>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Leon Portman <Leon.Portman@nice.com>, "Ken Rehor \(krehor\)" <krehor@cisco.com>
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Ken,=20

=20

Just wanted to say (before I forgot) that your idea about using the
diagram to clarify the def'ns is a good one I think....I just recommend
to link/match the different parts of the diagram (Call Metadata,
Recording Control, Recorded Media) to the definitions.....to
describe/show what parts in the diagram are related to what
def'ns.....e.g., where is SRP in the diagram, does this Draft cover the
different aspects of Recording Control (and if so, which ones &
where....we discussed another aspect of control in today's mtg), etc...

I think that really would help.

=20

Regards,

Dave=20

=20

=20


				=09
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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Ken=
, <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'><o:=
p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Jus=
t wanted
to say (before I forgot) that your idea about using the diagram to clarif=
y the
def&#8217;ns is a good one I think&#8230;.I just recommend to link/match =
the different
parts of the diagram (Call Metadata, Recording Control, Recorded Media) t=
o the
definitions&#8230;..to describe/show what parts in the diagram are relate=
d to
what def&#8217;ns&#8230;..e.g., where is SRP in the diagram, does this Dr=
aft
cover the different aspects of Recording Control (and if so, which ones &=
amp;
where&#8230;.we discussed another aspect of control in today&#8217;s mtg)=
, etc&#8230;<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>I t=
hink that
really would help.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'><o:=
p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Reg=
ards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'>Dav=
e <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif"'><o:=
p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

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mail is hereby excluded. If you are not the intended recipient, please no=
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tachments and do not disclose the contents to any other person, use it fo=
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From scottlawrenc@avaya.com  Tue Feb  2 10:31:00 2010
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On Thu, 2010-01-28 at 13:45 -0500, Dale Worley wrote:
> On Sun, 2009-12-13 at 07:41 -0500, Scott Lawrence wrote:
> > The draft doesn't go into much detail on the specifics of the advantages
> > over the method described in 5359 (and other methods in use); it could
> > probably use some expansion in that area because the advantages are
> > real.
> 
> The current text is:
> 
>    This technique for providing music-on-hold has advantages over other
>    methods now in use:
> 
>    1.  The original dialog is not transferred to another UA, so the
>        "remote endpoint URI" displayed by the remote endpoint's user
>        interface and dialog event package[dialog-event] does not change
>        during the call.[service-examples]
> 
>    2.  Compared to [service-examples], this method does not require use
>        of an out-of-dialog REFER, which is not otherwise used much in
>        SIP and may not be handled correctly by authorization mechanisms.
> 
>    3.  The music-on-hold media are sent directly from the music-on-hold
>        source to the remote UA, rather than being relayed through the
>        executing UA.
> 
>    4.  The remote UA sees, in the incoming SDP, the address/port that
>        the MOH source will send MOH media from, thus allowing it to
>        render the media, even if it is filtering incoming media based on
>        originating address as a media security measure.
> 
>    5.  The technique requires relatively simple manipulation of SDP, and
>        in particular: (1) does not require a SIP element to modify
>        unrelated SDP to be acceptable to be sent within an already
>        established sequence of SDP (a problem with
>        [service-examples-11]), and (2) does not require converting an
>        SDP answer into an SDP offer (which was a problem with the -00
>        version of this document, as well as with [service-examples-11]).
> 
>    6.  It complies with the payload type number rules.[offer-answer]
> 
> What do you think needs to be added or expanded?  In regard to RFC 5359
> specifically, items 1 and 2 call out the improved behavior.

Since the goal is to update 5359, I think that there should be an
explicit section that says, in effect: "Here is what is wrong with how
it is done in 5359, and why this proposal is better".  For each of the
above, rather than saying "this way doesn't do xyz", say
"service-examples-11 does xyz, and that is a problem because...".

Some specifics that you leave unsaid:

What's wrong with changing the remote party id on a call just because
you get put on hold?

In what way is sending an out-of-dialog REFER unreliable?  When wouldn't
it work?  (neither From headers nor Contact addresses are always
routable).

What's wrong with relaying media through the UA executing the hold?  

What's wrong with manipulating SDP?  

What's the big deal with payload number rules?  What problem does that
cause?




From Dave.Smith@genesyslab.com  Tue Feb  2 14:33:25 2010
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Raj,

Regarding this input:

[rj] The SRP, which manages how the recording media gets delivered to
the recorders, is based on SIP (SRP is a set of extensions to the SIP
core protocol). The sessions that SRP manages are called "Recording
Sessions" and they are SIP sessions. Therefore, RC and RS are, by
definition, SIP UAs. The sessions that get actually recorded (e.g.
communication between a call center agent and an external caller) may or
may not be SIP (this depends on the kind of the PBX phone the agent).=20

=20

Can you please provide a few examples of the different types of
protocols that might be possible/practical for the sessions that get
actually recorded (which are dependent on the kind of PBX phone used)?

=20

Regards,

Dave

=20

=20

From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]=20
Sent: Monday, February 01, 2010 3:28 PM
To: Henry Lum; Leon Portman; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: [dispatch] Comments
ondraft-rehor-dispatch-session-recording-req-00

=20

Henry,

=20

Thanks for your comments. Inline:

=20

-          Section 3 - Recording Client - If the SRP requirement does
not mandate to use SIP as the actual recording protocol, why does the
recording client have to be a SIP user agent or B2BUA?

[LeonP] Actually there is a difference between Recorded session (the
actual call to be recorded) that can be not SIP based and Recording
Session (part of SRP) that is used for triggering recording and it is
SIP based. So both RC and RS are SIP UA.

[[hlum]] I now understand that the recorded session does not have to be
SIP based, but the wording in the definition of SRP itself mentions that
it may be SIP; then why does RC and RS have to be a SIP UA if SRP does
not have to be SIP?

=20

[rj] The SRP, which manages how the recording media gets delivered to
the recorders, is based on SIP (SRP is a set of extensions to the SIP
core protocol). The sessions that SRP manages are called "Recording
Sessions" and they are SIP sessions. Therefore, RC and RS are, by
definition, SIP UAs. The sessions that get actually recorded (e.g.
communication between a call center agent and an external caller) may or
may not be SIP (this depends on the kind of the PBX phone the agent).=20

=20

=20

-          Section 4 - use case 5 - if the recording can include
recording for an IVR application (ie. VoiceXML application), does that
mean the call metadata must include application contexts as well?

[LeonP] Yes, like specific script or input results

[[hlum]] I assume what kind of call metadata and how it is delivered is
outside of the scope of this document?

=20

[rj] Correct.

=20

o   REQ-003 - it's unclear what is the target of the recording session
here since this not directly related to a single recorded session
(perhaps many)

[LeonP] To minimize media clipping associated with recording initiation

[[hlum]] You actually answered my question above about what a persistent
recording mean and why it's useful, but what is unclear to me that is
whether there is something called a persistent recording session that
targets something like an agent device.

=20

[rj] I guess, Persistent Recording is a bit confusing term (we've
received other feedback on this as well). If I can use the term
always-on recording from the wing-srtp draft then these sessions get
established when the agent logs on and torn off when the agent logs off.
Codecs such as G.729 and G.711 with VAD are used to ensure that silence
is not recorded (for instance, when the agent isn't on a call).

=20

o   REQ-009 - the wording here is a bit unclear here. Do you mean the
recorded session represents a multi-party session where there are
multiple media streams from multiple recorded sessions, and the media
streams may be mixed by a conference mixer?

[LeonP]  It is more for trading floor environments where multiple
concurrent calls on the same device (turret) recorded together. It is
not related to summation for conference participants at the same call.

[[hlum]] I'm not familiar with trading floor environments so I may not
understand the intent here. If you have multiple concurrent calls then
wouldn't you have multiple recording sessions to describe each call
individually? Does the endpoint device perform mixing function for
multiple calls so that's why you need to club them together in a single
recording session?

=20

[rj] Due to bandwidth and other cost burdens, the customer demand is
that you mix multiple _recorded sessions_ in a single _recording
session_.

=20

=20

o   REQ-028 - the wording redirected looks out of context here; the
requirement only applies when recording client initiates a recording
session to an available recorder server?

[LeonP] We want to support both directions

[[hlum]] Okay, then I think we should re-word to mention something about
availability in general.  (eg. "A request for a new recording session
must reach an available recording client or server").=20

=20

[rj] The example test you quoted doesn't reflect the actual intent here.
The reason for supporting both directions is dependent on where the
policy to record or not resides - RC or RS. The INVITE always comes from
either of these two places.

=20

Thanks,

Raj Jain

=20

=20

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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'>Raj,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'>Regarding this input:<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;color:#1F497D'>[rj] =
The SRP,
which manages how the recording media gets delivered to the recorders, is=
 based
on SIP (SRP is a set of extensions to the SIP core protocol). The session=
s that
SRP manages are called &#8220;Recording Sessions&#8221; and they are SIP
sessions. Therefore, RC and RS are, by definition, SIP UAs. The sessions =
that
get actually recorded (e.g. communication between a call center agent and=
 an
external caller) may or may not be SIP (this depends on the kind of the P=
BX
phone the agent). <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'>Can you please provide a few examples of the different types=
 of
protocols that might be possible/practical for </span><span style=3D'font=
-size:
12.0pt;color:#1F497D'>the sessions that get actually recorded</span><span=

style=3D'font-size:12.0pt;font-family:"Arial","sans-serif";color:black'> =
(which
are dependent on the kind of PBX phone used)?<o:p></o:p></span></p>

<p class=3DMsoNormal><b><span style=3D'font-size:12.0pt;font-family:"Aria=
l","sans-serif";
color:black'><o:p>&nbsp;</o:p></span></b></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'>Regards,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Arial",=
"sans-serif";
color:black'>Dave<o:p></o:p></span></p>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Aria=
l","sans-serif";
color:black'><o:p>&nbsp;</o:p></span></b></p>

<p class=3DMsoNormal><span style=3D'font-family:"Arial","sans-serif";colo=
r:#993366'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0i=
n 0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Taho=
ma","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Jain, Rajni=
sh
[mailto:Rajnish.Jain@ipc.com] <br>
<b>Sent:</b> Monday, February 01, 2010 3:28 PM<br>
<b>To:</b> Henry Lum; Leon Portman; dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> RE: [dispatch] Comments
ondraft-rehor-dispatch-session-recording-req-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Henry,<o:p></o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks for your commen=
ts. Inline:<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0i=
n 4.0pt'>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 &#8211; Recording Client &#8211; If the=
 SRP
requirement does not mandate to use SIP as the actual recording protocol,=
 why
does the recording client have to be a SIP user agent or B2BUA?<o:p></o:p=
></p>

<p class=3DMsoNormal><b><i>[LeonP] Actually there is a difference between=

Recorded session (the actual call to be recorded) that can be not SIP bas=
ed and
Recording Session (part of SRP) that is used for triggering recording and=
 it is
SIP based. So both RC and RS are SIP UA.<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I now understand=
 that the
recorded session does not have to be SIP based, but the wording in the
definition of SRP itself mentions that it may be SIP; then why does RC an=
d RS
have to be a SIP UA if SRP does not have to be SIP?<o:p></o:p></span></p>=


<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] The SRP, which ma=
nages how
the recording media gets delivered to the recorders, is based on SIP (SRP=
 is a
set of extensions to the SIP core protocol). The sessions that SRP manage=
s are
called &#8220;Recording Sessions&#8221; and they are SIP sessions. Theref=
ore,
RC and RS are, by definition, SIP UAs. The sessions that get actually rec=
orded
(e.g. communication between a call center agent and an external caller) m=
ay or
may not be SIP (this depends on the kind of the PBX phone the agent). <o:=
p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level=
1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4 &#8211; use case 5 &#8211; if the recor=
ding
can include recording for an IVR application (ie. VoiceXML application), =
does that
mean the call metadata must include application contexts as well?<o:p></o=
:p></p>

<p class=3DMsoNormal><b><i>[LeonP] Yes, like specific script or input res=
ults<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I assume what ki=
nd of call
metadata and how it is delivered is outside of the scope of this document=
?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] Correct.<o:p></o:=
p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-003 &#8211; it&#8217;s unclear what is=
 the
target of the recording session here since this not directly related to a=

single recorded session (perhaps many)<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] To minimize media clipping associated =
with
recording initiation<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] You actually ans=
wered my
question above about what a persistent recording mean and why it&#8217;s
useful, but what is unclear to me that is whether there is something call=
ed a
persistent recording session that targets something like an agent device.=
<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] I guess, Persiste=
nt
Recording is a bit confusing term (we&#8217;ve received other feedback on=
 this
as well). If I can use the term always-on recording from the wing-srtp dr=
aft
then these sessions get established when the agent logs on and torn off w=
hen
the agent logs off. Codecs such as G.729 and G.711 with VAD are used to e=
nsure
that silence is not recorded (for instance, when the agent isn&#8217;t on=
 a
call).<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-009 &#8211; the wording here is a bit
unclear here. Do you mean the recorded session represents a multi-party s=
ession
where there are multiple media streams from multiple recorded sessions, a=
nd the
media streams may be mixed by a conference mixer?<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] <span style=3D'color:#1F497D'>&nbsp;</=
span>It is
more for trading floor environments where multiple concurrent calls on th=
e same
device (turret) recorded together. It is not related to summation for
conference participants at the same call.<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I&#8217;m not fa=
miliar
with trading floor environments so I may not understand the intent here. =
If you
have multiple concurrent calls then wouldn&#8217;t you have multiple reco=
rding
sessions to describe each call individually? Does the endpoint device per=
form
mixing function for multiple calls so that&#8217;s why you need to club t=
hem
together in a single recording session?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] Due to bandwidth =
and other
cost burdens, the customer demand is that you mix multiple _<i>recorded
sessions</i>_ in a single _<i>recording session</i>_.<o:p></o:p></span></=
p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in=
;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:=
"Courier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&=
nbsp;&nbsp;
</span></span></span><![endif]>REQ-028 &#8211; the wording redirected loo=
ks out
of context here; the requirement only applies when recording client initi=
ates a
recording session to an available recorder server?<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] We want to support both directions<o:p=
></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] Okay, then I thi=
nk we
should re-word to mention something about availability in general.&nbsp; =
(eg.
&#8220;A request for a new recording session must reach an available reco=
rding
client or server&#8221;).</span><span style=3D'color:#1F497D'> </span><o:=
p></o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] The example test =
you quoted
doesn&#8217;t reflect the actual intent here. The reason for supporting b=
oth
directions is dependent on where the policy to record or not resides &#82=
11; RC
or RS. The INVITE always comes from either of these two places.<o:p></o:p=
></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks,<o:p></o:p></sp=
an></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Raj Jain<o:p></o:p></s=
pan></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></spa=
n></p>

</div>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times N=
ew Roman","serif"'><o:p>&nbsp;</o:p></span></p>

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

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

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To: Dave Smith <Dave.Smith@genesyslab.com>, Henry Lum <Henry.Lum@genesyslab.com>, Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Tue, 2 Feb 2010 18:42:55 -0500
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Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Dave,

Inline:

From: Dave Smith [mailto:Dave.Smith@genesyslab.com]
Sent: Tuesday, February 02, 2010 5:33 PM
To: Jain, Rajnish; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Raj,
Regarding this input:
[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).

Can you please provide a few examples of the different types of protocols t=
hat might be possible/practical for the sessions that get actually recorded=
 (which are dependent on the kind of PBX phone used)?

[rj] This can range anywhere from proprietary VoIP media and signaling prot=
ocols to things such as SCCP, UNISTIM, H.323, TDM and analog phones. Most s=
uch phones are not SIP UAs. Some other entity (such as a SIP-enabled mixer)=
 may provide the RC media forking point functionality in these cases.

Thanks,
Raj

From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
Sent: Monday, February 01, 2010 3:28 PM
To: Henry Lum; Leon Portman; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Henry,

Thanks for your comments. Inline:


-        Section 3 - Recording Client - If the SRP requirement does not man=
date to use SIP as the actual recording protocol, why does the recording cl=
ient have to be a SIP user agent or B2BUA?
[LeonP] Actually there is a difference between Recorded session (the actual=
 call to be recorded) that can be not SIP based and Recording Session (part=
 of SRP) that is used for triggering recording and it is SIP based. So both=
 RC and RS are SIP UA.
[[hlum]] I now understand that the recorded session does not have to be SIP=
 based, but the wording in the definition of SRP itself mentions that it ma=
y be SIP; then why does RC and RS have to be a SIP UA if SRP does not have =
to be SIP?

[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).



-        Section 4 - use case 5 - if the recording can include recording fo=
r an IVR application (ie. VoiceXML application), does that mean the call me=
tadata must include application contexts as well?
[LeonP] Yes, like specific script or input results
[[hlum]] I assume what kind of call metadata and how it is delivered is out=
side of the scope of this document?

[rj] Correct.


o   REQ-003 - it's unclear what is the target of the recording session here=
 since this not directly related to a single recorded session (perhaps many=
)
[LeonP] To minimize media clipping associated with recording initiation
[[hlum]] You actually answered my question above about what a persistent re=
cording mean and why it's useful, but what is unclear to me that is whether=
 there is something called a persistent recording session that targets some=
thing like an agent device.

[rj] I guess, Persistent Recording is a bit confusing term (we've received =
other feedback on this as well). If I can use the term always-on recording =
from the wing-srtp draft then these sessions get established when the agent=
 logs on and torn off when the agent logs off. Codecs such as G.729 and G.7=
11 with VAD are used to ensure that silence is not recorded (for instance, =
when the agent isn't on a call).


o   REQ-009 - the wording here is a bit unclear here. Do you mean the recor=
ded session represents a multi-party session where there are multiple media=
 streams from multiple recorded sessions, and the media streams may be mixe=
d by a conference mixer?
[LeonP]  It is more for trading floor environments where multiple concurren=
t calls on the same device (turret) recorded together. It is not related to=
 summation for conference participants at the same call.
[[hlum]] I'm not familiar with trading floor environments so I may not unde=
rstand the intent here. If you have multiple concurrent calls then wouldn't=
 you have multiple recording sessions to describe each call individually? D=
oes the endpoint device perform mixing function for multiple calls so that'=
s why you need to club them together in a single recording session?

[rj] Due to bandwidth and other cost burdens, the customer demand is that y=
ou mix multiple _recorded sessions_ in a single _recording session_.



o   REQ-028 - the wording redirected looks out of context here; the require=
ment only applies when recording client initiates a recording session to an=
 available recorder server?
[LeonP] We want to support both directions
[[hlum]] Okay, then I think we should re-word to mention something about av=
ailability in general.  (eg. "A request for a new recording session must re=
ach an available recording client or server").

[rj] The example test you quoted doesn't reflect the actual intent here. Th=
e reason for supporting both directions is dependent on where the policy to=
 record or not resides - RC or RS. The INVITE always comes from either of t=
hese two places.

Thanks,
Raj Jain


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<div class=3DSection1>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Dave,<o:p></o:p></span><=
/p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Inline:<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Dave Smith
[mailto:Dave.Smith@genesyslab.com] <br>
<b>Sent:</b> Tuesday, February 02, 2010 5:33 PM<br>
<b>To:</b> Jain, Rajnish; Henry Lum; Leon Portman; dispatch@ietf.org<br>
<b>Cc:</b> krehor@cisco.com<br>
<b>Subject:</b> RE: [dispatch] Comments
ondraft-rehor-dispatch-session-recording-req-00<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Raj,<o:p></o:p></p>

<p class=3DMsoNormal>Regarding this input:<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'font-size:12.0pt;color:#1F497D'>[rj] Th=
e SRP,
which manages how the recording media gets delivered to the recorders, is b=
ased
on SIP (SRP is a set of extensions to the SIP core protocol). The sessions =
that
SRP manages are called &#8220;Recording Sessions&#8221; and they are SIP
sessions. Therefore, RC and RS are, by definition, SIP UAs. The sessions th=
at
get actually recorded (e.g. communication between a call center agent and a=
n
external caller) may or may not be SIP (this depends on the kind of the PBX
phone the agent). <o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Can you please provide a few examples of the different=
 types
of protocols that might be possible/practical for <span style=3D'font-size:=
12.0pt;
color:#1F497D'>the sessions that get actually recorded</span> (which are
dependent on the kind of PBX phone used)?<o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] This can range anyw=
here
from proprietary VoIP media and signaling protocols to things such as SCCP,=
 UNISTIM,
H.323, TDM and analog phones. Most such phones are not SIP UAs. Some other
entity (such as a SIP-enabled mixer) may provide the RC media forking point=
 functionality
in these cases. <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks,<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Raj<o:p></o:p></span></p=
>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b>From:</b> Jain, Rajnish [mailto:Rajnish.Jain@ipc.co=
m] <br>
<b>Sent:</b> Monday, February 01, 2010 3:28 PM<br>
<b>To:</b> Henry Lum; Leon Portman; dispatch@ietf.org<br>
<b>Cc:</b> Dave Smith; krehor@cisco.com<br>
<b>Subject:</b> RE: [dispatch] Comments
ondraft-rehor-dispatch-session-recording-req-00<o:p></o:p></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Henry,<o:p></o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks for your comments=
.
Inline:<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt'>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level1 =
lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 3 &#8211; Recording Client &#8211; If the S=
RP
requirement does not mandate to use SIP as the actual recording protocol, w=
hy
does the recording client have to be a SIP user agent or B2BUA?<o:p></o:p><=
/p>

<p class=3DMsoNormal><b><i>[LeonP] Actually there is a difference between
Recorded session (the actual call to be recorded) that can be not SIP based=
 and
Recording Session (part of SRP) that is used for triggering recording and i=
t is
SIP based. So both RC and RS are SIP UA.<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I now understand t=
hat the
recorded session does not have to be SIP based, but the wording in the
definition of SRP itself mentions that it may be SIP; then why does RC and =
RS have
to be a SIP UA if SRP does not have to be SIP?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] The SRP, which mana=
ges how
the recording media gets delivered to the recorders, is based on SIP (SRP i=
s a
set of extensions to the SIP core protocol). The sessions that SRP manages =
are
called &#8220;Recording Sessions&#8221; and they are SIP sessions. Therefor=
e,
RC and RS are, by definition, SIP UAs. The sessions that get actually recor=
ded
(e.g. communication between a call center agent and an external caller) may=
 or
may not be SIP (this depends on the kind of the PBX phone the agent). <o:p>=
</o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 level1 =
lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Section 4 &#8211; use case 5 &#8211; if the recordi=
ng
can include recording for an IVR application (ie. VoiceXML application), do=
es
that mean the call metadata must include application contexts as well?<o:p>=
</o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] Yes, like specific script or input resul=
ts<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I assume what kind=
 of call
metadata and how it is delivered is outside of the scope of this document?<=
o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] Correct.<o:p></o:p>=
</span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]>REQ-003 &#8211; it&#8217;s unclear what is t=
he
target of the recording session here since this not directly related to a
single recorded session (perhaps many)<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] To minimize media clipping associated wi=
th
recording initiation<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] You actually answe=
red my
question above about what a persistent recording mean and why it&#8217;s us=
eful,
but what is unclear to me that is whether there is something called a
persistent recording session that targets something like an agent device.<o=
:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] I guess, Persistent
Recording is a bit confusing term (we&#8217;ve received other feedback on t=
his
as well). If I can use the term always-on recording from the wing-srtp draf=
t
then these sessions get established when the agent logs on and torn off whe=
n
the agent logs off. Codecs such as G.729 and G.711 with VAD are used to ens=
ure
that silence is not recorded (for instance, when the agent isn&#8217;t on a
call).<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]>REQ-009 &#8211; the wording here is a bit
unclear here. Do you mean the recorded session represents a multi-party ses=
sion
where there are multiple media streams from multiple recorded sessions, and=
 the
media streams may be mixed by a conference mixer?<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] <span style=3D'color:#1F497D'>&nbsp;</sp=
an>It is
more for trading floor environments where multiple concurrent calls on the =
same
device (turret) recorded together. It is not related to summation for
conference participants at the same call.<o:p></o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] I&#8217;m not fami=
liar
with trading floor environments so I may not understand the intent here. If=
 you
have multiple concurrent calls then wouldn&#8217;t you have multiple record=
ing
sessions to describe each call individually? Does the endpoint device perfo=
rm
mixing function for multiple calls so that&#8217;s why you need to club the=
m
together in a single recording session?<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] Due to bandwidth an=
d other
cost burdens, the customer demand is that you mix multiple _<i>recorded
sessions</i>_ in a single _<i>recording session</i>_.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoListParagraph style=3D'margin-left:1.0in;text-indent:-.25in;
mso-list:l0 level2 lfo2'><![if !supportLists]><span style=3D'font-family:"C=
ourier New"'><span
style=3D'mso-list:Ignore'>o<span style=3D'font:7.0pt "Times New Roman"'>&nb=
sp;&nbsp;
</span></span></span><![endif]>REQ-028 &#8211; the wording redirected looks=
 out
of context here; the requirement only applies when recording client initiat=
es a
recording session to an available recorder server?<o:p></o:p></p>

<p class=3DMsoNormal><b><i>[LeonP] We want to support both directions<o:p><=
/o:p></i></b></p>

<p class=3DMsoNormal><span style=3D'color:blue'>[[hlum]] Okay, then I think=
 we
should re-word to mention something about availability in general.&nbsp; (e=
g.
&#8220;A request for a new recording session must reach an available record=
ing
client or server&#8221;).</span><span style=3D'color:#1F497D'> </span><o:p>=
</o:p></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>[rj] The example test yo=
u quoted
doesn&#8217;t reflect the actual intent here. The reason for supporting bot=
h
directions is dependent on where the policy to record or not resides &#8211=
; RC
or RS. The INVITE always comes from either of these two places.<o:p></o:p><=
/span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Thanks,<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Raj Jain<o:p></o:p></spa=
n></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span>=
</p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

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

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From zhouzp@huawei.com  Wed Feb  3 01:51:59 2010
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Date: Wed, 03 Feb 2010 17:51:58 +0800
From: zhouzhipeng 54906 <zhouzp@huawei.com>
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Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Dear all,
I have got much valuable information from related emails.
As I cited as following,

> [rj] This can range anywhere from proprietary VoIP media and 
> signaling protocols to things such as SCCP, UNISTIM, H.323, TDM 
> and analog phones. Most such phones are not SIP UAs. Some other 
> entity (such as a SIP-enabled mixer) may provide the RC media 
> forking point functionality in these cases.

there have been many years for the application of recording. 
So, may the author give us a brief summary what have been advanced based on current recording solutions, for example the call center architecture. So we can easily catch the core point of this document.

Another point is: does the diagram need to add the interface between UA to the RS for the search or query functions.

Thanks very much.
Zhipeng


From henry.sinnreich@gmail.com  Wed Feb  3 10:17:17 2010
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Date: Wed, 03 Feb 2010 12:17:52 -0600
From: Henry Sinnreich <henry.sinnreich@gmail.com>
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We have published an I-D on "SIP APIs for Web Communications" and would
appreciate any comments and input.

Here is the link and the announcement:

http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


 Title  : SIP APIs for Communications on the Web
 Author(s) : H. Sinnreich, A. Johnston
 Filename : draft-sinnreich-sip-web-apis-00.txt
 Pages  : 19
 Date  : 2010-1-19
 
   Interactive communications are becoming part of rich Internet
   applications (RIA). This can be accomplished entirely by using Web
   technology and developing it as an Internet standard. Voice over IP
   (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
   can be preserved in rich multimedia communications and applications
   on the web.  Hyper Text Transport Protocol (HTTP) is used for control
   and signaling data and RTP for interactive media transport
   respectively. No other network application protocols are required.
   Web, fixed and mobile device application development tools can thus
   be used to include components and widgets for Internet presence,
   Instant Messaging (IM), voice and video.  This opens Internet
   communications, including voice to application developers and will
   also enhance the user experience with real-time Web and mobile
   communications.  Usage scenarios include service providers, private
   IP networks, device application developers and media companies
   providing a mix of integrated communications, applications and media.
   This memo argues for the development and standardization of
   Application Programming Interfaces (APIs) to allow the benefits of
   SIP to be used by Rich Internet Applications (RIA), for fixed and
   mobile devices.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt

------ End of Forwarded Message

Thanks in advance for your comments.

Henry



From peter.musgrave@magorcorp.com  Wed Feb  3 12:20:33 2010
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--0016e6d46d7da31f88047eb7f542
Content-Type: text/plain; charset=ISO-8859-1

Hi Henry,

(Based on a one-pass read through)

I like the ideas expressed in this draft. The summary of SIP 2.0 is on
target and the definition of a web API is a good idea. It is something I can
use.

Moving to HIP is a good idea.

Regards,

Peter Musgrave

On Wed, Feb 3, 2010 at 1:17 PM, Henry Sinnreich
<henry.sinnreich@gmail.com>wrote:

> We have published an I-D on "SIP APIs for Web Communications" and would
> appreciate any comments and input.
>
> Here is the link and the announcement:
>
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>
>  Title  : SIP APIs for Communications on the Web
>  Author(s) : H. Sinnreich, A. Johnston
>  Filename : draft-sinnreich-sip-web-apis-00.txt
>  Pages  : 19
>  Date  : 2010-1-19
>
>   Interactive communications are becoming part of rich Internet
>   applications (RIA). This can be accomplished entirely by using Web
>   technology and developing it as an Internet standard. Voice over IP
>   (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>   can be preserved in rich multimedia communications and applications
>   on the web.  Hyper Text Transport Protocol (HTTP) is used for control
>   and signaling data and RTP for interactive media transport
>   respectively. No other network application protocols are required.
>   Web, fixed and mobile device application development tools can thus
>   be used to include components and widgets for Internet presence,
>   Instant Messaging (IM), voice and video.  This opens Internet
>   communications, including voice to application developers and will
>   also enhance the user experience with real-time Web and mobile
>   communications.  Usage scenarios include service providers, private
>   IP networks, device application developers and media companies
>   providing a mix of integrated communications, applications and media.
>   This memo argues for the development and standardization of
>   Application Programming Interfaces (APIs) to allow the benefits of
>   SIP to be used by Rich Internet Applications (RIA), for fixed and
>   mobile devices.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt
>
> ------ End of Forwarded Message
>
> Thanks in advance for your comments.
>
> Henry
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

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Hi Henry,=A0<div><br></div><div>(Based on a one-pass read through)</div><di=
v><br></div><div>I like the ideas expressed in this draft. The summary of S=
IP 2.0 is on target and the definition of a web API is a good idea. It is s=
omething I can use.</div>
<div><br></div><div>Moving to HIP is a good idea.</div><div><br></div><div>=
Regards,=A0</div><div><br></div><div>Peter Musgrave<br><br><div class=3D"gm=
ail_quote">On Wed, Feb 3, 2010 at 1:17 PM, Henry Sinnreich <span dir=3D"ltr=
">&lt;<a href=3D"mailto:henry.sinnreich@gmail.com">henry.sinnreich@gmail.co=
m</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">We have published an I-D on &quot;SIP APIs =
for Web Communications&quot; and would<br>
appreciate any comments and input.<br>
<br>
Here is the link and the announcement:<br>
<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis=
-00.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-sinnre=
ich-sip-web-apis-00.txt</a><br>
<br>
A New Internet-Draft is available from the on-line Internet-Drafts<br>
directories.<br>
<br>
<br>
=A0Title =A0: SIP APIs for Communications on the Web<br>
=A0Author(s) : H. Sinnreich, A. Johnston<br>
=A0Filename : draft-sinnreich-sip-web-apis-00.txt<br>
=A0Pages =A0: 19<br>
=A0Date =A0: 2010-1-19<br>
<br>
 =A0 Interactive communications are becoming part of rich Internet<br>
 =A0 applications (RIA). This can be accomplished entirely by using Web<br>
 =A0 technology and developing it as an Internet standard. Voice over IP<br=
>
 =A0 (VoIP) features developed for Session Initiation Protocol (SIP 2.0)<br=
>
 =A0 can be preserved in rich multimedia communications and applications<br=
>
 =A0 on the web. =A0Hyper Text Transport Protocol (HTTP) is used for contro=
l<br>
 =A0 and signaling data and RTP for interactive media transport<br>
 =A0 respectively. No other network application protocols are required.<br>
 =A0 Web, fixed and mobile device application development tools can thus<br=
>
 =A0 be used to include components and widgets for Internet presence,<br>
 =A0 Instant Messaging (IM), voice and video. =A0This opens Internet<br>
 =A0 communications, including voice to application developers and will<br>
 =A0 also enhance the user experience with real-time Web and mobile<br>
 =A0 communications. =A0Usage scenarios include service providers, private<=
br>
 =A0 IP networks, device application developers and media companies<br>
 =A0 providing a mix of integrated communications, applications and media.<=
br>
 =A0 This memo argues for the development and standardization of<br>
 =A0 Application Programming Interfaces (APIs) to allow the benefits of<br>
 =A0 SIP to be used by Rich Internet Applications (RIA), for fixed and<br>
 =A0 mobile devices.<br>
<br>
A URL for this Internet-Draft is:<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis=
-00.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-sinnre=
ich-sip-web-apis-00.txt</a><br>
<br>
------ End of Forwarded Message<br>
<br>
Thanks in advance for your comments.<br>
<br>
Henry<br>
<br>
<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</blockquote></div><br></div>

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From bernard_aboba@hotmail.com  Wed Feb  3 12:55:16 2010
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There is considerable work already underway to address some of the issues
raised in this draft.  For example:

1. Work in W3C HTML5 WGs to add native support for audio and video. 
2. Work in IETF "HyBi" WG to make HTTP more suitable for realtime signaling
3. Work in XSF to provide enhanced telephony support within XMPP (e.g.
Jingle ICE, Jingle Transfer, etc.)
4. Work in IETF XMPP WG to revise RFC 3920/3921, address SIP/XMPP interop,
etc. 

Given this, my reading of the document raised several questions:

1.  What problems exist with existing XML over HTTP approaches to signaling
(e.g. XMPP over HTTP w/Jingle)? 
2. What additional work beyond what is already under discussion is necessary
to complete the vision you describe? 

==========================================================
Henry said:

"We have published an I-D on "SIP APIs for Web Communications" and would
appreciate any comments and input.

Here is the link and the announcement:

http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


 Title  : SIP APIs for Communications on the Web
 Author(s) : H. Sinnreich, A. Johnston
 Filename : draft-sinnreich-sip-web-apis-00.txt
 Pages  : 19
 Date  : 2010-1-19
 
   Interactive communications are becoming part of rich Internet
   applications (RIA). This can be accomplished entirely by using Web
   technology and developing it as an Internet standard. Voice over IP
   (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
   can be preserved in rich multimedia communications and applications
   on the web.  Hyper Text Transport Protocol (HTTP) is used for control
   and signaling data and RTP for interactive media transport
   respectively. No other network application protocols are required.
   Web, fixed and mobile device application development tools can thus
   be used to include components and widgets for Internet presence,
   Instant Messaging (IM), voice and video.  This opens Internet
   communications, including voice to application developers and will
   also enhance the user experience with real-time Web and mobile
   communications.  Usage scenarios include service providers, private
   IP networks, device application developers and media companies
   providing a mix of integrated communications, applications and media.
   This memo argues for the development and standardization of
   Application Programming Interfaces (APIs) to allow the benefits of
   SIP to be used by Rich Internet Applications (RIA), for fixed and
   mobile devices.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt"




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From: Henry Sinnreich <henry.sinnreich@gmail.com>
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Hi Bernard,

Thanks for drilling down on some of the issues.

> 1. Work in W3C HTML5 WGs to add native support for audio and video.
> 2. Work in IETF "HyBi" WG to make HTTP more suitable for realtime signaling

Both confirm we believe our approach, though no SIP-specific functionality
for rendezvous and session initiation  is envisaged, as is the focus in our
I-D. Even less on how to port the very rich SIP voice functionality to
HTML/HTTP.

Such items as support for the SPEEX codec (in the example) in
http://dev.w3.org/html5/spec/Overview.html and for bidirectional HTTP (as
mentioned in our I-D), at http://trac.tools.ietf.org/bof/trac/wiki/HyBi are
really heartening.

> 1.  What problems exist with existing XML over HTTP approaches to signaling
> (e.g. XMPP over HTTP w/Jingle)?

No problems at all with XMPP, except sharing the same problem with SIP,
SIMPLE and all the proprietary IM protocols that are dominant in the IM
market (no names mentioned here): Too complex for generalists Web
application developers to afford the time and resources to learn and to use.

Though it would be useful to have similar APIs developed in the IM industry
ported to Web communications as well.

There is no reason not to replace them all with HTTP transport and APIs for
the client and/or server applications.
Section 1.2 Background in our I-D says:
"  Note that in a similar fashion other network application protocols;
   XMPP [7] or MSRP [8], and RTSP [9] have also one single significant
   application each: IM chat and network media player respectively. This
   is due to the '90s view that new Internet apps require new protocols.
   The advent of RIA has made this view obsolete at present, given the
   seemingly boundless number of new rich Internet applications using
   just HTTP as the only standard network application layer protocol."
(for data only as mentioned elsewhere, and having RTP/UPD as a possible
companion protocol for real-time interactive media).

> 2. What additional work beyond what is already under discussion is necessary
> to complete the vision you describe?

The biggest work item IMO is replacing the stale and problematic SDP with
metadata as detailed in section 2.3.3 and thus also fixing the
non-compliance of SIP with "UNSAF" RFC 3424. Our I-D deals only with SIP at
this point. Replacing SDP and its consequences for NAT traversal requires
indeed some work, possibly not in scope here.

Much more than SDP codec negotiation (and providing misleading transport
addresses) is required to support adequate user experience, depending on
device capabilities, no matter what screen size, keyboard, mouse, touch
screen, remote control, etc. All this can be far better accomplished using
metadata, as in RFC 5013.

Thanks again for focusing on these issues.

Henry 


On 2/3/10 2:55 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:

> There is considerable work already underway to address some of the issues
> raised in this draft.  For example:
> 
> 1. Work in W3C HTML5 WGs to add native support for audio and video.
> 2. Work in IETF "HyBi" WG to make HTTP more suitable for realtime signaling
> 3. Work in XSF to provide enhanced telephony support within XMPP (e.g.
> Jingle ICE, Jingle Transfer, etc.)
> 4. Work in IETF XMPP WG to revise RFC 3920/3921, address SIP/XMPP interop,
> etc. 
> 
> Given this, my reading of the document raised several questions:
> 
> 1.  What problems exist with existing XML over HTTP approaches to signaling
> (e.g. XMPP over HTTP w/Jingle)?
> 2. What additional work beyond what is already under discussion is necessary
> to complete the vision you describe?
> 
> ==========================================================
> Henry said:
> 
> "We have published an I-D on "SIP APIs for Web Communications" and would
> appreciate any comments and input.
> 
> Here is the link and the announcement:
> 
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 
>  Title  : SIP APIs for Communications on the Web
>  Author(s) : H. Sinnreich, A. Johnston
>  Filename : draft-sinnreich-sip-web-apis-00.txt
>  Pages  : 19
>  Date  : 2010-1-19
>  
>    Interactive communications are becoming part of rich Internet
>    applications (RIA). This can be accomplished entirely by using Web
>    technology and developing it as an Internet standard. Voice over IP
>    (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>    can be preserved in rich multimedia communications and applications
>    on the web.  Hyper Text Transport Protocol (HTTP) is used for control
>    and signaling data and RTP for interactive media transport
>    respectively. No other network application protocols are required.
>    Web, fixed and mobile device application development tools can thus
>    be used to include components and widgets for Internet presence,
>    Instant Messaging (IM), voice and video.  This opens Internet
>    communications, including voice to application developers and will
>    also enhance the user experience with real-time Web and mobile
>    communications.  Usage scenarios include service providers, private
>    IP networks, device application developers and media companies
>    providing a mix of integrated communications, applications and media.
>    This memo argues for the development and standardization of
>    Application Programming Interfaces (APIs) to allow the benefits of
>    SIP to be used by Rich Internet Applications (RIA), for fixed and
>    mobile devices.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt"
> 
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch



From bernard_aboba@hotmail.com  Wed Feb  3 18:51:50 2010
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In response to
(http://www.ietf.org/mail-archive/web/dispatch/current/msg01291.html )

Henry Sinnreich said:

"No problems at all with XMPP, except sharing the same problem with SIP,
SIMPLE and all the proprietary IM protocols that are dominant in the IM
market (no names mentioned here): Too complex for generalists Web
application developers to afford the time and resources to learn and to
use."

[BA] Despite all the extensions (see http://xmpp.org/extensions/ ), XMPP as
described in RFC 3920bis and 3921bis is not a complex protocol, nor
is it difficult to incorporate XMPP into applications.  

Have you seen the XMPP SDKs that are available?  Some of them (like
SleekXMPP) are incredibly powerful and easy to use.  Chapter 14 of the
O'Reilly book "XMPP: The Definitive Guide" by Peter St. Andre provides an
example
"micro-blogging" application written using SleekXMPP that requires very
little Python code, and is easy to understand even if you don't know
much Python. 

Now that XMPP is a supported platform within the Google cloud platform,
I expect that the situation will only get better.  

Henry said:

"There is no reason not to replace them all with HTTP transport and APIs for
the client and/or server applications."

[BA] Since there are XMPP SDKs available for PHP, Python and any other 
web development environment you might desire, using XMPP over HTTP should
not require much beyond support for BOSH:
http://xmpp.org/extensions/xep-0124.html
http://xmpp.org/extensions/xep-0206.html

Henry said:

"The biggest work item IMO is replacing the stale and problematic SDP with
metadata as detailed in section 2.3.3 and thus also fixing the
non-compliance of SIP with "UNSAF" RFC 3424. Our I-D deals only with SIP at
this point. Replacing SDP and its consequences for NAT traversal requires
indeed some work, possibly not in scope here."

[BA] I'd be interested in your take on the XMPP session description and
codec functionality:

http://xmpp.org/extensions/xep-0167.html
http://xmpp.org/extensions/xep-0266.html
http://xmpp.org/extensions/xep-0181.html
http://xmpp.org/extensions/xep-0176.html






From bernie@ietf.hoeneisen.ch  Thu Feb  4 01:17:02 2010
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From: Bernie Hoeneisen <bernie@ietf.hoeneisen.ch>
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To: IETF DISPATCH list <dispatch@ietf.org>, IETF ENUM list <enum@ietf.org>,  dnsop@ietf.org, dnsext@ietf.org, EPP Provreg <ietf-provreg@cafax.se>
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Subject: [dispatch] BOF / New Mailing List E2MD (E.164 To MetaData) DDDS application
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Dear DNS, ENUM and RAI experts,

Please be informed that a new mailing list has been created for discussion 
related to a proposed DDDS application 'E.164 to MetaData' (E2MD).

If you are interested in this topic, please subscribe to the new Mailing 
List via:

   https://www.ietf.org/mailman/listinfo/e2md

E2MD is proposed to hold E.164-related information that does not fit the 
normal use cases for E2U. Known use cases include information that 
describes the structure of the E.164 numbering plan, and records which do 
not represent end-user communication URIs.

A BOF about E2MD has been proposed for the upcoming IETF in Anaheim:

   http://trac.tools.ietf.org/bof/trac/#RAI


Note: E2MD has been formerly known as E2M (a naming conflict with a 
legacy SIP WG item required a change of the acronym for this BOF)


cheers,
  Bernie

From john.elwell@siemens-enterprise.com  Thu Feb  4 01:43:03 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Henry Sinnreich <henry.sinnreich@gmail.com>, dispatch mailing list <dispatch@ietf.org>
Date: Thu, 4 Feb 2010 10:43:45 +0100
Thread-Topic: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
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Henry,

Where the draft talks about "porting core SIP functions to standards XML ba=
sed API" (section 4.5), can you shed some more light on this? Is it just a =
case of adopting existing SIP semantics and defining a new syntax (XML)? If=
 so, this would seem to require the web application to support the same com=
plex SIP state machine, with its notions of transactions, dialogs, early di=
alogs, backwards compatibility capabilities, tolerance of unreliable transp=
orts, offer-answer rules, and at least some of the many SIP extensions. Or =
had you intended some sort of simplification, and if so can you give more d=
etail?

Thanks,

John


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
> Sent: 03 February 2010 18:18
> To: dispatch mailing list
> Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
>=20
> We have published an I-D on "SIP APIs for Web Communications"=20
> and would
> appreciate any comments and input.
>=20
> Here is the link and the announcement:
>=20
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
> is-00.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
>=20
>  Title  : SIP APIs for Communications on the Web
>  Author(s) : H. Sinnreich, A. Johnston
>  Filename : draft-sinnreich-sip-web-apis-00.txt
>  Pages  : 19
>  Date  : 2010-1-19
> =20
>    Interactive communications are becoming part of rich Internet
>    applications (RIA). This can be accomplished entirely by using Web
>    technology and developing it as an Internet standard. Voice over IP
>    (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>    can be preserved in rich multimedia communications and applications
>    on the web.  Hyper Text Transport Protocol (HTTP) is used=20
> for control
>    and signaling data and RTP for interactive media transport
>    respectively. No other network application protocols are required.
>    Web, fixed and mobile device application development tools can thus
>    be used to include components and widgets for Internet presence,
>    Instant Messaging (IM), voice and video.  This opens Internet
>    communications, including voice to application developers and will
>    also enhance the user experience with real-time Web and mobile
>    communications.  Usage scenarios include service providers, private
>    IP networks, device application developers and media companies
>    providing a mix of integrated communications, applications=20
> and media.
>    This memo argues for the development and standardization of
>    Application Programming Interfaces (APIs) to allow the benefits of
>    SIP to be used by Rich Internet Applications (RIA), for fixed and
>    mobile devices.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
> is-00.txt
>=20
> ------ End of Forwarded Message
>=20
> Thanks in advance for your comments.
>=20
> Henry
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From andrew.hutton@siemens-enterprise.com  Thu Feb  4 02:11:23 2010
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, Dave Smith <Dave.Smith@genesyslab.com>, Henry Lum <Henry.Lum@genesyslab.com>, Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Feb 2010 11:12:11 +0100
Thread-Topic: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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I can see that the recorded session could in theory be something other than=
 a SIP session however I think that for the purposes of this draft and the =
working group which we hope will be formed shortly that we should assume th=
at the recorded session is a SIP session.

We have requirements and proposed charter text that impact the recorded ses=
sion and will require protocol to be defined.

Of course the recording client can be a SIP gateway which provides interwor=
king with other protocols but this is outside the scope so I think we shoul=
d state that the recorded session is a SIP session.

Regards
Andy


________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Jain, Rajnish
Sent: 02 February 2010 23:43
To: Dave Smith; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Dave,

Inline:

From: Dave Smith [mailto:Dave.Smith@genesyslab.com]
Sent: Tuesday, February 02, 2010 5:33 PM
To: Jain, Rajnish; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Raj,
Regarding this input:
[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).

Can you please provide a few examples of the different types of protocols t=
hat might be possible/practical for the sessions that get actually recorded=
 (which are dependent on the kind of PBX phone used)?

[rj] This can range anywhere from proprietary VoIP media and signaling prot=
ocols to things such as SCCP, UNISTIM, H.323, TDM and analog phones. Most s=
uch phones are not SIP UAs. Some other entity (such as a SIP-enabled mixer)=
 may provide the RC media forking point functionality in these cases.

Thanks,
Raj

From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]
Sent: Monday, February 01, 2010 3:28 PM
To: Henry Lum; Leon Portman; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Henry,

Thanks for your comments. Inline:


-        Section 3 - Recording Client - If the SRP requirement does not man=
date to use SIP as the actual recording protocol, why does the recording cl=
ient have to be a SIP user agent or B2BUA?
[LeonP] Actually there is a difference between Recorded session (the actual=
 call to be recorded) that can be not SIP based and Recording Session (part=
 of SRP) that is used for triggering recording and it is SIP based. So both=
 RC and RS are SIP UA.
[[hlum]] I now understand that the recorded session does not have to be SIP=
 based, but the wording in the definition of SRP itself mentions that it ma=
y be SIP; then why does RC and RS have to be a SIP UA if SRP does not have =
to be SIP?

[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).



-        Section 4 - use case 5 - if the recording can include recording fo=
r an IVR application (ie. VoiceXML application), does that mean the call me=
tadata must include application contexts as well?
[LeonP] Yes, like specific script or input results
[[hlum]] I assume what kind of call metadata and how it is delivered is out=
side of the scope of this document?

[rj] Correct.


o   REQ-003 - it's unclear what is the target of the recording session here=
 since this not directly related to a single recorded session (perhaps many=
)
[LeonP] To minimize media clipping associated with recording initiation
[[hlum]] You actually answered my question above about what a persistent re=
cording mean and why it's useful, but what is unclear to me that is whether=
 there is something called a persistent recording session that targets some=
thing like an agent device.

[rj] I guess, Persistent Recording is a bit confusing term (we've received =
other feedback on this as well). If I can use the term always-on recording =
from the wing-srtp draft then these sessions get established when the agent=
 logs on and torn off when the agent logs off. Codecs such as G.729 and G.7=
11 with VAD are used to ensure that silence is not recorded (for instance, =
when the agent isn't on a call).


o   REQ-009 - the wording here is a bit unclear here. Do you mean the recor=
ded session represents a multi-party session where there are multiple media=
 streams from multiple recorded sessions, and the media streams may be mixe=
d by a conference mixer?
[LeonP]  It is more for trading floor environments where multiple concurren=
t calls on the same device (turret) recorded together. It is not related to=
 summation for conference participants at the same call.
[[hlum]] I'm not familiar with trading floor environments so I may not unde=
rstand the intent here. If you have multiple concurrent calls then wouldn't=
 you have multiple recording sessions to describe each call individually? D=
oes the endpoint device perform mixing function for multiple calls so that'=
s why you need to club them together in a single recording session?

[rj] Due to bandwidth and other cost burdens, the customer demand is that y=
ou mix multiple _recorded sessions_ in a single _recording session_.



o   REQ-028 - the wording redirected looks out of context here; the require=
ment only applies when recording client initiates a recording session to an=
 available recorder server?
[LeonP] We want to support both directions
[[hlum]] Okay, then I think we should re-word to mention something about av=
ailability in general.  (eg. "A request for a new recording session must re=
ach an available recording client or server").

[rj] The example test you quoted doesn't reflect the actual intent here. Th=
e reason for supporting both directions is dependent on where the policy to=
 record or not resides - RC or RS. The INVITE always comes from either of t=
hese two places.

Thanks,
Raj Jain


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>I can see that the recorded session could in theor=
y be=20
something other than a SIP session however I think that for the purposes of=
 this=20
draft and the working group which we hope will be formed shortly that we sh=
ould=20
assume that the recorded session is a SIP session.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>We&nbsp;have requirements and proposed charter=20
text&nbsp;that impact the recorded session and will require protocol to be=
=20
defined.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Of course the recording client can be a SIP gatewa=
y which=20
provides interworking with other protocols but this is outside the scope so=
 I=20
think we should state that the recorded session is a SIP=20
session.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Andy</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D968440210-04022010><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DTahoma size=3D2><B>From:</B>=20
dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] <B>On Behalf O=
f=20
</B>Jain, Rajnish<BR><B>Sent:</B> 02 February 2010 23:43<BR><B>To:</B> Dave=
=20
Smith; Henry Lum; Leon Portman; dispatch@ietf.org<BR><B>Cc:</B>=20
krehor@cisco.com<BR><B>Subject:</B> Re: [dispatch] Comments=20
ondraft-rehor-dispatch-session-recording-req-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Dave,<o:p></o:p></SPAN>=
</P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Inline:<o:p></o:p></SPA=
N></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid=
; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df=
 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium non=
e; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN><=
/B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Dave Smith=20
[mailto:Dave.Smith@genesyslab.com] <BR><B>Sent:</B> Tuesday, February 02, 2=
010=20
5:33 PM<BR><B>To:</B> Jain, Rajnish; Henry Lum; Leon Portman;=20
dispatch@ietf.org<BR><B>Cc:</B> krehor@cisco.com<BR><B>Subject:</B> RE:=20
[dispatch] Comments=20
ondraft-rehor-dispatch-session-recording-req-00<o:p></o:p></SPAN></P></DIV>=
</DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>Raj,<o:p></o:p></P>
<P class=3DMsoNormal>Regarding this input:<o:p></o:p></P>
<P class=3DMsoNormal><SPAN style=3D"FONT-SIZE: 12pt; COLOR: #1f497d">[rj] T=
he SRP,=20
which manages how the recording media gets delivered to the recorders, is b=
ased=20
on SIP (SRP is a set of extensions to the SIP core protocol). The sessions =
that=20
SRP manages are called &#8220;Recording Sessions&#8221; and they are SIP se=
ssions.=20
Therefore, RC and RS are, by definition, SIP UAs. The sessions that get act=
ually=20
recorded (e.g. communication between a call center agent and an external ca=
ller)=20
may or may not be SIP (this depends on the kind of the PBX phone the agent)=
.=20
<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal>Can you please provide a few examples of the different=
 types=20
of protocols that might be possible/practical for <SPAN=20
style=3D"FONT-SIZE: 12pt; COLOR: #1f497d">the sessions that get actually=20
recorded</SPAN> (which are dependent on the kind of PBX phone=20
used)?<o:p></o:p></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj] This can range any=
where=20
from proprietary VoIP media and signaling protocols to things such as SCCP,=
=20
UNISTIM, H.323, TDM and analog phones. Most such phones are not SIP UAs. So=
me=20
other entity (such as a SIP-enabled mixer) may provide the RC media forking=
=20
point functionality in these cases. <o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Thanks,<o:p></o:p></SPA=
N></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Raj<o:p></o:p></SPAN></=
P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df=
 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium non=
e; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B>From:</B> Jain, Rajnish [mailto:Rajnish.Jain@ipc.co=
m]=20
<BR><B>Sent:</B> Monday, February 01, 2010 3:28 PM<BR><B>To:</B> Henry Lum;=
 Leon=20
Portman; dispatch@ietf.org<BR><B>Cc:</B> Dave Smith;=20
krehor@cisco.com<BR><B>Subject:</B> RE: [dispatch] Comments=20
ondraft-rehor-dispatch-session-recording-req-00<o:p></o:p></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Henry,<o:p></o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">Thanks for your comment=
s.=20
Inline:<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: medium =
none; PADDING-LEFT: 4pt; PADDING-BOTTOM: 0in; BORDER-LEFT: blue 1.5pt solid=
; PADDING-TOP: 0in; BORDER-BOTTOM: medium none">
<P class=3DMsoListParagraph=20
style=3D"TEXT-INDENT: -0.25in; mso-list: l0 level1 lfo2"><![if !supportList=
s]><SPAN=20
style=3D"mso-list: Ignore">-<SPAN=20
style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;=20
</SPAN></SPAN><![endif]>Section 3 &#8211; Recording Client &#8211; If the S=
RP requirement=20
does not mandate to use SIP as the actual recording protocol, why does the=
=20
recording client have to be a SIP user agent or B2BUA?<o:p></o:p></P>
<P class=3DMsoNormal><B><I>[LeonP] Actually there is a difference between R=
ecorded=20
session (the actual call to be recorded) that can be not SIP based and Reco=
rding=20
Session (part of SRP) that is used for triggering recording and it is SIP b=
ased.=20
So both RC and RS are SIP UA.<o:p></o:p></I></B></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: blue">[[hlum]] I now understand =
that the=20
recorded session does not have to be SIP based, but the wording in the=20
definition of SRP itself mentions that it may be SIP; then why does RC and =
RS=20
have to be a SIP UA if SRP does not have to be SIP?<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj] The SRP, which man=
ages how=20
the recording media gets delivered to the recorders, is based on SIP (SRP i=
s a=20
set of extensions to the SIP core protocol). The sessions that SRP manages =
are=20
called &#8220;Recording Sessions&#8221; and they are SIP sessions. Therefor=
e, RC and RS are,=20
by definition, SIP UAs. The sessions that get actually recorded (e.g.=20
communication between a call center agent and an external caller) may or ma=
y not=20
be SIP (this depends on the kind of the PBX phone the agent).=20
<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoListParagraph=20
style=3D"TEXT-INDENT: -0.25in; mso-list: l0 level1 lfo2"><![if !supportList=
s]><SPAN=20
style=3D"mso-list: Ignore">-<SPAN=20
style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;=20
</SPAN></SPAN><![endif]>Section 4 &#8211; use case 5 &#8211; if the recordi=
ng can include=20
recording for an IVR application (ie. VoiceXML application), does that mean=
 the=20
call metadata must include application contexts as well?<o:p></o:p></P>
<P class=3DMsoNormal><B><I>[LeonP] Yes, like specific script or input=20
results<o:p></o:p></I></B></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: blue">[[hlum]] I assume what kin=
d of call=20
metadata and how it is delivered is outside of the scope of this=20
document?<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj]=20
Correct.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoListParagraph=20
style=3D"MARGIN-LEFT: 1in; TEXT-INDENT: -0.25in; mso-list: l0 level2 lfo2">=
<![if !supportLists]><SPAN=20
style=3D"FONT-FAMILY: 'Courier New'"><SPAN style=3D"mso-list: Ignore">o<SPA=
N=20
style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;=20
</SPAN></SPAN></SPAN><![endif]>REQ-003 &#8211; it&#8217;s unclear what is t=
he target of the=20
recording session here since this not directly related to a single recorded=
=20
session (perhaps many)<o:p></o:p></P>
<P class=3DMsoNormal><B><I>[LeonP] To minimize media clipping associated wi=
th=20
recording initiation<o:p></o:p></I></B></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: blue">[[hlum]] You actually answ=
ered my=20
question above about what a persistent recording mean and why it&#8217;s us=
eful, but=20
what is unclear to me that is whether there is something called a persisten=
t=20
recording session that targets something like an agent=20
device.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj] I guess, Persisten=
t=20
Recording is a bit confusing term (we&#8217;ve received other feedback on t=
his as=20
well). If I can use the term always-on recording from the wing-srtp draft t=
hen=20
these sessions get established when the agent logs on and torn off when the=
=20
agent logs off. Codecs such as G.729 and G.711 with VAD are used to ensure =
that=20
silence is not recorded (for instance, when the agent isn&#8217;t on a=20
call).<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoListParagraph=20
style=3D"MARGIN-LEFT: 1in; TEXT-INDENT: -0.25in; mso-list: l0 level2 lfo2">=
<![if !supportLists]><SPAN=20
style=3D"FONT-FAMILY: 'Courier New'"><SPAN style=3D"mso-list: Ignore">o<SPA=
N=20
style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;=20
</SPAN></SPAN></SPAN><![endif]>REQ-009 &#8211; the wording here is a bit un=
clear here.=20
Do you mean the recorded session represents a multi-party session where the=
re=20
are multiple media streams from multiple recorded sessions, and the media=20
streams may be mixed by a conference mixer?<o:p></o:p></P>
<P class=3DMsoNormal><B><I>[LeonP] <SPAN style=3D"COLOR: #1f497d">&nbsp;</S=
PAN>It is=20
more for trading floor environments where multiple concurrent calls on the =
same=20
device (turret) recorded together. It is not related to summation for confe=
rence=20
participants at the same call.<o:p></o:p></I></B></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: blue">[[hlum]] I&#8217;m not fam=
iliar with=20
trading floor environments so I may not understand the intent here. If you =
have=20
multiple concurrent calls then wouldn&#8217;t you have multiple recording s=
essions to=20
describe each call individually? Does the endpoint device perform mixing=20
function for multiple calls so that&#8217;s why you need to club them toget=
her in a=20
single recording session?<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj] Due to bandwidth a=
nd other=20
cost burdens, the customer demand is that you mix multiple _<I>recorded=20
sessions</I>_ in a single _<I>recording session</I>_.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoListParagraph=20
style=3D"MARGIN-LEFT: 1in; TEXT-INDENT: -0.25in; mso-list: l0 level2 lfo2">=
<![if !supportLists]><SPAN=20
style=3D"FONT-FAMILY: 'Courier New'"><SPAN style=3D"mso-list: Ignore">o<SPA=
N=20
style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp;=20
</SPAN></SPAN></SPAN><![endif]>REQ-028 &#8211; the wording redirected looks=
 out of=20
context here; the requirement only applies when recording client initiates =
a=20
recording session to an available recorder server?<o:p></o:p></P>
<P class=3DMsoNormal><B><I>[LeonP] We want to support both=20
directions<o:p></o:p></I></B></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: blue">[[hlum]] Okay, then I thin=
k we=20
should re-word to mention something about availability in general.&nbsp; (e=
g. &#8220;A=20
request for a new recording session must reach an available recording clien=
t or=20
server&#8221;).</SPAN><SPAN style=3D"COLOR: #1f497d"> </SPAN><o:p></o:p></P=
>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d"><o:p>&nbsp;</o:p></SPAN=
></P>
<P class=3DMsoNormal><SPAN style=3D"COLOR: #1f497d">[rj] The example test y=
ou quoted=20
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From gonzalo.camarillo@ericsson.com  Thu Feb  4 04:12:16 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [dispatch] ISSUES on draft-jesske-dispatch-reason-in-responses-01 -- Interworking of Responses
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Hi,

talking as a chair, most of the discussions we had last time were about
using Reason in responses versus using encapsulated ISUP REL messages.
Some people claimed that REL messages are rather small and that the
overhead of both approaches would be comparable.

I suggest you try and address those concerns.

Thanks,

Gonzalo
DISPATCH co-chair

R.Jesske@telekom.de wrote:
> Dear all,
> The discussion about the use of the Reason header within responses was
> triggered by the used case where ISUP-SIP-ISUP traversal will happen
> without using SIP-T.
> 
> During the discussion at the last IETF meeting it was asked how the pure
> mapping of ISUP Cause to SIP Response and back will cause the ISUP
> network. Here is the consideration of this issue.
> 
> The case where a call will be released within a pure ISUP network it
> will cause a announcement based on  the ISUP Release Cause.
> 
> When the call instead has to traverse a SIP network the ISUP cause will
> change in most of the cases and the related announcement.
> 
> The figure below shows the principle of such a call ISUP-SIP-ISUP.
> 
>  Phone     PTSN SWITCH            Gateway A          Gateway B     
>        B
>   
>  |               |                   |                  |                  |
>  |  Setup Message|                   |                 
> |                  |
>  |  Phone        |        IAM        |                  |                  |
>  |-------------->|------------------>|     INVITE       |                  |
>  |               |                   |----------------->|      IAM         |
>  |               |                   |     100 Trying   |----------------->|
>  |               |                   |<-----------------|    100 Trying    |
>  |               |                   |   403 Forbidden  |                  |
>  |               |                   |  Reason Q850 #87 |  REL Cause #87   |
>  |  play         |   REL Cause #87   |                 
> |<-----------------|
>  |  Announcement |                   |<-----------------|                  |
>  |<--------------|<----------------- |                  |                  |
>  |               |                   |                  |                  |
>  |               |                   |                  |                  |
>  |               |                   |                  |                  |
> 
> The table below shows now the mappings of the Cause and the relating
> announcements
> Problem is that the mapping regarding RFC 3398 will result in the
> following table.
> 
> EXAMPLE:
> The Call is released with Cause 27 from the user B
> This Cause will normally trigger an announcement B.
> 
> Due to the rules in ISUP the Announcement/Tone is put into the media
> path at the originating switch.
> 
> So the first gateway maps (RFC3398) the ISUP RELEASE message with Cause
> 27 to a SIP response 502 Bad Gateway
> 
> Now mapping back (RFC3398) to ISUP the release cause 38 will be taken
> and this Release Cause will trigger an congestion tone.
> 
> The following table shows now the mapping for our PSTN network and the
> corresponding announcements. 
> 
>          Mapping ISUP SIP  GW-B               Mapping SIP ISUP GW-A
>          ------------------------>             ------------------>
>  
> ISUP  Tone or                                         ISUP
> CAUSE Announcement             SIP Response                 CAUSE      
> Tone or Announcement
> 
>  1      ANNOUNC. A             404 Not Found                 1  ANNOUNC. A
> *2      ANNOUNC. C             404 Not found                 1  ANNOUNC. A
>  3      ANNOUNC. A       404 Not found               1  ANNOUNC. A
>  16     Congestion tone    --- (*)              
>  17     Busy Tone              486 Busy here                17  Busy tone
> *18     Busy Tone        408 Request Timeout          102      
> Congestion tone
>  19     Busy Tone        480 Temporarily unavailable  18        Busy tone
> *20     ANNOUNC. B             480 Temporarily unavailable  18  Busy tone
>  21     ANNOUNC. B             403 Forbidden (+)              21       
> ANNOUNC. B
>  22     ANNOUNC. A             410 Gone                   22    ANNOUNC. A
> *23     Congestion tone    410 Gone                       22    ANNOUNC. A
> *26     Congestion tone    404 Not Found (=)           1        ANNOUNC. A
> 
> *27     ANNOUNC. B             502 Bad Gateway              38 
> Congestion tone
> 
> *28     Congestion tone  484 Address incomplete       28       
> Congestion tone
>  29     ANNOUNC. D          501 Not implemented       79        ANNOUNC. D
> *31     Congestion tone    480 Temporarily unavailable  18      Busy tone
>  34     Congestion tone    503 Service unavailable      41     
> Congestion tone
>  38     Congestion tone    503 Service unavailable      41     
> Congestion tone
>  41     Congestion tone  503 Service unavailable      41       
> Congestion tone
> *42     ANNOUNC. B             503 Service unavailable      41 
> Congestion tone
>  47     Congestion tone  503 Service unavailable      41       
> Congestion tone
> *55     ANNOUNC. D             403 Forbidden                21  ANNOUNC. B
> *57     ANNOUNC. D             403 Forbidden                21  ANNOUNC. B
>  58     Congestion tone    503 Service unavailable      41     
> Congestion tone
> *65     ANNOUNC. D             488 Not Acceptable Here      127
> Congestion tone  (**) 
> *70     ANNOUNC. D             488 Not Acceptable Here      127
> Congestion tone  (**)
>  79     ANNOUNC. D             501 Not implemented            79       
> ANNOUNC. D
> *87     ANNOUNC. D             403 Forbidden                21  ANNOUNC. B
> *88     ANNOUNC. D             503 Service unavailable      41 
> Congestion tone
>  102    Congestion tone    504 Gateway timeout        102      
> Congestion tone
> *111    ANNOUNC. D             500 Server internal error    41 
> Congestion tone
>  127    Congestion tone    500 Server internal error    41     
> Congestion tone
> 
> (*) Announcement change due to passing the SIP domain
> (**)No mapping in RFC 3398 described in the field 127 is used then as
> default mapping.
> 
> Half of the mapped Responses will end in a ISUP Cause that will initiate
> an other announcement than intended from the originator of the Release
> cause.
> 
> That is why a mapping of a Release Cause into an Reason Header and back
> to the ISUP cause will help in an easy using an existing SIP header that
> is intended to carry ISUP causes.
> 
> 
> Kind regards
> Roland Jesske
> 
> 
> Deutsche Telekom Netzproduktion GmbH
> Technical Engineering Center
> Roland Jesske
> Heinrich-Hertz-Straße 3-7, 64295 Darmstadt, Germany
> +49 6151 628-2766 (Phone)
> +49 521 9210-3753  (Fax)
> +49 171 8618-445  (Mobile)
> E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de>
> www.telekom.com <http:www.telekom.com> 
> 
> Life is for sharing.
> 
> Deutsche Telekom Netzproduktion GmbH
> Supervisory Board: Dr. Steffen Roehn (Chairman)
> Managing Board: Dr. Bruno Jacobfeuerborn (Chairman), Albert Matheis,
> Klaus Peren
> Commercial register: Local court Bonn HRB 14190
> Registered office: Bonn
> VAT identification no. DE 814645262
> 


From alan.b.johnston@gmail.com  Thu Feb  4 05:36:09 2010
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John,

Thanks for your comments.

We are not thinking about just an XML encoding for SIP - you are
correct, that would still expose way too much complexity.  We were
thinking about starting with the functionality in RFC 5638, then see how
much we can abstract and separate the "plumbing" of SIP from what an app
developer needs/cares about.  We hope to avoid replicating the entire
SIP state machine.

However, this is just a starting point, and feedback and input from
others who share our goal is most welcome.

- Alan -

On 2/4/10 3:43 AM, Elwell, John wrote:
> Henry,
> 
> Where the draft talks about "porting core SIP functions to standards XML based API" (section 4.5), can you shed some more light on this? Is it just a case of adopting existing SIP semantics and defining a new syntax (XML)? If so, this would seem to require the web application to support the same complex SIP state machine, with its notions of transactions, dialogs, early dialogs, backwards compatibility capabilities, tolerance of unreliable transports, offer-answer rules, and at least some of the many SIP extensions. Or had you intended some sort of simplification, and if so can you give more detail?
> 
> Thanks,
> 
> John
> 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
>> Sent: 03 February 2010 18:18
>> To: dispatch mailing list
>> Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
>>
>> We have published an I-D on "SIP APIs for Web Communications" 
>> and would
>> appreciate any comments and input.
>>
>> Here is the link and the announcement:
>>
>> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>> is-00.txt
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>>
>>  Title  : SIP APIs for Communications on the Web
>>  Author(s) : H. Sinnreich, A. Johnston
>>  Filename : draft-sinnreich-sip-web-apis-00.txt
>>  Pages  : 19
>>  Date  : 2010-1-19
>>  
>>    Interactive communications are becoming part of rich Internet
>>    applications (RIA). This can be accomplished entirely by using Web
>>    technology and developing it as an Internet standard. Voice over IP
>>    (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>>    can be preserved in rich multimedia communications and applications
>>    on the web.  Hyper Text Transport Protocol (HTTP) is used 
>> for control
>>    and signaling data and RTP for interactive media transport
>>    respectively. No other network application protocols are required.
>>    Web, fixed and mobile device application development tools can thus
>>    be used to include components and widgets for Internet presence,
>>    Instant Messaging (IM), voice and video.  This opens Internet
>>    communications, including voice to application developers and will
>>    also enhance the user experience with real-time Web and mobile
>>    communications.  Usage scenarios include service providers, private
>>    IP networks, device application developers and media companies
>>    providing a mix of integrated communications, applications 
>> and media.
>>    This memo argues for the development and standardization of
>>    Application Programming Interfaces (APIs) to allow the benefits of
>>    SIP to be used by Rich Internet Applications (RIA), for fixed and
>>    mobile devices.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>> is-00.txt
>>
>> ------ End of Forwarded Message
>>
>> Thanks in advance for your comments.
>>
>> Henry
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

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From: "Jain, Rajnish" <Rajnish.Jain@ipc.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Dave Smith <Dave.Smith@genesyslab.com>, Henry Lum <Henry.Lum@genesyslab.com>, Leon Portman <Leon.Portman@nice.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Feb 2010 08:50:37 -0500
Thread-Topic: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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I agree with this proposal in the interest of moving this work forward.

Regards,
Raj


From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]=20
Sent: Thursday, February 04, 2010 5:12 AM
To: Jain, Rajnish; Dave Smith; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

I can see that the recorded session could in theory be something other than=
 a SIP session however I think that for the purposes of this draft and the =
working group which we hope will be formed shortly that we should assume th=
at the recorded session is a SIP session.
=A0
We=A0have requirements and proposed charter text=A0that impact the recorded=
 session and will require protocol to be defined.
=A0
Of course the recording client can be a SIP gateway which provides interwor=
king with other protocols but this is outside the scope so I think we shoul=
d state that the recorded session is a SIP session.
=A0
Regards
Andy
=A0
=A0
________________________________________
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Jain, Rajnish
Sent: 02 February 2010 23:43
To: Dave Smith; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00
Dave,

Inline:

From: Dave Smith [mailto:Dave.Smith@genesyslab.com]=20
Sent: Tuesday, February 02, 2010 5:33 PM
To: Jain, Rajnish; Henry Lum; Leon Portman; dispatch@ietf.org
Cc: krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Raj,
Regarding this input:
[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).=20

Can you please provide a few examples of the different types of protocols t=
hat might be possible/practical for the sessions that get actually recorded=
 (which are dependent on the kind of PBX phone used)?

[rj] This can range anywhere from proprietary VoIP media and signaling prot=
ocols to things such as SCCP, UNISTIM, H.323, TDM and analog phones. Most s=
uch phones are not SIP UAs. Some other entity (such as a SIP-enabled mixer)=
 may provide the RC media forking point functionality in these cases.=20

Thanks,
Raj

From: Jain, Rajnish [mailto:Rajnish.Jain@ipc.com]=20
Sent: Monday, February 01, 2010 3:28 PM
To: Henry Lum; Leon Portman; dispatch@ietf.org
Cc: Dave Smith; krehor@cisco.com
Subject: RE: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Henry,

Thanks for your comments. Inline:

-=A0=A0=A0=A0=A0=A0=A0 Section 3 - Recording Client - If the SRP requiremen=
t does not mandate to use SIP as the actual recording protocol, why does th=
e recording client have to be a SIP user agent or B2BUA?
[LeonP] Actually there is a difference between Recorded session (the actual=
 call to be recorded) that can be not SIP based and Recording Session (part=
 of SRP) that is used for triggering recording and it is SIP based. So both=
 RC and RS are SIP UA.
[[hlum]] I now understand that the recorded session does not have to be SIP=
 based, but the wording in the definition of SRP itself mentions that it ma=
y be SIP; then why does RC and RS have to be a SIP UA if SRP does not have =
to be SIP?

[rj] The SRP, which manages how the recording media gets delivered to the r=
ecorders, is based on SIP (SRP is a set of extensions to the SIP core proto=
col). The sessions that SRP manages are called "Recording Sessions" and the=
y are SIP sessions. Therefore, RC and RS are, by definition, SIP UAs. The s=
essions that get actually recorded (e.g. communication between a call cente=
r agent and an external caller) may or may not be SIP (this depends on the =
kind of the PBX phone the agent).=20


-=A0=A0=A0=A0=A0=A0=A0 Section 4 - use case 5 - if the recording can includ=
e recording for an IVR application (ie. VoiceXML application), does that me=
an the call metadata must include application contexts as well?
[LeonP] Yes, like specific script or input results
[[hlum]] I assume what kind of call metadata and how it is delivered is out=
side of the scope of this document?

[rj] Correct.

o=A0=A0 REQ-003 - it's unclear what is the target of the recording session =
here since this not directly related to a single recorded session (perhaps =
many)
[LeonP] To minimize media clipping associated with recording initiation
[[hlum]] You actually answered my question above about what a persistent re=
cording mean and why it's useful, but what is unclear to me that is whether=
 there is something called a persistent recording session that targets some=
thing like an agent device.

[rj] I guess, Persistent Recording is a bit confusing term (we've received =
other feedback on this as well). If I can use the term always-on recording =
from the wing-srtp draft then these sessions get established when the agent=
 logs on and torn off when the agent logs off. Codecs such as G.729 and G.7=
11 with VAD are used to ensure that silence is not recorded (for instance, =
when the agent isn't on a call).

o=A0=A0 REQ-009 - the wording here is a bit unclear here. Do you mean the r=
ecorded session represents a multi-party session where there are multiple m=
edia streams from multiple recorded sessions, and the media streams may be =
mixed by a conference mixer?
[LeonP] =A0It is more for trading floor environments where multiple concurr=
ent calls on the same device (turret) recorded together. It is not related =
to summation for conference participants at the same call.
[[hlum]] I'm not familiar with trading floor environments so I may not unde=
rstand the intent here. If you have multiple concurrent calls then wouldn't=
 you have multiple recording sessions to describe each call individually? D=
oes the endpoint device perform mixing function for multiple calls so that'=
s why you need to club them together in a single recording session?

[rj] Due to bandwidth and other cost burdens, the customer demand is that y=
ou mix multiple _recorded sessions_ in a single _recording session_.


o=A0=A0 REQ-028 - the wording redirected looks out of context here; the req=
uirement only applies when recording client initiates a recording session t=
o an available recorder server?
[LeonP] We want to support both directions
[[hlum]] Okay, then I think we should re-word to mention something about av=
ailability in general.=A0 (eg. "A request for a new recording session must =
reach an available recording client or server").=20

[rj] The example test you quoted doesn't reflect the actual intent here. Th=
e reason for supporting both directions is dependent on where the policy to=
 record or not resides - RC or RS. The INVITE always comes from either of t=
hese two places.

Thanks,
Raj Jain


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From peter.musgrave@magorcorp.com  Thu Feb  4 06:15:50 2010
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Subject: Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
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Henry/Alan,

I think some mention of other web technologies in the draft might help frame
the effort.

e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP-like
media constructs, uses ICE and if I recall correctly does put address inside
messages).

If I had a web API which allowed direct connections to SIP endpoints/media
servers I think that would be powerful. In the past I have done things like
created a "B2B-like" entity with a Jingle stack and a SIP stack - but then I
end up with identity issues and issues with simultaneous sessions to media
servers.

I agree that a very simple call state machine is desired - even if the SIP
mechanics need to be strongly restricted. I also agree that SDP is getting a
bit tired.

Peter

On Thu, Feb 4, 2010 at 8:36 AM, Alan Johnston <alan.b.johnston@gmail.com>wrote:

> John,
>
> Thanks for your comments.
>
> We are not thinking about just an XML encoding for SIP - you are
> correct, that would still expose way too much complexity.  We were
> thinking about starting with the functionality in RFC 5638, then see how
> much we can abstract and separate the "plumbing" of SIP from what an app
> developer needs/cares about.  We hope to avoid replicating the entire
> SIP state machine.
>
> However, this is just a starting point, and feedback and input from
> others who share our goal is most welcome.
>
> - Alan -
>
> On 2/4/10 3:43 AM, Elwell, John wrote:
> > Henry,
> >
> > Where the draft talks about "porting core SIP functions to standards XML
> based API" (section 4.5), can you shed some more light on this? Is it just a
> case of adopting existing SIP semantics and defining a new syntax (XML)? If
> so, this would seem to require the web application to support the same
> complex SIP state machine, with its notions of transactions, dialogs, early
> dialogs, backwards compatibility capabilities, tolerance of unreliable
> transports, offer-answer rules, and at least some of the many SIP
> extensions. Or had you intended some sort of simplification, and if so can
> you give more detail?
> >
> > Thanks,
> >
> > John
> >
> >
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
> >> Sent: 03 February 2010 18:18
> >> To: dispatch mailing list
> >> Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
> >>
> >> We have published an I-D on "SIP APIs for Web Communications"
> >> and would
> >> appreciate any comments and input.
> >>
> >> Here is the link and the announcement:
> >>
> >> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
> >> is-00.txt
> >>
> >> A New Internet-Draft is available from the on-line Internet-Drafts
> >> directories.
> >>
> >>
> >>  Title  : SIP APIs for Communications on the Web
> >>  Author(s) : H. Sinnreich, A. Johnston
> >>  Filename : draft-sinnreich-sip-web-apis-00.txt
> >>  Pages  : 19
> >>  Date  : 2010-1-19
> >>
> >>    Interactive communications are becoming part of rich Internet
> >>    applications (RIA). This can be accomplished entirely by using Web
> >>    technology and developing it as an Internet standard. Voice over IP
> >>    (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
> >>    can be preserved in rich multimedia communications and applications
> >>    on the web.  Hyper Text Transport Protocol (HTTP) is used
> >> for control
> >>    and signaling data and RTP for interactive media transport
> >>    respectively. No other network application protocols are required.
> >>    Web, fixed and mobile device application development tools can thus
> >>    be used to include components and widgets for Internet presence,
> >>    Instant Messaging (IM), voice and video.  This opens Internet
> >>    communications, including voice to application developers and will
> >>    also enhance the user experience with real-time Web and mobile
> >>    communications.  Usage scenarios include service providers, private
> >>    IP networks, device application developers and media companies
> >>    providing a mix of integrated communications, applications
> >> and media.
> >>    This memo argues for the development and standardization of
> >>    Application Programming Interfaces (APIs) to allow the benefits of
> >>    SIP to be used by Rich Internet Applications (RIA), for fixed and
> >>    mobile devices.
> >>
> >> A URL for this Internet-Draft is:
> >> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
> >> is-00.txt
> >>
> >> ------ End of Forwarded Message
> >>
> >> Thanks in advance for your comments.
> >>
> >> Henry
> >>
> >>
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

--000e0cd138e062db70047ec6fb00
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Henry/Alan,=A0<div><br></div><div>I think some mention of other web technol=
ogies in the draft might help frame the effort.</div><div><br></div><div>e.=
g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP-like =
media constructs, uses ICE and if I recall correctly does put address insid=
e messages).=A0</div>
<div><br></div><div>If I had a web API which allowed direct connections to =
SIP endpoints/media servers I think that would be powerful. In the past I h=
ave done things like created a &quot;B2B-like&quot; entity with a Jingle st=
ack and a SIP stack - but then I end up with identity issues and issues wit=
h simultaneous sessions to media servers.</div>
<div><br></div><div>I agree that a very simple call state machine is desire=
d - even if the SIP mechanics need to be strongly restricted. I also agree =
that SDP is getting a bit tired.=A0</div><div><br></div><div>Peter<br><br>
<div class=3D"gmail_quote">On Thu, Feb 4, 2010 at 8:36 AM, Alan Johnston <s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:alan.b.johnston@gmail.com">alan.b.joh=
nston@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
John,<br>
<br>
Thanks for your comments.<br>
<br>
We are not thinking about just an XML encoding for SIP - you are<br>
correct, that would still expose way too much complexity. =A0We were<br>
thinking about starting with the functionality in RFC 5638, then see how<br=
>
much we can abstract and separate the &quot;plumbing&quot; of SIP from what=
 an app<br>
developer needs/cares about. =A0We hope to avoid replicating the entire<br>
SIP state machine.<br>
<br>
However, this is just a starting point, and feedback and input from<br>
others who share our goal is most welcome.<br>
<font color=3D"#888888"><br>
- Alan -<br>
</font><div><div></div><div class=3D"h5"><br>
On 2/4/10 3:43 AM, Elwell, John wrote:<br>
&gt; Henry,<br>
&gt;<br>
&gt; Where the draft talks about &quot;porting core SIP functions to standa=
rds XML based API&quot; (section 4.5), can you shed some more light on this=
? Is it just a case of adopting existing SIP semantics and defining a new s=
yntax (XML)? If so, this would seem to require the web application to suppo=
rt the same complex SIP state machine, with its notions of transactions, di=
alogs, early dialogs, backwards compatibility capabilities, tolerance of un=
reliable transports, offer-answer rules, and at least some of the many SIP =
extensions. Or had you intended some sort of simplification, and if so can =
you give more detail?<br>

&gt;<br>
&gt; Thanks,<br>
&gt;<br>
&gt; John<br>
&gt;<br>
&gt;<br>
&gt;&gt; -----Original Message-----<br>
&gt;&gt; From: <a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-bounce=
s@ietf.org</a><br>
&gt;&gt; [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org">dispatch-boun=
ces@ietf.org</a>] On Behalf Of Henry Sinnreich<br>
&gt;&gt; Sent: 03 February 2010 18:18<br>
&gt;&gt; To: dispatch mailing list<br>
&gt;&gt; Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00=
.txt<br>
&gt;&gt;<br>
&gt;&gt; We have published an I-D on &quot;SIP APIs for Web Communications&=
quot;<br>
&gt;&gt; and would<br>
&gt;&gt; appreciate any comments and input.<br>
&gt;&gt;<br>
&gt;&gt; Here is the link and the announcement:<br>
&gt;&gt;<br>
&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip=
-web-ap" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-sinnre=
ich-sip-web-ap</a><br>
&gt;&gt; is-00.txt<br>
&gt;&gt;<br>
&gt;&gt; A New Internet-Draft is available from the on-line Internet-Drafts=
<br>
&gt;&gt; directories.<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; =A0Title =A0: SIP APIs for Communications on the Web<br>
&gt;&gt; =A0Author(s) : H. Sinnreich, A. Johnston<br>
&gt;&gt; =A0Filename : draft-sinnreich-sip-web-apis-00.txt<br>
&gt;&gt; =A0Pages =A0: 19<br>
&gt;&gt; =A0Date =A0: 2010-1-19<br>
&gt;&gt;<br>
&gt;&gt; =A0 =A0Interactive communications are becoming part of rich Intern=
et<br>
&gt;&gt; =A0 =A0applications (RIA). This can be accomplished entirely by us=
ing Web<br>
&gt;&gt; =A0 =A0technology and developing it as an Internet standard. Voice=
 over IP<br>
&gt;&gt; =A0 =A0(VoIP) features developed for Session Initiation Protocol (=
SIP 2.0)<br>
&gt;&gt; =A0 =A0can be preserved in rich multimedia communications and appl=
ications<br>
&gt;&gt; =A0 =A0on the web. =A0Hyper Text Transport Protocol (HTTP) is used=
<br>
&gt;&gt; for control<br>
&gt;&gt; =A0 =A0and signaling data and RTP for interactive media transport<=
br>
&gt;&gt; =A0 =A0respectively. No other network application protocols are re=
quired.<br>
&gt;&gt; =A0 =A0Web, fixed and mobile device application development tools =
can thus<br>
&gt;&gt; =A0 =A0be used to include components and widgets for Internet pres=
ence,<br>
&gt;&gt; =A0 =A0Instant Messaging (IM), voice and video. =A0This opens Inte=
rnet<br>
&gt;&gt; =A0 =A0communications, including voice to application developers a=
nd will<br>
&gt;&gt; =A0 =A0also enhance the user experience with real-time Web and mob=
ile<br>
&gt;&gt; =A0 =A0communications. =A0Usage scenarios include service provider=
s, private<br>
&gt;&gt; =A0 =A0IP networks, device application developers and media compan=
ies<br>
&gt;&gt; =A0 =A0providing a mix of integrated communications, applications<=
br>
&gt;&gt; and media.<br>
&gt;&gt; =A0 =A0This memo argues for the development and standardization of=
<br>
&gt;&gt; =A0 =A0Application Programming Interfaces (APIs) to allow the bene=
fits of<br>
&gt;&gt; =A0 =A0SIP to be used by Rich Internet Applications (RIA), for fix=
ed and<br>
&gt;&gt; =A0 =A0mobile devices.<br>
&gt;&gt;<br>
&gt;&gt; A URL for this Internet-Draft is:<br>
&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip=
-web-ap" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-sinnre=
ich-sip-web-ap</a><br>
&gt;&gt; is-00.txt<br>
&gt;&gt;<br>
&gt;&gt; ------ End of Forwarded Message<br>
&gt;&gt;<br>
&gt;&gt; Thanks in advance for your comments.<br>
&gt;&gt;<br>
&gt;&gt; Henry<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; _______________________________________________<br>
&gt;&gt; dispatch mailing list<br>
&gt;&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=
=3D"_blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;&gt;<br>
&gt; _______________________________________________<br>
&gt; dispatch mailing list<br>
&gt; <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
&gt;<br>
_______________________________________________<br>
dispatch mailing list<br>
<a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/dispatch</a><br>
</div></div></blockquote></div><br></div>

--000e0cd138e062db70047ec6fb00--

From john.elwell@siemens-enterprise.com  Thu Feb  4 06:16:20 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Feb 2010 15:16:57 +0100
Thread-Topic: [dispatch] ISSUES on draft-jesske-dispatch-reason-in-responses-01	-- Use Reason in Responses
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Roland,

Another use case we have come across is where we receive from Q.931 during =
outgoing call establishment a Disconnect message indicating busy and a prog=
ress indicator saying there is an in-band announcement. This needs to be ma=
pped to a SIP 183 response (include SDP answer if not already sent), rather=
 than a final response, so that the caller can hear the announcement from P=
STN. However, being able to indicate busy in the Reason header field would =
allow an appropriate display on the device.

John=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of R.Jesske@telekom.de
> Sent: 28 January 2010 07:16
> To: dispatch@ietf.org
> Subject: [dispatch] ISSUES on=20
> draft-jesske-dispatch-reason-in-responses-01 -- Use Reason in=20
> Responses
>=20
> Dear all,=20
> The discussion at the last IETF meeting due to the proposal=20
> to use the reason header within responses where controversial.
>=20
> I would like to start the discussion on mentioned concerns.=20
>=20
> One concern was that we define an new encapsulation mechanism=20
> for ISUP elements in parallel to SIP-T.=20
>=20
> The Reason header itself is a SIP generic Header defined in RFC3326.=20
>=20
> The syntax allows to put in a SIP Response code and/or an=20
> ISUP Q.850 Cause Code.=20
>=20
> As many times mentioned before RFC 3326 (The Reason Header=20
> Field for the Session Initiation Protocol (SIP)) defines the=20
> following.
>=20
>    Initially, the Reason header field defined here appears to be most=20
>    useful for BYE and CANCEL requests, but it can appear in=20
> any request=20
>    within a dialog, in any CANCEL request and in any response whose=20
>    status code explicitly allows the presence of this header field.=20
>=20
>    Note that the Reason header field is usually not needed in=20
> responses=20
>    because the status code and the reason phrase already provide=20
>    sufficient information.=20
>=20
>    Clients and servers are free to ignore this header field. =20
> It has no=20
>    impact on protocol processing.=20
>=20
> At that time where RFC3326 was written it was not assumed=20
> that a Reason in a response is usually not needed but it does=20
> not restrict the use of the Reason header within responses.=20
> It allows the use of the Reason header if the status code allows it.
>=20
> That was the reasoning for writing this draft to allow the=20
> Reason header within Responses and it is not an new=20
> encapsulation mechanism.
>=20
> Kind regards=20
> Roland Jesske=20
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH=20
> Technical Engineering Center=20
> Roland Jesske=20
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt, Germany=20
> +49 6151 628-2766 (Phone)=20
> +49 521 9210-3753  (Fax)=20
> +49 171 8618-445  (Mobile)=20
> E-Mail: r.jesske@telekom.de <mailto:r.jesske@telekom.de> =20
> www.telekom.com <http:www.telekom.com> =20
>=20
> Life is for sharing.=20
>=20
> Deutsche Telekom Netzproduktion GmbH=20
> Supervisory Board: Dr. Steffen Roehn (Chairman)=20
> Managing Board: Dr. Bruno Jacobfeuerborn (Chairman), Albert=20
> Matheis, Klaus Peren=20
> Commercial register: Local court Bonn HRB 14190=20
> Registered office: Bonn=20
> VAT identification no. DE 814645262=20
>=20
> =

From BeckW@telekom.de  Thu Feb  4 06:58:59 2010
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=20
> 2.1. SIP 2.0 is too expensive for Web application developers
The big question is whether SIP is to blame for this complexity or if =
it's in the nature of telephony. Web standards are as complex as SIP. =
Only very few web browsers pass the acid3 test. XML is not the most =
elegant technology either.

There must be a different reason why so many develop for the web and so =
few for voice. With Flash, it is already possible to do SIP-like =
voice/video connections between users. It's rarely used, though. When =
eBay bought Skype, everybody expected exciting mash-ups. Nothing =
happened.=20

> 4.1.1. Applications reside in endpoints of the Internet
Kind of. The success of Web 2.0 was the success of Javascript; =
applications run on the endpoint, but are maintained on the server. The =
user has no choice which javascript applet to use when accessing a web =
2.0 site. The site owner can update the applet without having to ask the =
browser owner for approval.


Wolfgang Beck


Deutsche Telekom Netzproduktion GmbH
Zentrum Technik Einf=FChrung
Wolfgang Beck
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
+49 6151 628 2832 (Tel.)
http://www.telekom.com =20

Erleben, was verbindet.=20

Deutsche Telekom Netzproduktion GmbH
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr. DE 814645262


From henry.sinnreich@gmail.com  Thu Feb  4 07:31:18 2010
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Thanks Wolfgang for raising these interesting issues.

> The big question is whether SIP is to blame for this complexity or if it'=
s in
> the nature of telephony.
My vote goes it's in the nature of traditional telephony. It is arguable ho=
w
much traditional telephony is of interest today for new investments however=
,
given especially for younger users accustomed to Web applications and to
Skype.
Web communications may also lead to much better enabled "call centers".

>Web standards are as complex as SIP.
Very true, but APIs and IDEs completely hide that complexity.
Whatever the reasons (such as market size for developers), there are
zillions of Web applications, and only one major app for SIP: Re-engineerin=
g
traditional telephony over IP.

> XML is not the most elegant technology either.
Beauty is in the eye of the beholder :-)
We could think only of XML as a platform independent standard for APIs.
XML is also fully supported in IDE we happen to know, such as Eclipse and
Flash that you also mention. Let us know if have any alternative(s) to XML.

Thanks again,

Henry

On 2/4/10 8:56 AM, "BeckW@telekom.de" <BeckW@telekom.de> wrote:

> =20
>> 2.1. SIP 2.0 is too expensive for Web application developers
> The big question is whether SIP is to blame for this complexity or if it'=
s in
> the nature of telephony. Web standards are as complex as SIP. Only very f=
ew
> web browsers pass the acid3 test. XML is not the most elegant technology
> either.
>=20
> There must be a different reason why so many develop for the web and so f=
ew
> for voice. With Flash, it is already possible to do SIP-like voice/video
> connections between users. It's rarely used, though. When eBay bought Sky=
pe,
> everybody expected exciting mash-ups. Nothing happened.
>=20
>> 4.1.1. Applications reside in endpoints of the Internet
> Kind of. The success of Web 2.0 was the success of Javascript; applicatio=
ns
> run on the endpoint, but are maintained on the server. The user has no ch=
oice
> which javascript applet to use when accessing a web 2.0 site. The site ow=
ner
> can update the applet without having to ask the browser owner for approva=
l.
>=20
>=20
> Wolfgang Beck
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Zentrum Technik Einf=FChrung
> Wolfgang Beck
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628 2832 (Tel.)
> http://www.telekom.com
>=20
> Erleben, was verbindet.
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert Matheis=
,
> Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20



From henry.sinnreich@gmail.com  Thu Feb  4 07:55:25 2010
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Date: Thu, 04 Feb 2010 09:56:02 -0600
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>, <dispatch@ietf.org>
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There is no disagreement here about the merits of XMPP, however this I-D is
focused on SIP APIs and the expertise of the present authors is mainly in
SIP. 

It would be useful to have similar XMPP APIs for Web communications, as well
as APIs from the other dominant IM protocols.

Application developers would thus have a larger choice, depending on the
usage scenario.

> [BA] I'd be interested in your take on the XMPP session description and
> codec functionality
No disagreement here either, though the I-D argues for using metadata
instead of SDP, that goes way beyond codec negotiation. In such applications
as Web conferencing with screen sharing, telepresence for business and the
digital living room, much more platform-specific data needs to be considered
in the application than just codecs.

Screen sharing example: One, two or more screens in an IT customer support
scenario on both ends, with different screen resolutions. Without metadata
the parties will not see the same desired screens.

Thanks again for the good probing questions,

Henry


On 2/3/10 8:52 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:

> In response to
> (http://www.ietf.org/mail-archive/web/dispatch/current/msg01291.html )
> 
> Henry Sinnreich said:
> 
> "No problems at all with XMPP, except sharing the same problem with SIP,
> SIMPLE and all the proprietary IM protocols that are dominant in the IM
> market (no names mentioned here): Too complex for generalists Web
> application developers to afford the time and resources to learn and to
> use."
> 
> [BA] Despite all the extensions (see http://xmpp.org/extensions/ ), XMPP as
> described in RFC 3920bis and 3921bis is not a complex protocol, nor
> is it difficult to incorporate XMPP into applications.
> 
> Have you seen the XMPP SDKs that are available?  Some of them (like
> SleekXMPP) are incredibly powerful and easy to use.  Chapter 14 of the
> O'Reilly book "XMPP: The Definitive Guide" by Peter St. Andre provides an
> example
> "micro-blogging" application written using SleekXMPP that requires very
> little Python code, and is easy to understand even if you don't know
> much Python. 
> 
> Now that XMPP is a supported platform within the Google cloud platform,
> I expect that the situation will only get better.
> 
> Henry said:
> 
> "There is no reason not to replace them all with HTTP transport and APIs for
> the client and/or server applications."
> 
> [BA] Since there are XMPP SDKs available for PHP, Python and any other
> web development environment you might desire, using XMPP over HTTP should
> not require much beyond support for BOSH:
> http://xmpp.org/extensions/xep-0124.html
> http://xmpp.org/extensions/xep-0206.html
> 
> Henry said:
> 
> "The biggest work item IMO is replacing the stale and problematic SDP with
> metadata as detailed in section 2.3.3 and thus also fixing the
> non-compliance of SIP with "UNSAF" RFC 3424. Our I-D deals only with SIP at
> this point. Replacing SDP and its consequences for NAT traversal requires
> indeed some work, possibly not in scope here."
> 
> [BA] I'd be interested in your take on the XMPP session description and
> codec functionality:
> 
> http://xmpp.org/extensions/xep-0167.html
> http://xmpp.org/extensions/xep-0266.html
> http://xmpp.org/extensions/xep-0181.html
> http://xmpp.org/extensions/xep-0176.html
> 
> 
> 
> 
> 



From Leon.Portman@nice.com  Thu Feb  4 08:04:52 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: zhouzhipeng 54906 <zhouzp@huawei.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Thu, 4 Feb 2010 18:05:33 +0200
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Zhipeng, Hi

The core point of this document is to define a SIP based protocol that will=
 allow Active VoIP recording, where a some entity from communication system=
 (gateway, phone, media server) will fork media of the call to recording se=
rver.

The interface for querying and playback of recorded sessions are currently =
out of scope of this work because it probably should be a web service or RE=
ST interface. However we are definitely considering to work on it as well.

Regards

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of zhouzhipeng 54906
Sent: Wednesday, February 03, 2010 11:52 AM
To: dispatch@ietf.org
Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Dear all,
I have got much valuable information from related emails.
As I cited as following,

> [rj] This can range anywhere from proprietary VoIP media and=20
> signaling protocols to things such as SCCP, UNISTIM, H.323, TDM=20
> and analog phones. Most such phones are not SIP UAs. Some other=20
> entity (such as a SIP-enabled mixer) may provide the RC media=20
> forking point functionality in these cases.

there have been many years for the application of recording.=20
So, may the author give us a brief summary what have been advanced based on=
 current recording solutions, for example the call center architecture. So =
we can easily catch the core point of this document.

Another point is: does the diagram need to add the interface between UA to =
the RS for the search or query functions.

Thanks very much.
Zhipeng

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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Peter Musgrave wrote:
>  I also agree that SDP is getting a bit tired. 

SDP was tired in 2000. It should have been buried long ago.

	Paul

From henry.sinnreich@gmail.com  Thu Feb  4 08:10:27 2010
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From: Henry Sinnreich <henry.sinnreich@gmail.com>
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Peter,

>e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP-li=
ke
media constructs, uses ICE and if I recall correctly does put address insid=
e
messages).=A0

As mentioned, our expertise and focus here is in SIP and similar APIs for
Jingle would give developers a bigger choice, depending on the usage
scenario.
As for ICE, we argue it to work in HIP for all application protocols alike,
besides all the other reasons for using HIP.

>If I had a web API which allowed direct connections to SIP endpoints/media
servers I think that would be powerful.
Yes, since all or most of the telecom gear supports SIP, we believe there
are lots of such apps.

Henry


On 2/4/10 8:16 AM, "Peter Musgrave" <peter.musgrave@magorcorp.com> wrote:

> Henry/Alan,=A0
>=20
> I think some mention of other web technologies in the draft might help fr=
ame
> the effort.
>=20
> e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP-l=
ike
> media constructs, uses ICE and if I recall correctly does put address ins=
ide
> messages).=A0
>=20
> If I had a web API which allowed direct connections to SIP endpoints/medi=
a
> servers I think that would be powerful. In the past I have done things li=
ke
> created a "B2B-like" entity with a Jingle stack and a SIP stack - but the=
n I
> end up with identity issues and issues with simultaneous sessions to medi=
a
> servers.
>=20
> I agree that a very simple call state machine is desired - even if the SI=
P
> mechanics need to be strongly restricted. I also agree that SDP is gettin=
g a
> bit tired.=A0
>=20
> Peter
>=20
> On Thu, Feb 4, 2010 at 8:36 AM, Alan Johnston <alan.b.johnston@gmail.com>
> wrote:
>> John,
>>=20
>> Thanks for your comments.
>>=20
>> We are not thinking about just an XML encoding for SIP - you are
>> correct, that would still expose way too much complexity. =A0We were
>> thinking about starting with the functionality in RFC 5638, then see how
>> much we can abstract and separate the "plumbing" of SIP from what an app
>> developer needs/cares about. =A0We hope to avoid replicating the entire
>> SIP state machine.
>>=20
>> However, this is just a starting point, and feedback and input from
>> others who share our goal is most welcome.
>>=20
>> - Alan -
>>=20
>> On 2/4/10 3:43 AM, Elwell, John wrote:
>>> > Henry,
>>> >
>>> > Where the draft talks about "porting core SIP functions to standards =
XML
>>> based API" (section 4.5), can you shed some more light on this? Is it j=
ust a
>>> case of adopting existing SIP semantics and defining a new syntax (XML)=
? If
>>> so, this would seem to require the web application to support the same
>>> complex SIP state machine, with its notions of transactions, dialogs, e=
arly
>>> dialogs, backwards compatibility capabilities, tolerance of unreliable
>>> transports, offer-answer rules, and at least some of the many SIP
>>> extensions. Or had you intended some sort of simplification, and if so =
can
>>> you give more detail?
>>> >
>>> > Thanks,
>>> >
>>> > John
>>> >
>>> >
>>>> >> -----Original Message-----
>>>> >> From: dispatch-bounces@ietf.org
>>>> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
>>>> >> Sent: 03 February 2010 18:18
>>>> >> To: dispatch mailing list
>>>> >> Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.=
txt
>>>> >>
>>>> >> We have published an I-D on "SIP APIs for Web Communications"
>>>> >> and would
>>>> >> appreciate any comments and input.
>>>> >>
>>>> >> Here is the link and the announcement:
>>>> >>
>>>> >> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>>>> >> is-00.txt
>>>> >>
>>>> >> A New Internet-Draft is available from the on-line Internet-Drafts
>>>> >> directories.
>>>> >>
>>>> >>
>>>> >> =A0Title =A0: SIP APIs for Communications on the Web
>>>> >> =A0Author(s) : H. Sinnreich, A. Johnston
>>>> >> =A0Filename : draft-sinnreich-sip-web-apis-00.txt
>>>> >> =A0Pages =A0: 19
>>>> >> =A0Date =A0: 2010-1-19
>>>> >>
>>>> >> =A0 =A0Interactive communications are becoming part of rich Internet
>>>> >> =A0 =A0applications (RIA). This can be accomplished entirely by using W=
eb
>>>> >> =A0 =A0technology and developing it as an Internet standard. Voice over=
 IP
>>>> >> =A0 =A0(VoIP) features developed for Session Initiation Protocol (SIP 2=
.0)
>>>> >> =A0 =A0can be preserved in rich multimedia communications and applicati=
ons
>>>> >> =A0 =A0on the web. =A0Hyper Text Transport Protocol (HTTP) is used
>>>> >> for control
>>>> >> =A0 =A0and signaling data and RTP for interactive media transport
>>>> >> =A0 =A0respectively. No other network application protocols are require=
d.
>>>> >> =A0 =A0Web, fixed and mobile device application development tools can t=
hus
>>>> >> =A0 =A0be used to include components and widgets for Internet presence,
>>>> >> =A0 =A0Instant Messaging (IM), voice and video. =A0This opens Internet
>>>> >> =A0 =A0communications, including voice to application developers and wi=
ll
>>>> >> =A0 =A0also enhance the user experience with real-time Web and mobile
>>>> >> =A0 =A0communications. =A0Usage scenarios include service providers, priv=
ate
>>>> >> =A0 =A0IP networks, device application developers and media companies
>>>> >> =A0 =A0providing a mix of integrated communications, applications
>>>> >> and media.
>>>> >> =A0 =A0This memo argues for the development and standardization of
>>>> >> =A0 =A0Application Programming Interfaces (APIs) to allow the benefits =
of
>>>> >> =A0 =A0SIP to be used by Rich Internet Applications (RIA), for fixed an=
d
>>>> >> =A0 =A0mobile devices.
>>>> >>
>>>> >> A URL for this Internet-Draft is:
>>>> >> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>>>> >> is-00.txt
>>>> >>
>>>> >> ------ End of Forwarded Message
>>>> >>
>>>> >> Thanks in advance for your comments.
>>>> >>
>>>> >> Henry
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> dispatch mailing list
>>>> >> dispatch@ietf.org
>>>> >> https://www.ietf.org/mailman/listinfo/dispatch
>>>> >>
>>> > _______________________________________________
>>> > dispatch mailing list
>>> > dispatch@ietf.org
>>> > https://www.ietf.org/mailman/listinfo/dispatch
>>> >
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>=20
>=20


--B_3348123066_1518226
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<HTML>
<HEAD>
<TITLE>Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt</TITLE=
>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>Peter,<BR>
<BR>
&gt;e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP=
-like media constructs, uses ICE and if I recall correctly does put address =
inside messages).=A0<BR>
<BR>
As mentioned, our expertise and focus here is in SIP and similar APIs for J=
ingle would give developers a bigger choice, depending on the usage scenario=
.<BR>
As for ICE, we argue it to work in HIP for all application protocols alike,=
 besides all the other reasons for using HIP.<BR>
<BR>
&gt;If I had a web API which allowed direct connections to SIP endpoints/me=
dia servers I think that would be powerful.<BR>
Yes, since all or most of the telecom gear supports SIP, we believe there a=
re lots of such apps.<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 2/4/10 8:16 AM, &quot;Peter Musgrave&quot; &lt;<a href=3D"peter.musgrave@m=
agorcorp.com">peter.musgrave@magorcorp.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'>Henry/Alan,=A0<BR>
<BR>
I think some mention of other web technologies in the draft might help fram=
e the effort.<BR>
<BR>
e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP-lik=
e media constructs, uses ICE and if I recall correctly does put address insi=
de messages).=A0<BR>
<BR>
If I had a web API which allowed direct connections to SIP endpoints/media =
servers I think that would be powerful. In the past I have done things like =
created a &quot;B2B-like&quot; entity with a Jingle stack and a SIP stack - =
but then I end up with identity issues and issues with simultaneous sessions=
 to media servers.<BR>
<BR>
I agree that a very simple call state machine is desired - even if the SIP =
mechanics need to be strongly restricted. I also agree that SDP is getting a=
 bit tired.=A0<BR>
<BR>
Peter<BR>
<BR>
On Thu, Feb 4, 2010 at 8:36 AM, Alan Johnston &lt;<a href=3D"alan.b.johnston@=
gmail.com">alan.b.johnston@gmail.com</a>&gt; wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'>John,<BR>
<BR>
Thanks for your comments.<BR>
<BR>
We are not thinking about just an XML encoding for SIP - you are<BR>
correct, that would still expose way too much complexity. =A0We were<BR>
thinking about starting with the functionality in RFC 5638, then see how<BR=
>
much we can abstract and separate the &quot;plumbing&quot; of SIP from what=
 an app<BR>
developer needs/cares about. =A0We hope to avoid replicating the entire<BR>
SIP state machine.<BR>
<BR>
However, this is just a starting point, and feedback and input from<BR>
others who share our goal is most welcome.<BR>
<FONT COLOR=3D"#888888"><BR>
- Alan -<BR>
</FONT><BR>
On 2/4/10 3:43 AM, Elwell, John wrote:<BR>
&gt; Henry,<BR>
&gt;<BR>
&gt; Where the draft talks about &quot;porting core SIP functions to standa=
rds XML based API&quot; (section 4.5), can you shed some more light on this?=
 Is it just a case of adopting existing SIP semantics and defining a new syn=
tax (XML)? If so, this would seem to require the web application to support =
the same complex SIP state machine, with its notions of transactions, dialog=
s, early dialogs, backwards compatibility capabilities, tolerance of unrelia=
ble transports, offer-answer rules, and at least some of the many SIP extens=
ions. Or had you intended some sort of simplification, and if so can you giv=
e more detail?<BR>
&gt;<BR>
&gt; Thanks,<BR>
&gt;<BR>
&gt; John<BR>
&gt;<BR>
&gt;<BR>
&gt;&gt; -----Original Message-----<BR>
&gt;&gt; From: <a href=3D"dispatch-bounces@ietf.org">dispatch-bounces@ietf.or=
g</a><BR>
&gt;&gt; [<a href=3D"mailto:dispatch-bounces@ietf.org">mailto:dispatch-bounce=
s@ietf.org</a>] On Behalf Of Henry Sinnreich<BR>
&gt;&gt; Sent: 03 February 2010 18:18<BR>
&gt;&gt; To: dispatch mailing list<BR>
&gt;&gt; Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00=
.txt<BR>
&gt;&gt;<BR>
&gt;&gt; We have published an I-D on &quot;SIP APIs for Web Communications&=
quot;<BR>
&gt;&gt; and would<BR>
&gt;&gt; appreciate any comments and input.<BR>
&gt;&gt;<BR>
&gt;&gt; Here is the link and the announcement:<BR>
&gt;&gt;<BR>
&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip-w=
eb-ap">http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap</a><BR=
>
&gt;&gt; is-00.txt<BR>
&gt;&gt;<BR>
&gt;&gt; A New Internet-Draft is available from the on-line Internet-Drafts=
<BR>
&gt;&gt; directories.<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; =A0Title =A0: SIP APIs for Communications on the Web<BR>
&gt;&gt; =A0Author(s) : H. Sinnreich, A. Johnston<BR>
&gt;&gt; =A0Filename : draft-sinnreich-sip-web-apis-00.txt<BR>
&gt;&gt; =A0Pages =A0: 19<BR>
&gt;&gt; =A0Date =A0: 2010-1-19<BR>
&gt;&gt;<BR>
&gt;&gt; =A0 =A0Interactive communications are becoming part of rich Internet<B=
R>
&gt;&gt; =A0 =A0applications (RIA). This can be accomplished entirely by using =
Web<BR>
&gt;&gt; =A0 =A0technology and developing it as an Internet standard. Voice ove=
r IP<BR>
&gt;&gt; =A0 =A0(VoIP) features developed for Session Initiation Protocol (SIP =
2.0)<BR>
&gt;&gt; =A0 =A0can be preserved in rich multimedia communications and applicat=
ions<BR>
&gt;&gt; =A0 =A0on the web. =A0Hyper Text Transport Protocol (HTTP) is used<BR>
&gt;&gt; for control<BR>
&gt;&gt; =A0 =A0and signaling data and RTP for interactive media transport<BR>
&gt;&gt; =A0 =A0respectively. No other network application protocols are requir=
ed.<BR>
&gt;&gt; =A0 =A0Web, fixed and mobile device application development tools can =
thus<BR>
&gt;&gt; =A0 =A0be used to include components and widgets for Internet presence=
,<BR>
&gt;&gt; =A0 =A0Instant Messaging (IM), voice and video. =A0This opens Internet<B=
R>
&gt;&gt; =A0 =A0communications, including voice to application developers and w=
ill<BR>
&gt;&gt; =A0 =A0also enhance the user experience with real-time Web and mobile<=
BR>
&gt;&gt; =A0 =A0communications. =A0Usage scenarios include service providers, pri=
vate<BR>
&gt;&gt; =A0 =A0IP networks, device application developers and media companies<=
BR>
&gt;&gt; =A0 =A0providing a mix of integrated communications, applications<BR>
&gt;&gt; and media.<BR>
&gt;&gt; =A0 =A0This memo argues for the development and standardization of<BR>
&gt;&gt; =A0 =A0Application Programming Interfaces (APIs) to allow the benefits=
 of<BR>
&gt;&gt; =A0 =A0SIP to be used by Rich Internet Applications (RIA), for fixed a=
nd<BR>
&gt;&gt; =A0 =A0mobile devices.<BR>
&gt;&gt;<BR>
&gt;&gt; A URL for this Internet-Draft is:<BR>
&gt;&gt; <a href=3D"http://www.ietf.org/internet-drafts/draft-sinnreich-sip-w=
eb-ap">http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap</a><BR=
>
&gt;&gt; is-00.txt<BR>
&gt;&gt;<BR>
&gt;&gt; ------ End of Forwarded Message<BR>
&gt;&gt;<BR>
&gt;&gt; Thanks in advance for your comments.<BR>
&gt;&gt;<BR>
&gt;&gt; Henry<BR>
&gt;&gt;<BR>
&gt;&gt;<BR>
&gt;&gt; _______________________________________________<BR>
&gt;&gt; dispatch mailing list<BR>
&gt;&gt; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://w=
ww.ietf.org/mailman/listinfo/dispatch</a><BR>
&gt;&gt;<BR>
&gt; _______________________________________________<BR>
&gt; dispatch mailing list<BR>
&gt; <a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.i=
etf.org/mailman/listinfo/dispatch</a><BR>
&gt;<BR>
_______________________________________________<BR>
dispatch mailing list<BR>
<a href=3D"dispatch@ietf.org">dispatch@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/dispatch">https://www.ietf.o=
rg/mailman/listinfo/dispatch</a><BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'><BR>
<BR>
</SPAN></FONT></BLOCKQUOTE>
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Henry wrote
> Peter Musgrave wrote:
>> If I had a web API which allowed direct connections to SIP =
endpoints/media servers I think that would be powerful.
> Yes, since all or most of the telecom gear supports SIP, we believe =
there are lots of such apps.

do you think of something like this: =
http://www.developergarden.com/startseite ? It's a REST-based API to set =
up voice calls or conferences. It's an attempt to hide SIP's complexity =
from web developers.=20
=20

Wolfgang Beck

=20

=20

Deutsche Telekom Netzproduktion GmbH
Zentrum Technik Einf=FChrung
Wolfgang Beck
Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
+49 6151 628 2832 (Tel.)
www.telekom.com=20

Erleben, was verbindet.=20

=20

Deutsche Telekom Netzproduktion GmbH
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
Handelsregister: Amtsgericht Bonn HRB 14190
Sitz der Gesellschaft: Bonn
USt-IdNr. DE 814645262

=20

=20

________________________________

Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im =
Auftrag von Henry Sinnreich
Gesendet: Donnerstag, 4. Februar 2010 17:11
An: Peter Musgrave; Alan Johnston
Cc: dispatch mailing list
Betreff: Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt


Peter,

>e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has =
SDP-like media constructs, uses ICE and if I recall correctly does put =
address inside messages).=20

As mentioned, our expertise and focus here is in SIP and similar APIs =
for Jingle would give developers a bigger choice, depending on the usage =
scenario.
As for ICE, we argue it to work in HIP for all application protocols =
alike, besides all the other reasons for using HIP.

>If I had a web API which allowed direct connections to SIP =
endpoints/media servers I think that would be powerful.
Yes, since all or most of the telecom gear supports SIP, we believe =
there are lots of such apps.

Henry




	=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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ACTION:draft-sinnreich-sip-web-apis-00.txt</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3660" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DCalibri size=3D4><SPAN=20
class=3D884121416-04022010>Henry wrote</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DCalibri size=3D4><SPAN=20
class=3D884121416-04022010>&gt; Peter Musgrave =
wrote:</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><FONT face=3DCalibri=20
color=3D#000000 size=3D4>&gt;<SPAN =
class=3D884121416-04022010>&gt;</SPAN><SPAN=20
class=3D884121416-04022010> </SPAN>If I had a web API which allowed =
direct=20
connections to SIP endpoints/media servers I think that would be=20
powerful.<BR><SPAN class=3D884121416-04022010>&gt; </SPAN>Yes, since all =
or most=20
of the telecom gear supports SIP, we believe there are lots of such=20
apps.<BR></FONT></FONT><FONT face=3DArial color=3D#0000ff =
size=3D2></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D884121416-04022010><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>do you think of something like this: <A=20
href=3D"http://www.developergarden.com/startseite">http://www.developerga=
rden.com/startseite</A>&nbsp;?=20
It's a REST-based API to set up voice calls or conferences. It's an =
attempt to=20
hide SIP's complexity from web developers. </FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV class=3DSection1>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: Arial; =
mso-ansi-language: EN-GB">Wolfgang=20
Beck</SPAN><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><?xml:namespace=20
prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office"=20
/><o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">&nbsp;<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">&nbsp;<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; COLOR: #979797; FONT-FAMILY: Arial; =
mso-ansi-language: EN-GB">Deutsche=20
Telekom Netzproduktion GmbH<BR>Zentrum Technik Einf=FChrung<BR>Wolfgang=20
Beck<BR>Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt<BR>+49 6151 628 =
2832=20
(Tel.)<BR><A href=3D"http:www.telekom.com">www.telekom.com</A> =
</SPAN><SPAN=20
lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; COLOR: #e20074; FONT-FAMILY: Arial; =
mso-ansi-language: EN-GB"><BR><BR>Erleben,=20
was verbindet.</SPAN><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; COLOR: olive; FONT-FAMILY: Arial; =
mso-ansi-language: EN-GB">=20
</SPAN><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB=20
style=3D"FONT-SIZE: 8pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">&nbsp;<o:p></o:p></SPAN></P>
<P style=3D"MARGIN: 0cm 0cm 0pt"><SPAN=20
style=3D"FONT-SIZE: 8pt; COLOR: #979797; FONT-FAMILY: Arial">Deutsche =
Telekom=20
Netzproduktion GmbH<BR>Aufsichtsrat: Dr. Steffen Roehn=20
(Vorsitzender)<BR>Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn =
(Vorsitzender),=20
Albert Matheis, Klaus Peren<BR>Handelsregister: Amtsgericht Bonn HRB=20
14190<BR>Sitz der Gesellschaft: Bonn<BR>USt-IdNr. DE =
814645262</SPAN><SPAN=20
style=3D"FONT-SIZE: 8pt; FONT-FAMILY: Arial"><o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></P></DIV>
<DIV>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Dde dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>Von:</B> dispatch-bounces@ietf.org=20
[mailto:dispatch-bounces@ietf.org] <B>Im Auftrag von </B>Henry=20
Sinnreich<BR><B>Gesendet:</B> Donnerstag, 4. Februar 2010 =
17:11<BR><B>An:</B>=20
Peter Musgrave; Alan Johnston<BR><B>Cc:</B> dispatch mailing=20
list<BR><B>Betreff:</B> Re: [dispatch] I-D=20
ACTION:draft-sinnreich-sip-web-apis-00.txt<BR></FONT><BR></DIV>
<DIV></DIV><FONT face=3D"Calibri, Verdana, Helvetica, Arial"><SPAN=20
style=3D"FONT-SIZE: 13pt">Peter,<BR><BR>&gt;e.g. How does a SIP API =
compare to=20
e.g. Jingle (which uses RTP, has SDP-like media constructs, uses ICE and =
if I=20
recall correctly does put address inside messages).&nbsp;<BR><BR>As =
mentioned,=20
our expertise and focus here is in SIP and similar APIs for Jingle would =
give=20
developers a bigger choice, depending on the usage scenario.<BR>As for =
ICE, we=20
argue it to work in HIP for all application protocols alike, besides all =
the=20
other reasons for using HIP.<BR><BR>&gt;If I had a web API which allowed =
direct=20
connections to SIP endpoints/media servers I think that would be=20
powerful.<BR>Yes, since all or most of the telecom gear supports SIP, we =
believe=20
there are lots of such apps.<BR><BR>Henry<BR><BR><BR></SPAN></FONT>
<BLOCKQUOTE><FONT =
face=3DCalibri></FONT>&nbsp;</BLOCKQUOTE></BODY></HTML>

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Date: Thu, 04 Feb 2010 10:36:05 -0600
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: <BeckW@telekom.de>, <peter.musgrave@magorcorp.com>, Alan Johnston <alan.b.johnston@gmail.com>
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http://www.developergarden.com/startseite

Shows we don=B9t have a monopoly in trying to hide SIP complexity from
developers.
Examples like Conf Call light are very convincing.

The next step is allowing Web application developers to mix and match RIA
and SIP-like apps in a seamless manner within the same IDE.
Or has someone done it already as well?

Henry


On 2/4/10 10:18 AM, "BeckW@telekom.de" <BeckW@telekom.de> wrote:

> Henry wrote
>> > Peter Musgrave wrote:
>>> >> If I had a web API which allowed direct connections to SIP
>>> endpoints/media servers I think that would be powerful.
>> > Yes, since all or most of the telecom gear supports SIP, we believe th=
ere
>> are lots of such apps.
> do you think of something like this: http://www.developergarden.com/start=
seite
> ? It's a REST-based API to set up voice calls or conferences. It's an att=
empt
> to hide SIP's complexity from web developers.
> =20
> Wolfgang Beck
>=20
> =20
>=20
> =20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Zentrum Technik Einf=FChrung
> Wolfgang Beck
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628 2832 (Tel.)
> www.telekom.com <http:www.telekom.com>
>=20
> Erleben, was verbindet.
>=20
> =20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert Matheis=
,
> Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> =20
> =20
>=20
>=20
> Von: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] Im Auft=
rag
> von Henry Sinnreich
> Gesendet: Donnerstag, 4. Februar 2010 17:11
> An: Peter Musgrave; Alan Johnston
> Cc: dispatch mailing list
> Betreff: Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
>=20
> Peter,
>=20
>> >e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP=
-like
>> media constructs, uses ICE and if I recall correctly does put address in=
side
>> messages).=20
>=20
> As mentioned, our expertise and focus here is in SIP and similar APIs for
> Jingle would give developers a bigger choice, depending on the usage scen=
ario.
> As for ICE, we argue it to work in HIP for all application protocols alik=
e,
> besides all the other reasons for using HIP.
>=20
>> >If I had a web API which allowed direct connections to SIP endpoints/me=
dia
>> servers I think that would be powerful.
> Yes, since all or most of the telecom gear supports SIP, we believe there=
 are
> lots of such apps.
>=20
> Henry
>=20
>=20
>> =20
>=20


--B_3348124569_1619064
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<HTML>
<HEAD>
<TITLE>Re: AW: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt</T=
ITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'><a href=3D"http://www.developergarden.com/startseite">http://www.developerga=
rden.com/startseite</a><BR>
<BR>
Shows we don&#8217;t have a monopoly in trying to hide SIP complexity from =
developers.<BR>
Examples like Conf Call light are very convincing.<BR>
<BR>
The next step is allowing Web application developers to mix and match RIA a=
nd SIP-like apps in a seamless manner within the same IDE.<BR>
Or has someone done it already as well?<BR>
<BR>
Henry<BR>
<BR>
<BR>
On 2/4/10 10:18 AM, &quot;<a href=3D"BeckW@telekom.de">BeckW@telekom.de</a>&q=
uot; &lt;<a href=3D"BeckW@telekom.de">BeckW@telekom.de</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
FONT SIZE=3D"5"><SPAN STYLE=3D'font-size:24pt'>Henry wrote<BR>
&gt; Peter Musgrave wrote:<BR>
&gt;&gt;</SPAN></FONT><SPAN STYLE=3D'font-size:13pt'> If I had a web API whic=
h allowed direct connections to SIP endpoints/media servers I think that wou=
ld be powerful.<BR>
&gt; Yes, since all or most of the telecom gear supports SIP, we believe th=
ere are lots of such apps.<BR>
</SPAN></FONT><SPAN STYLE=3D'font-size:13pt'><FONT COLOR=3D"#0000FF"><FONT FACE=
=3D"Arial">do you think of something like this: <a href=3D"http://www.developerg=
arden.com/startseite">http://www.developergarden.com/startseite</a> ? It's a=
 REST-based API to set up voice calls or conferences. It's an attempt to hid=
e SIP's complexity from web developers. <BR>
</FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN><FONT SIZE=3D"1"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:10pt=
'>Wolfgang Beck<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:8pt'=
> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:8pt'=
> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#979797"><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:8pt'>Deutsche Telekom Netzproduktion GmbH<BR>
Zentrum Technik Einf&uuml;hrung<BR>
Wolfgang Beck<BR>
Heinrich-Hertz-Stra&szlig;e 3-7, 64295 Darmstadt<BR>
+49 6151 628 2832 (Tel.)<BR>
www.telekom.com &lt;<a href=3D"http:www.telekom.com">http:www.telekom.com</a>=
&gt; &nbsp;<BR>
</SPAN></FONT></FONT></FONT><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN STYLE=3D'=
font-size:8pt'><FONT COLOR=3D"#E20074"><BR>
Erleben, was verbindet.</FONT><FONT COLOR=3D"#808000"> <BR>
</FONT></SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:8pt'=
> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT COLOR=3D"#979797"><FONT SIZE=3D"0"><FONT FACE=3D"Arial"><SPAN=
 STYLE=3D'font-size:8pt'>Deutsche Telekom Netzproduktion GmbH<BR>
Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)<BR>
Gesch&auml;ftsf&uuml;hrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert=
 Matheis, Klaus Peren<BR>
Handelsregister: Amtsgericht Bonn HRB 14190<BR>
Sitz der Gesellschaft: Bonn<BR>
USt-IdNr. DE 814645262<BR>
</SPAN></FONT></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"=
><SPAN STYLE=3D'font-size:13pt'><BR>
</SPAN></FONT><FONT SIZE=3D"1"><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:10pt=
'> <BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'> <BR>
<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"100%"></SPAN></FONT><SPAN STYLE=3D'font-size=
:13pt'><FONT FACE=3D"Tahoma, Verdana, Helvetica, Arial"><B>Von:</B> <a href=3D"d=
ispatch-bounces@ietf.org">dispatch-bounces@ietf.org</a> [<a href=3D"mailto:dis=
patch-bounces@ietf.org">mailto:dispatch-bounces@ietf.org</a>] <B>Im Auftrag =
von </B>Henry Sinnreich<BR>
<B>Gesendet:</B> Donnerstag, 4. Februar 2010 17:11<BR>
<B>An:</B> Peter Musgrave; Alan Johnston<BR>
<B>Cc:</B> dispatch mailing list<BR>
<B>Betreff:</B> Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.t=
xt<BR>
</FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><BR>
Peter,<BR>
<BR>
&gt;e.g. How does a SIP API compare to e.g. Jingle (which uses RTP, has SDP=
-like media constructs, uses ICE and if I recall correctly does put address =
inside messages). <BR>
<BR>
As mentioned, our expertise and focus here is in SIP and similar APIs for J=
ingle would give developers a bigger choice, depending on the usage scenario=
.<BR>
As for ICE, we argue it to work in HIP for all application protocols alike,=
 besides all the other reasons for using HIP.<BR>
<BR>
&gt;If I had a web API which allowed direct connections to SIP endpoints/me=
dia servers I think that would be powerful.<BR>
Yes, since all or most of the telecom gear supports SIP, we believe there a=
re lots of such apps.<BR>
<BR>
Henry<BR>
<BR>
<BR>
</FONT></SPAN><BLOCKQUOTE><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri,=
 Verdana, Helvetica, Arial"> <BR>
</FONT></SPAN></BLOCKQUOTE><SPAN STYLE=3D'font-size:13pt'><FONT FACE=3D"Calibri=
, Verdana, Helvetica, Arial"><BR>
</FONT></SPAN></BLOCKQUOTE>
</BODY>
</HTML>


--B_3348124569_1619064--



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Subject: Re: [dispatch] I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
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I like this draft. I find it much more to the point than was the simple 
sip draft. And it makes the case much better.

I agree with John that unless some simplification is done, a "web api" 
won't be any simpler than what we have.

The key to that is coming up with a higher level abstraction, that has 
all the degrees of freedom that are needed, while hiding everything 
else. This is very hard to do. And doing so successfully requires very 
carefully identifying the audience.

ISTM that developing such a new level of abstraction is a prerequisite 
to other progress on this front. I'm uncertain if IETF is or is not the 
right place to do that.

	Thanks,
	Paul

Elwell, John wrote:
> Henry,
> 
> Where the draft talks about "porting core SIP functions to standards XML based API" (section 4.5), can you shed some more light on this? Is it just a case of adopting existing SIP semantics and defining a new syntax (XML)? If so, this would seem to require the web application to support the same complex SIP state machine, with its notions of transactions, dialogs, early dialogs, backwards compatibility capabilities, tolerance of unreliable transports, offer-answer rules, and at least some of the many SIP extensions. Or had you intended some sort of simplification, and if so can you give more detail?
> 
> Thanks,
> 
> John
> 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Henry Sinnreich
>> Sent: 03 February 2010 18:18
>> To: dispatch mailing list
>> Subject: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
>>
>> We have published an I-D on "SIP APIs for Web Communications" 
>> and would
>> appreciate any comments and input.
>>
>> Here is the link and the announcement:
>>
>> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>> is-00.txt
>>
>> A New Internet-Draft is available from the on-line Internet-Drafts
>> directories.
>>
>>
>>  Title  : SIP APIs for Communications on the Web
>>  Author(s) : H. Sinnreich, A. Johnston
>>  Filename : draft-sinnreich-sip-web-apis-00.txt
>>  Pages  : 19
>>  Date  : 2010-1-19
>>  
>>    Interactive communications are becoming part of rich Internet
>>    applications (RIA). This can be accomplished entirely by using Web
>>    technology and developing it as an Internet standard. Voice over IP
>>    (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>>    can be preserved in rich multimedia communications and applications
>>    on the web.  Hyper Text Transport Protocol (HTTP) is used 
>> for control
>>    and signaling data and RTP for interactive media transport
>>    respectively. No other network application protocols are required.
>>    Web, fixed and mobile device application development tools can thus
>>    be used to include components and widgets for Internet presence,
>>    Instant Messaging (IM), voice and video.  This opens Internet
>>    communications, including voice to application developers and will
>>    also enhance the user experience with real-time Web and mobile
>>    communications.  Usage scenarios include service providers, private
>>    IP networks, device application developers and media companies
>>    providing a mix of integrated communications, applications 
>> and media.
>>    This memo argues for the development and standardization of
>>    Application Programming Interfaces (APIs) to allow the benefits of
>>    SIP to be used by Rich Internet Applications (RIA), for fixed and
>>    mobile devices.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-ap
>> is-00.txt
>>
>> ------ End of Forwarded Message
>>
>> Thanks in advance for your comments.
>>
>> Henry
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From lorenzo@meetecho.com  Thu Feb  4 08:44:15 2010
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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Leon Portman <Leon.Portman@nice.com>
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, zhouzhipeng 54906 <zhouzp@huawei.com>
Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Hi Leon and Zhipeng,

just for your knowledge, I remind the ML of the existence of a draft we
wrote a few months ago that is related to session recording and
playback:

http://tools.ietf.org/html/draft-romano-dcon-recording-01

It is not currently based on web services or RESTful interfaces, but it
still is a standard approach, since it makes use of SMIL to synchronize
the recorded streams and play them back to interested users. In case
anyone is interested in working on a document related to this topic just
let us know.

Lorenzo


On Thu, 4 Feb 2010 18:05:33 +0200
Leon Portman <Leon.Portman@nice.com> wrote:

> Zhipeng, Hi
> 
> The core point of this document is to define a SIP based protocol that will allow Active VoIP recording, where a some entity from communication system (gateway, phone, media server) will fork media of the call to recording server.
> 
> The interface for querying and playback of recorded sessions are currently out of scope of this work because it probably should be a web service or REST interface. However we are definitely considering to work on it as well.
> 
> Regards
> 
> Leon
> 
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behalf Of zhouzhipeng 54906
> Sent: Wednesday, February 03, 2010 11:52 AM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
> 
> Dear all,
> I have got much valuable information from related emails.
> As I cited as following,
> 
> > [rj] This can range anywhere from proprietary VoIP media and 
> > signaling protocols to things such as SCCP, UNISTIM, H.323, TDM 
> > and analog phones. Most such phones are not SIP UAs. Some other 
> > entity (such as a SIP-enabled mixer) may provide the RC media 
> > forking point functionality in these cases.
> 
> there have been many years for the application of recording. 
> So, may the author give us a brief summary what have been advanced based on current recording solutions, for example the call center architecture. So we can easily catch the core point of this document.
> 
> Another point is: does the diagram need to add the interface between UA to the RS for the search or query functions.
> 
> Thanks very much.
> Zhipeng
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 
Lorenzo Miniero
Meetecho s.r.l.
http://www.meetecho.com/

From Leon.Portman@nice.com  Thu Feb  4 08:49:04 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: Lorenzo Miniero <lorenzo@meetecho.com>
Date: Thu, 4 Feb 2010 18:49:47 +0200
Thread-Topic: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-req-00
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Lorenzo, Hi

It will be very important to continue this work for standardization how rec=
ordings will be represented, specially for complex scenarios with multiple =
medias and segments

Regards

Leon=20

-----Original Message-----
From: Lorenzo Miniero [mailto:lorenzo@meetecho.com]=20
Sent: Thursday, February 04, 2010 6:44 PM
To: Leon Portman
Cc: zhouzhipeng 54906; dispatch@ietf.org
Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording-r=
eq-00

Hi Leon and Zhipeng,

just for your knowledge, I remind the ML of the existence of a draft we
wrote a few months ago that is related to session recording and
playback:

http://tools.ietf.org/html/draft-romano-dcon-recording-01

It is not currently based on web services or RESTful interfaces, but it
still is a standard approach, since it makes use of SMIL to synchronize
the recorded streams and play them back to interested users. In case
anyone is interested in working on a document related to this topic just
let us know.

Lorenzo


On Thu, 4 Feb 2010 18:05:33 +0200
Leon Portman <Leon.Portman@nice.com> wrote:

> Zhipeng, Hi
>=20
> The core point of this document is to define a SIP based protocol that wi=
ll allow Active VoIP recording, where a some entity from communication syst=
em (gateway, phone, media server) will fork media of the call to recording =
server.
>=20
> The interface for querying and playback of recorded sessions are currentl=
y out of scope of this work because it probably should be a web service or =
REST interface. However we are definitely considering to work on it as well=
.
>=20
> Regards
>=20
> Leon
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf Of zhouzhipeng 54906
> Sent: Wednesday, February 03, 2010 11:52 AM
> To: dispatch@ietf.org
> Subject: Re: [dispatch] Comments ondraft-rehor-dispatch-session-recording=
-req-00
>=20
> Dear all,
> I have got much valuable information from related emails.
> As I cited as following,
>=20
> > [rj] This can range anywhere from proprietary VoIP media and=20
> > signaling protocols to things such as SCCP, UNISTIM, H.323, TDM=20
> > and analog phones. Most such phones are not SIP UAs. Some other=20
> > entity (such as a SIP-enabled mixer) may provide the RC media=20
> > forking point functionality in these cases.
>=20
> there have been many years for the application of recording.=20
> So, may the author give us a brief summary what have been advanced based =
on current recording solutions, for example the call center architecture. S=
o we can easily catch the core point of this document.
>=20
> Another point is: does the diagram need to add the interface between UA t=
o the RS for the search or query functions.
>=20
> Thanks very much.
> Zhipeng
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


--=20
Lorenzo Miniero
Meetecho s.r.l.
http://www.meetecho.com/

From xavier.marjou@gmail.com  Thu Feb  4 09:00:39 2010
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Subject: [dispatch] When will UUI WG be created?
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Hi,

The first two WG proposals of IETF75 have been created (codec,
sipclf), but what about Call Control User-to-User Information WG? When
will it be created?

Thank you,
Xavier



On Thu, Jan 21, 2010 at 7:56 PM, Mary Barnes <mary.barnes@nortel.com> wrote=
:
> Hi all,
>
> I've added a summary of the dispatched items to the Wiki on the DISPATCH
> WG tools page:
> http://trac.tools.ietf.org/wg/dispatch/trac/wiki
>
> I've included both the work that has been chartered or completely
> dispatched along with that for which the chartering work is still
> underway. I'll update it with the appropriate WG information as it
> becomes available.
>
> Please let me know if the format is useful or if you find any
> inaccuracies.
>
> Thanks,
> Mary.
>
> -----Original Message-----
> From: Barnes, Mary (RICH2:AR00)
> Sent: Wednesday, December 09, 2009 9:33 AM
> To: Brett Tate; dispatch@ietf.org
> Subject: RE: [dispatch] locating spawns of DISPATCH
>
> Hi Brett,
>
> That's a good point. I had been summarizing such in the status emails,
> but can see the value in keeping this in a single place. I'll add a page
> to the supplementary webpage on softarmor and send a note to the list
> when that's done.
>
> Thanks,
> Mary.
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Brett Tate
> Sent: Wednesday, December 09, 2009 7:04 AM
> To: dispatch@ietf.org
> Subject: [dispatch] locating spawns of DISPATCH
>
> Greetings,
>
> Are the working group spawns of DISPATCH collected somewhere to easily
> notice them (without searching email archive)? =A0For instance, will they
> be linked somehow to DISPATCH charter, status, or a supplemental
> DISPATCH webpage?
>
> Thanks,
> Brett
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From tom111.taylor@bell.net  Thu Feb  4 09:46:58 2010
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Joerg and his student certainly tried, with SDPng. Maybe with the rise of XML 
SDPng's time has come.

Paul Kyzivat wrote:
> 
> 
> Peter Musgrave wrote:
>>  I also agree that SDP is getting a bit tired. 
> 
> SDP was tired in 2000. It should have been buried long ago.
> 
>     Paul
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 
> 

From gonzalo.camarillo@ericsson.com  Thu Feb  4 11:17:31 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [dispatch] When will UUI WG be created?
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Hi,

as DISPATCH chairs, once we send a charter proposal to the ADs, it is in
their hands. You can check with them (I know they have been quite busy).

Cheers,

Gonzalo


Xavier Marjou wrote:
> Hi,
> 
> The first two WG proposals of IETF75 have been created (codec,
> sipclf), but what about Call Control User-to-User Information WG? When
> will it be created?
> 
> Thank you,
> Xavier
> 
> 
> 
> On Thu, Jan 21, 2010 at 7:56 PM, Mary Barnes <mary.barnes@nortel.com> wrote:
>> Hi all,
>>
>> I've added a summary of the dispatched items to the Wiki on the DISPATCH
>> WG tools page:
>> http://trac.tools.ietf.org/wg/dispatch/trac/wiki
>>
>> I've included both the work that has been chartered or completely
>> dispatched along with that for which the chartering work is still
>> underway. I'll update it with the appropriate WG information as it
>> becomes available.
>>
>> Please let me know if the format is useful or if you find any
>> inaccuracies.
>>
>> Thanks,
>> Mary.
>>
>> -----Original Message-----
>> From: Barnes, Mary (RICH2:AR00)
>> Sent: Wednesday, December 09, 2009 9:33 AM
>> To: Brett Tate; dispatch@ietf.org
>> Subject: RE: [dispatch] locating spawns of DISPATCH
>>
>> Hi Brett,
>>
>> That's a good point. I had been summarizing such in the status emails,
>> but can see the value in keeping this in a single place. I'll add a page
>> to the supplementary webpage on softarmor and send a note to the list
>> when that's done.
>>
>> Thanks,
>> Mary.
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Brett Tate
>> Sent: Wednesday, December 09, 2009 7:04 AM
>> To: dispatch@ietf.org
>> Subject: [dispatch] locating spawns of DISPATCH
>>
>> Greetings,
>>
>> Are the working group spawns of DISPATCH collected somewhere to easily
>> notice them (without searching email archive)?  For instance, will they
>> be linked somehow to DISPATCH charter, status, or a supplemental
>> DISPATCH webpage?
>>
>> Thanks,
>> Brett
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>


From hgs@cs.columbia.edu  Thu Feb  4 11:28:43 2010
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I think for most of us, voice human-to-human, real-time communication is =
just a smaller part of the communication universe. The average cell =
phone user talks for 459 minutes a month (claims =
http://wiki.answers.com/Q/How_many_minutes_does_the_average_cell_phone_cus=
tomer_use_per_month), i.e., 15 minutes a day. Most of us spend far more =
time on other Internet applications. Except for tele marketers and other =
voice-dependent professions, most of our needs so far tend to be fairly =
straightforward: reach a person (rather than a device) reliably and =
cheaply, preferably without disturbing the person accidentally.

Henning

On Feb 4, 2010, at 9:56 AM, <BeckW@telekom.de> <BeckW@telekom.de> wrote:

>=20
>> 2.1. SIP 2.0 is too expensive for Web application developers
> The big question is whether SIP is to blame for this complexity or if =
it's in the nature of telephony. Web standards are as complex as SIP. =
Only very few web browsers pass the acid3 test. XML is not the most =
elegant technology either.
>=20
> There must be a different reason why so many develop for the web and =
so few for voice. With Flash, it is already possible to do SIP-like =
voice/video connections between users. It's rarely used, though. When =
eBay bought Skype, everybody expected exciting mash-ups. Nothing =
happened.=20
>=20
>> 4.1.1. Applications reside in endpoints of the Internet
> Kind of. The success of Web 2.0 was the success of Javascript; =
applications run on the endpoint, but are maintained on the server. The =
user has no choice which javascript applet to use when accessing a web =
2.0 site. The site owner can update the applet without having to ask the =
browser owner for approval.
>=20
>=20
> Wolfgang Beck
>=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Zentrum Technik Einf=FChrung
> Wolfgang Beck
> Heinrich-Hertz-Stra=DFe 3-7, 64295 Darmstadt
> +49 6151 628 2832 (Tel.)
> http://www.telekom.com =20
>=20
> Erleben, was verbindet.=20
>=20
> Deutsche Telekom Netzproduktion GmbH
> Aufsichtsrat: Dr. Steffen Roehn (Vorsitzender)
> Gesch=E4ftsf=FChrung: Dr. Bruno Jacobfeuerborn (Vorsitzender), Albert =
Matheis, Klaus Peren
> Handelsregister: Amtsgericht Bonn HRB 14190
> Sitz der Gesellschaft: Bonn
> USt-IdNr. DE 814645262
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20


From eburger@standardstrack.com  Thu Feb  4 17:08:16 2010
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Subject: Re: [dispatch] FW: I-D ACTION:draft-sinnreich-sip-web-apis-00.txt
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What is totally missing is a comparison to Parlay X. We need something a =
bit deeper than REST vs. Web Serivces.

On Feb 3, 2010, at 3:21 PM, Peter Musgrave wrote:

> Hi Henry,=20
>=20
> (Based on a one-pass read through)
>=20
> I like the ideas expressed in this draft. The summary of SIP 2.0 is on =
target and the definition of a web API is a good idea. It is something I =
can use.
>=20
> Moving to HIP is a good idea.
>=20
> Regards,=20
>=20
> Peter Musgrave
>=20
> On Wed, Feb 3, 2010 at 1:17 PM, Henry Sinnreich =
<henry.sinnreich@gmail.com> wrote:
> We have published an I-D on "SIP APIs for Web Communications" and =
would
> appreciate any comments and input.
>=20
> Here is the link and the announcement:
>=20
> =
http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>=20
>=20
>  Title  : SIP APIs for Communications on the Web
>  Author(s) : H. Sinnreich, A. Johnston
>  Filename : draft-sinnreich-sip-web-apis-00.txt
>  Pages  : 19
>  Date  : 2010-1-19
>=20
>   Interactive communications are becoming part of rich Internet
>   applications (RIA). This can be accomplished entirely by using Web
>   technology and developing it as an Internet standard. Voice over IP
>   (VoIP) features developed for Session Initiation Protocol (SIP 2.0)
>   can be preserved in rich multimedia communications and applications
>   on the web.  Hyper Text Transport Protocol (HTTP) is used for =
control
>   and signaling data and RTP for interactive media transport
>   respectively. No other network application protocols are required.
>   Web, fixed and mobile device application development tools can thus
>   be used to include components and widgets for Internet presence,
>   Instant Messaging (IM), voice and video.  This opens Internet
>   communications, including voice to application developers and will
>   also enhance the user experience with real-time Web and mobile
>   communications.  Usage scenarios include service providers, private
>   IP networks, device application developers and media companies
>   providing a mix of integrated communications, applications and =
media.
>   This memo argues for the development and standardization of
>   Application Programming Interfaces (APIs) to allow the benefits of
>   SIP to be used by Rich Internet Applications (RIA), for fixed and
>   mobile devices.
>=20
> A URL for this Internet-Draft is:
> =
http://www.ietf.org/internet-drafts/draft-sinnreich-sip-web-apis-00.txt
>=20
> ------ End of Forwarded Message
>=20
> Thanks in advance for your comments.
>=20
> Henry
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch


From gonzalo.camarillo@ericsson.com  Fri Feb  5 03:12:31 2010
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Subject: [dispatch] Feedback on the process of chartering mini WGs
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Folks,

Now that we are gaining experience with the new RAI processes, it is
important to get feedback on how they are working.

We have received some emails indicating that the fact that mini WGs
sometimes take a while to get chartered can slow things down. Of course,
the faster the mini WGs get chartered the better. However, it is
important that folks continue with their technical work in parallel with
the chartering process. That is, the technical work does not need to
stop when DISPATCH decides to charter a given effort and resume when the
mini WG is chartered. It is expected that people continue working while
the ADs take care of the chartering process. We have created many
mailing lists before the chartering process is over and even
face-to-face meetings could be arranged as DISPATCH ad-hocs, if really
needed (we have already indicated that work can often progress without
face-to-face meetings). So, the tools for people to perform their
technical work should be available.

Also, we have received feedback regarding chairing mini WGs. It seems
some companies see chairing a WG as a long-term commitment and do not
make it easy for their people to take on chair roles. Note that the
charter of the mini WGs we will be chartering will say that they are
short-term focused efforts. Therefore, after two or three IETF cycles
the WG will be either done with its deliverables or will be shut down
because it did not manage to deliver. In both cases, the chair is not
required to make a long-term commitment.

Getting further feedback and ideas on these or related issues would of
course be great.

Thanks,

Gonzalo
DISPATCH co-chair

From john.elwell@siemens-enterprise.com  Fri Feb  5 03:40:52 2010
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To: DISPATCH <dispatch@ietf.org>
Date: Fri, 5 Feb 2010 12:41:38 +0100
Thread-Topic: Session Recording (RE: [dispatch] Feedback on the process of chartering mini WGs)
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Cc: Mary Barnes <mbarnes@nortelnetworks.com>
Subject: [dispatch] Session Recording (RE: Feedback on the process of chartering mini WGs)
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With particular reference to the proposed session recording WG, it seems I =
will be called upon to chair this, when it is finally established. I believ=
e those interested in this work are keen to see it progressing quickly, so =
we should aim to have topics to discuss at IETF#77, either at a meeting of =
the new WG, if it is formed by then, or at a DISPATCH ad hoc meeting. So du=
ring the next week or so, please give some thought to this and suggest poss=
ible agenda items.

John

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: 05 February 2010 11:13
> To: DISPATCH
> Cc: Mary Barnes
> Subject: [dispatch] Feedback on the process of chartering mini WGs
>=20
> Folks,
>=20
> Now that we are gaining experience with the new RAI processes, it is
> important to get feedback on how they are working.
>=20
> We have received some emails indicating that the fact that mini WGs
> sometimes take a while to get chartered can slow things down.=20
> Of course,
> the faster the mini WGs get chartered the better. However, it is
> important that folks continue with their technical work in=20
> parallel with
> the chartering process. That is, the technical work does not need to
> stop when DISPATCH decides to charter a given effort and=20
> resume when the
> mini WG is chartered. It is expected that people continue=20
> working while
> the ADs take care of the chartering process. We have created many
> mailing lists before the chartering process is over and even
> face-to-face meetings could be arranged as DISPATCH ad-hocs, if really
> needed (we have already indicated that work can often progress without
> face-to-face meetings). So, the tools for people to perform their
> technical work should be available.
>=20
> Also, we have received feedback regarding chairing mini WGs. It seems
> some companies see chairing a WG as a long-term commitment and do not
> make it easy for their people to take on chair roles. Note that the
> charter of the mini WGs we will be chartering will say that they are
> short-term focused efforts. Therefore, after two or three IETF cycles
> the WG will be either done with its deliverables or will be shut down
> because it did not manage to deliver. In both cases, the chair is not
> required to make a long-term commitment.
>=20
> Getting further feedback and ideas on these or related issues would of
> course be great.
>=20
> Thanks,
>=20
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From dromasca@avaya.com  Fri Feb  5 04:20:34 2010
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>, "DISPATCH" <dispatch@ietf.org>
Cc: Mary Barnes <mbarnes@nortelnetworks.com>
Subject: Re: [dispatch] Feedback on the process of chartering mini WGs
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=20

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

>=20
> Also, we have received feedback regarding chairing mini WGs.=20
> It seems some companies see chairing a WG as a long-term=20
> commitment and do not make it easy for their people to take=20
> on chair roles. Note that the charter of the mini WGs we will=20
> be chartering will say that they are short-term focused=20
> efforts. Therefore, after two or three IETF cycles the WG=20
> will be either done with its deliverables or will be shut=20
> down because it did not manage to deliver. In both cases, the=20
> chair is not required to make a long-term commitment.
>=20

Do we already have experience with any dispatch-originated mini-WG
having completed in 2-3 IETF cycles? We need to be careful here, I
think, as the task of the WG chairs does not end with the submission of
the documents to the IESG but continues during all the review period
(with strong involvment in the cases when the chair is also shepherding
documents) and we tend to declare WG's concluded and shut them down
after the last document in the charter was published. So, shorter life
cycles than the usual WG's - yes, two-three cycles as typical life time
- I do not know.=20

Dan

From ranjit@motorola.com  Fri Feb  5 04:24:11 2010
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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Hi All

Any comments on how we can progress this I-D?

Thanks=20


Regards
Ranjit

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Avasarala Ranjit-A20990
Sent: Friday, January 29, 2010 2:26 PM
To: Cullen Jennings; DISPATCH
Subject: Re: [dispatch]Commentsrequestedon
draft-avasarala-dispatch-comm-div-notification

Hi Cullen

This draft is kind of straight forward as it proposes a SIP based XML
event package for reporting call diversions that occur in the network on
behalf of the subscribers that set call forwarding rules based on
various criteria like "incoming caller identity", "time-of-day", etc.

The IPR is essentially based on the proposed event package and the call
diversion notification service that gets triggered whenever a call gets
diverted on behalf of a subscriber. This service is standardized in 3GPP
TS 24.604. Now since 3GPP does not standardize XML packages, there was a
need to submit an IETF draft and get it standardized.  With this intent,
we proposed an informational I-D on communication diversion notification
event package.

So I really do not think there is any other alternate way to accomplish
what we are trying to propose in this I-D.=20

Comments welcome

Thanks

Regards
Ranjit

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Cullen Jennings
Sent: Wednesday, January 27, 2010 10:37 PM
To: DISPATCH
Subject: Re: [dispatch] Commentsrequestedon
draft-avasarala-dispatch-comm-div-notification



There are way to accomplish the same end goals as this draft suggests
using things that are very likely to predate RIM's IPR on the topic. For
example the ideas on using things similar to the dialog events package
or reporting events about call forking on proxies. I have a strong
preference for doing this in an IPR free way. I strongly encourage the
WG to explore alternative ways to solve these problems before deciding
what to do with this draft.=20

Cullen <in my individual contributor role>



On Jan 15, 2010, at 2:49 AM, Avasarala Ranjit-A20990 wrote:

> Hi John
>=20
> Yes the notifications are generated for the AoR (contact) for which=20
> the diversion has occurred.
>=20
> So even though there could be multiple contacts registered for a=20
> particular AoR, if the diversion has occurred for a particular contact

> say alice at cellphone, then the notification information would go=20
> only to her cellphone and not to all registered contacts.  This way we

> would be limiting the number of notifications that are sent.
>=20
> In current call diversion service configurations, there is no concept=20
> of notifying subscribers when a particular call gets diverted based on

> a particular rule.
>=20
> Considering the case of Alice@example.com. Lets say Alice has setup a=20
> call diversion rule for her cellphone to divert all calls from a=20
> particular caller between 9 AM and 10 AM to her voicemail. But for=20
> some reason she could have wrongly configured the rule as 9 to 10 AM=20
> or could have put the caller id as wrong one. So it would be very=20
> difficult for her to manually spot this error by reviewing the=20
> complete set of call diversion services. So in order to facilitate=20
> Alice to effectively find out the error, a notification mechanism is
required.
>=20
> So here th URI for subscription would be the URI that is associated=20
> with diversion . E.g. Alice@cellhone or Alice@deskphone, etc.
>=20
> Thanks
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On=20
> Behalf Of Elwell, John
> Sent: Friday, January 15, 2010 2:57 PM
> To: Gonzalo Camarillo; DISPATCH
> Subject: Re: [dispatch] Comments requestedon=20
> draft-avasarala-dispatch-comm-div-notification
>=20
> I reserve judgement on how useful this would be. Concerning comments=20
> from Paul, I could imagine that the SUBSCRIBE request is addressed to=20
> the AOR URI for which notifications of diverted inbound requests are=20
> required, and that information from that AOR's domain proxy would be=20
> used as the basis for notifications. So the notifier would need=20
> somehow to be coupled to the domain proxy. This could certainly do=20
> with clarification.
>=20
> It is unclear to me how one would handle a subscription in the=20
> following circumstances. The subscription is to alice@example.com. At=20
> any given time, there might be several contacts registered against=20
> alice@example.com. According to relative priorities of those contacts,

> scripting rules at the proxy, caller preferences, etc., a given=20
> inbound request to Alice would go to one or more of those contacts. Is

> it the intention that those contacts that do not receive that inbound=20
> request would receive a notification within the context of this event
package?
> So if Alice's deskphone is one contact and her cellphone is another=20
> contact, if her current rules are for calls to go to her cellphone,=20
> her deskphone would receive notifications, and vice versa?
>=20
> Or is that out of scope? Is the intention to limit the scope to cases=20
> where an inbound communication is diverted away from the AOR
concerned?
>=20
> John
>=20
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>> Sent: 14 January 2010 14:03
>> To: DISPATCH
>> Subject: [dispatch] Comments requested on=20
>> draft-avasarala-dispatch-comm-div-notification
>>=20
>> Hi,
>>=20
>> we would like to ask for comments on the following draft. Do you find

>> it useful? Do you think it should be published as an RFC?
>>=20
>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>> otification-02
>>=20
>> Based on the comments received, we will talk to our ADs about this=20
>> piece of work.
>>=20
>> Thanks,
>>=20
>> Gonzalo
>> DISPATCH co-chair
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From gonzalo.camarillo@ericsson.com  Fri Feb  5 05:29:58 2010
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: Mary Barnes <mbarnes@nortelnetworks.com>, DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Feedback on the process of chartering mini WGs
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Hi Dan,

good point. This is something worthwhile considering. In any case, after
the WG submits all its documents to the IESG, the chair/shepherd's
commitment level is comparable to that of a document editor. That is, it
has to monitor (and help with) the progression of the draft but does not
need to perform many other administrative-oriented activities chairs
normally do. Therefore, their work load should be lower at that point in
time.

Thanks for the feedback!

Gonzalo

Romascanu, Dan (Dan) wrote:
>  
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> 
>> Also, we have received feedback regarding chairing mini WGs. 
>> It seems some companies see chairing a WG as a long-term 
>> commitment and do not make it easy for their people to take 
>> on chair roles. Note that the charter of the mini WGs we will 
>> be chartering will say that they are short-term focused 
>> efforts. Therefore, after two or three IETF cycles the WG 
>> will be either done with its deliverables or will be shut 
>> down because it did not manage to deliver. In both cases, the 
>> chair is not required to make a long-term commitment.
>>
> 
> Do we already have experience with any dispatch-originated mini-WG
> having completed in 2-3 IETF cycles? We need to be careful here, I
> think, as the task of the WG chairs does not end with the submission of
> the documents to the IESG but continues during all the review period
> (with strong involvment in the cases when the chair is also shepherding
> documents) and we tend to declare WG's concluded and shut them down
> after the last document in the charter was published. So, shorter life
> cycles than the usual WG's - yes, two-three cycles as typical life time
> - I do not know. 
> 
> Dan


From scottlawrenc@avaya.com  Fri Feb  5 05:34:55 2010
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From: "Scott Lawrence" <scottlawrenc@avaya.com>
To: Spencer Dawkins <spencer@wonderhamster.org>
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Cc: dispatch@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hadriel Kaplan wrote:

> >>  I've re-submitted a draft I originally submitted to the SIP WG back
> >>  when it was in existence.  The draft defines a session correlation
> >>  identifier for troubleshooting purposes.

> >>  > There are several reasons for having a globally unique session
> >>  > identifier for the same SIP session, which can be maintained across
> >>  > B2BUAs and other SIP middle-boxes.  This draft proposes a new SIP
> >>  > header to carry such a value: Session-ID.

Spencer Dawkins adds:

> I'm remembering that Hadriel had two drafts at about the same time - one was 
> a BCP on Call-ID, and one was on Session ID, and I'm remembering a lot of 
> confusion during discussions.
> 
> So I'm hoping that at least some of the objection was caused by confusion!
> 
> https://datatracker.ietf.org/drafts/draft-kaplan-sip-secure-call-id/ and
> https://datatracker.ietf.org/drafts/draft-kaplan-sip-session-id/, just for 
> the record...

I agree with Hadriel that the 'secure call id' draft actually solves
some different problems (and think that draft should certainly be
progressed - we've already started implementing the ideas in it in some
of our user agents). 

The problem of correlation across middleboxes of various kinds is
certainly an important one, and deserving of attention.  

I'd like to point out that there is another draft that helps with some
of the other more complex scenarios in which one is attempting to
determine the relationship between different sessions: 

        http://tools.ietf.org/html/draft-worley-references
        The References Header for SIP

we've also been using this in our products and found it very useful.

With respect to what to do about the session-id proposal: I think that
in some perfect world that we don't live in, it would be better to adopt
the secure-call-id proposal and also make the rule that such ids MUST
NOT be changed when passing through a B2BUA; the call-id would then have
the characteristics that the session-id needs.  I accept, however, that
this big a change to the handling of call-id by existing products is
unlikely.

It seems to me that session-id as proposed is valuable iff we can reach
a consensus that existing non-proxy middleboxes (whether they are called
SBCs or Application Servers or whatever) will pass these through
unmodified, or at least that in some reasonable time they can be
convinced to start doing so.  Based on the discussion to date, I'm not
yet convinced of that...

I wonder whether or not it's time that we bit the bullet and try to
actually comprehensively write down what the correct and harmful
behaviors for non-proxy middleboxes are?




From peter.musgrave@magorcorp.com  Fri Feb  5 05:47:08 2010
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--0016e6db2ae7cd735b047edab249
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> I wonder whether or not it's time that we bit the bullet and try to
> actually comprehensively write down what the correct and harmful
> behaviors for non-proxy middleboxes are?
>

I think there is merit is providing guidance. SIP middle boxes are here to
stay.

There is some stuff along these line in the expired
http://tools.ietf.org/html/draft-marjou-sipping-b2bua-01

(although I Just noticed Dale's new draft
http://tools.ietf.org/html/draft-worley-sipcore-b2bua-passthru-00 and I'm
not sure if there is any overlap).

Peter

--0016e6db2ae7cd735b047edab249
Content-Type: text/html; charset=ISO-8859-1
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<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
I wonder whether or not it&#39;s time that we bit the bullet and try to<br>
actually comprehensively write down what the correct and harmful<br>
behaviors for non-proxy middleboxes are?<br></blockquote><div><br></div><di=
v>I think there is merit is providing guidance. SIP middle boxes are here t=
o stay.</div><div><br></div><div>There is some stuff along these line in th=
e expired <a href=3D"http://tools.ietf.org/html/draft-marjou-sipping-b2bua-=
01">http://tools.ietf.org/html/draft-marjou-sipping-b2bua-01</a></div>
<div><br></div><div>(although I Just noticed Dale&#39;s new draft =A0<a hre=
f=3D"http://tools.ietf.org/html/draft-worley-sipcore-b2bua-passthru-00">htt=
p://tools.ietf.org/html/draft-worley-sipcore-b2bua-passthru-00</a> and I&#3=
9;m not sure if there is any overlap).</div>
<div><br></div><div>Peter</div></div><br>

--0016e6db2ae7cd735b047edab249--

From john.elwell@siemens-enterprise.com  Fri Feb  5 05:55:44 2010
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To: Scott Lawrence <scottlawrenc@avaya.com>, Spencer Dawkins <spencer@wonderhamster.org>
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=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
> Sent: 05 February 2010 13:35
> To: Spencer Dawkins
> Cc: dispatch@ietf.org; 'Hadriel Kaplan'
> Subject: Re: [dispatch] FW:=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> Hadriel Kaplan wrote:
>=20
> > >>  I've re-submitted a draft I originally submitted to the=20
> SIP WG back
> > >>  when it was in existence.  The draft defines a session=20
> correlation
> > >>  identifier for troubleshooting purposes.
>=20
> > >>  > There are several reasons for having a globally unique session
> > >>  > identifier for the same SIP session, which can be=20
> maintained across
> > >>  > B2BUAs and other SIP middle-boxes.  This draft=20
> proposes a new SIP
> > >>  > header to carry such a value: Session-ID.
>=20
> Spencer Dawkins adds:
>=20
> > I'm remembering that Hadriel had two drafts at about the=20
> same time - one was=20
> > a BCP on Call-ID, and one was on Session ID, and I'm=20
> remembering a lot of=20
> > confusion during discussions.
> >=20
> > So I'm hoping that at least some of the objection was=20
> caused by confusion!
> >=20
> >=20
> https://datatracker.ietf.org/drafts/draft-kaplan-sip-secure-ca
> ll-id/ and
> >=20
> https://datatracker.ietf.org/drafts/draft-kaplan-sip-session-i
> d/, just for=20
> > the record...
>=20
> I agree with Hadriel that the 'secure call id' draft actually solves
> some different problems (and think that draft should certainly be
> progressed - we've already started implementing the ideas in=20
> it in some
> of our user agents).=20
>=20
> The problem of correlation across middleboxes of various kinds is
> certainly an important one, and deserving of attention. =20
>=20
> I'd like to point out that there is another draft that helps with some
> of the other more complex scenarios in which one is attempting to
> determine the relationship between different sessions:=20
>=20
>         http://tools.ietf.org/html/draft-worley-references
>         The References Header for SIP
>=20
> we've also been using this in our products and found it very useful.
>=20
> With respect to what to do about the session-id proposal: I think that
> in some perfect world that we don't live in, it would be=20
> better to adopt
> the secure-call-id proposal and also make the rule that such ids MUST
> NOT be changed when passing through a B2BUA; the call-id=20
> would then have
> the characteristics that the session-id needs.  I accept,=20
> however, that
> this big a change to the handling of call-id by existing products is
> unlikely.
>=20
> It seems to me that session-id as proposed is valuable iff we=20
> can reach
> a consensus that existing non-proxy middleboxes (whether they=20
> are called
> SBCs or Application Servers or whatever) will pass these through
> unmodified, or at least that in some reasonable time they can be
> convinced to start doing so.  Based on the discussion to date, I'm not
> yet convinced of that...
>=20
> I wonder whether or not it's time that we bit the bullet and try to
> actually comprehensively write down what the correct and harmful
> behaviors for non-proxy middleboxes are?
[JRE] If somebody is able to put together a draft, that would be good. Howe=
ver, bear in mind that some non-proxy middle-boxes carry out 3PCC, and this=
 can lead to situations where the two ends of a call can comprise component=
s of two former calls, and therefore will not have the same call-ID even if=
 the two former calls had the same call-ID end-to-end. Yes, I know SIP has =
capabilities that theoretically make 3PCC unnecessary, but practical situat=
ions are likely to mean it stays around for the foreseeable future, e.g., p=
olicies that don't admit a REFER from another domain.

John

>=20
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From scottlawrenc@avaya.com  Fri Feb  5 06:02:59 2010
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On Fri, 2010-02-05 at 14:56 +0100, Elwell, John wrote:
> [JRE] If somebody is able to put together a draft, that would be good.
> However, bear in mind that some non-proxy middle-boxes carry out 3PCC,
> and this can lead to situations where the two ends of a call can
> comprise components of two former calls, and therefore will not have
> the same call-ID even if the two former calls had the same call-ID
> end-to-end. Yes, I know SIP has capabilities that theoretically make
> 3PCC unnecessary, but practical situations are likely to mean it stays
> around for the foreseeable future, e.g., policies that don't admit a
> REFER from another domain.

Adding a References header as proposed in Dales draft helps with
correlating those situations.



From Peter.Dawes@vodafone.com  Fri Feb  5 06:59:02 2010
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Hello,
Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03 +0200) as
"1) We have Hadriel's draft, 2) We also have the following draft, which
eventually led to the disaggregated media work:
http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog-correlati
on-01.txt 3) We also have the SIP-XMPP work, which may also need to
correlate sessions somehow."

I would like to add
http://tools.ietf.org/html/draft-dawes-sipping-debug-02 to this
discussion of use cases for some form of correlation identifier. This
draft relates to debugging/troubleshooting to meet documented
requirements from 3GPP and OMA, and proposes a kind of correlation
identifier in a new header.

Regards,
Peter


Message: 2
Date: Fri, 22 Jan 2010 11:05:57 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [dispatch] FW:
	I-DAction:draft-kaplan-dispatch-session-id-00.txt
To: L.Liess@telekom.de
Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
	HKaplan@acmepacket.com,	Martin.Huelsemann@telekom.de,
	Gerold.Pinker@telekom.de
Message-ID: <4B59CCE5.2010202@cisco.com>
Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed



L.Liess@telekom.de wrote:
> Paul,
>=20
> Thank you for the examples. The scenarios can be really complex. I
hope we can get these correlation identifiers, whatever their names, to
work correctly in all cases....=20
>=20
> I am not sure I understood your message correctly. Do you say we
should have two different identifiers because we use them for two
different things?=20

I think there may be more than two distinct usages that will require=20
different ones.

IMO use cases should be gathered and then bucketed into sets that can be

satisfied by a single mechanism, without assuming initially how many=20
buckets there will be.

(But I'm still on record as not being convinced that a correlation id is

needed *at all* for the case when multiple sip UAs are participating on=20
behalf of a single user in a call.)

> Thanks a lot
> Laura
>=20

From spencer@wonderhamster.org  Fri Feb  5 07:10:51 2010
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: "Dawes, Peter, VF-Group" <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>
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This is all excellent, of course.

My question is, what will it take for us to move forward on some flavor of a 
Session-ID (or other header field) that SIPCLF (for example) can use to 
correlate log entries collected from various SIP entities?

Bar BOF in Anaheim? Or can we propose something more concrete in the next 
small number of weeks?

Thanks,

Spencer

----- Original Message ----- 
From: "Dawes, Peter, VF-Group" <Peter.Dawes@vodafone.com>
To: <dispatch@ietf.org>
Cc: <Martin.Huelsemann@telekom.de>; <Gerold.Pinker@telekom.de>; 
<gonzalo.camarillo@ericsson.com>; <L.Liess@telekom.de>; 
<HKaplan@acmepacket.com>
Sent: Friday, February 05, 2010 8:59 AM
Subject: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt


> Hello,
> Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03 +0200) as
> "1) We have Hadriel's draft, 2) We also have the following draft, which
> eventually led to the disaggregated media work:
> http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog-correlati
> on-01.txt 3) We also have the SIP-XMPP work, which may also need to
> correlate sessions somehow."
>
> I would like to add
> http://tools.ietf.org/html/draft-dawes-sipping-debug-02 to this
> discussion of use cases for some form of correlation identifier. This
> draft relates to debugging/troubleshooting to meet documented
> requirements from 3GPP and OMA, and proposes a kind of correlation
> identifier in a new header.
>
> Regards,
> Peter
>
>
> Message: 2
> Date: Fri, 22 Jan 2010 11:05:57 -0500
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Subject: Re: [dispatch] FW:
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
> To: L.Liess@telekom.de
> Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,
> Gerold.Pinker@telekom.de
> Message-ID: <4B59CCE5.2010202@cisco.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
>
> L.Liess@telekom.de wrote:
>> Paul,
>>
>> Thank you for the examples. The scenarios can be really complex. I
> hope we can get these correlation identifiers, whatever their names, to
> work correctly in all cases....
>>
>> I am not sure I understood your message correctly. Do you say we
> should have two different identifiers because we use them for two
> different things?
>
> I think there may be more than two distinct usages that will require
> different ones.
>
> IMO use cases should be gathered and then bucketed into sets that can be
>
> satisfied by a single mechanism, without assuming initially how many
> buckets there will be.
>
> (But I'm still on record as not being convinced that a correlation id is
>
> needed *at all* for the case when multiple sip UAs are participating on
> behalf of a single user in a call.)
>
>> Thanks a lot
>> Laura
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch 


From adam@nostrum.com  Fri Feb  5 07:47:11 2010
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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Commentsrequestedon	draft-avasarala-dispatch-comm-div-notification
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Sorry; I just now tripped across this.

Probably the most important work the IETF does to foster innovation lies 
not in what we choose to standardize, but in what we choose to leave 
unstandardized. With rare exception, the IETF does not standardize how 
to implement services in SIP. One of the key philosophies that has been 
central to the development of the SIP protocol is that the IETF defines 
a rich and powerful set of protocol tools, and allows application 
developers to implement services using these tools.

This draft, however, is standardizing a service. No part of this is 
re-usable in any way.

So, what we really need to do here is back up to requirements: what are 
you trying to achieve? I looked through 24.604 briefly, and think I 
understand what you're trying to do here. And I'm sure you can define 
some generally useful, re-usable tools to achieve your goals.

I would offer that a general-purpose means for subscribing to the target 
generation steps at a SIP proxy would be both generally useful and 
sufficient to implement the service you have in mind.

/a

On 2/5/10 6:24 AM, Avasarala Ranjit-A20990 wrote:
> Hi All
>
> Any comments on how we can progress this I-D?
>
> Thanks
>
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Avasarala Ranjit-A20990
> Sent: Friday, January 29, 2010 2:26 PM
> To: Cullen Jennings; DISPATCH
> Subject: Re: [dispatch]Commentsrequestedon
> draft-avasarala-dispatch-comm-div-notification
>
> Hi Cullen
>
> This draft is kind of straight forward as it proposes a SIP based XML
> event package for reporting call diversions that occur in the network on
> behalf of the subscribers that set call forwarding rules based on
> various criteria like "incoming caller identity", "time-of-day", etc.
>
> The IPR is essentially based on the proposed event package and the call
> diversion notification service that gets triggered whenever a call gets
> diverted on behalf of a subscriber. This service is standardized in 3GPP
> TS 24.604. Now since 3GPP does not standardize XML packages, there was a
> need to submit an IETF draft and get it standardized.  With this intent,
> we proposed an informational I-D on communication diversion notification
> event package.
>
> So I really do not think there is any other alternate way to accomplish
> what we are trying to propose in this I-D.
>
> Comments welcome
>
> Thanks
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Wednesday, January 27, 2010 10:37 PM
> To: DISPATCH
> Subject: Re: [dispatch] Commentsrequestedon
> draft-avasarala-dispatch-comm-div-notification
>
>
>
> There are way to accomplish the same end goals as this draft suggests
> using things that are very likely to predate RIM's IPR on the topic. For
> example the ideas on using things similar to the dialog events package
> or reporting events about call forking on proxies. I have a strong
> preference for doing this in an IPR free way. I strongly encourage the
> WG to explore alternative ways to solve these problems before deciding
> what to do with this draft.
>
> Cullen<in my individual contributor role>
>
>
>
> On Jan 15, 2010, at 2:49 AM, Avasarala Ranjit-A20990 wrote:
>
>    
>> Hi John
>>
>> Yes the notifications are generated for the AoR (contact) for which
>> the diversion has occurred.
>>
>> So even though there could be multiple contacts registered for a
>> particular AoR, if the diversion has occurred for a particular contact
>>      
>    
>> say alice at cellphone, then the notification information would go
>> only to her cellphone and not to all registered contacts.  This way we
>>      
>    
>> would be limiting the number of notifications that are sent.
>>
>> In current call diversion service configurations, there is no concept
>> of notifying subscribers when a particular call gets diverted based on
>>      
>    
>> a particular rule.
>>
>> Considering the case of Alice@example.com. Lets say Alice has setup a
>> call diversion rule for her cellphone to divert all calls from a
>> particular caller between 9 AM and 10 AM to her voicemail. But for
>> some reason she could have wrongly configured the rule as 9 to 10 AM
>> or could have put the caller id as wrong one. So it would be very
>> difficult for her to manually spot this error by reviewing the
>> complete set of call diversion services. So in order to facilitate
>> Alice to effectively find out the error, a notification mechanism is
>>      
> required.
>    
>> So here th URI for subscription would be the URI that is associated
>> with diversion . E.g. Alice@cellhone or Alice@deskphone, etc.
>>
>> Thanks
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
>> Behalf Of Elwell, John
>> Sent: Friday, January 15, 2010 2:57 PM
>> To: Gonzalo Camarillo; DISPATCH
>> Subject: Re: [dispatch] Comments requestedon
>> draft-avasarala-dispatch-comm-div-notification
>>
>> I reserve judgement on how useful this would be. Concerning comments
>> from Paul, I could imagine that the SUBSCRIBE request is addressed to
>> the AOR URI for which notifications of diverted inbound requests are
>> required, and that information from that AOR's domain proxy would be
>> used as the basis for notifications. So the notifier would need
>> somehow to be coupled to the domain proxy. This could certainly do
>> with clarification.
>>
>> It is unclear to me how one would handle a subscription in the
>> following circumstances. The subscription is to alice@example.com. At
>> any given time, there might be several contacts registered against
>> alice@example.com. According to relative priorities of those contacts,
>>      
>    
>> scripting rules at the proxy, caller preferences, etc., a given
>> inbound request to Alice would go to one or more of those contacts. Is
>>      
>    
>> it the intention that those contacts that do not receive that inbound
>> request would receive a notification within the context of this event
>>      
> package?
>    
>> So if Alice's deskphone is one contact and her cellphone is another
>> contact, if her current rules are for calls to go to her cellphone,
>> her deskphone would receive notifications, and vice versa?
>>
>> Or is that out of scope? Is the intention to limit the scope to cases
>> where an inbound communication is diverted away from the AOR
>>      
> concerned?
>    
>> John
>>
>>
>>      
>>> -----Original Message-----
>>> From: dispatch-bounces@ietf.org
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>>> Sent: 14 January 2010 14:03
>>> To: DISPATCH
>>> Subject: [dispatch] Comments requested on
>>> draft-avasarala-dispatch-comm-div-notification
>>>
>>> Hi,
>>>
>>> we would like to ask for comments on the following draft. Do you find
>>>        
>    
>>> it useful? Do you think it should be published as an RFC?
>>>
>>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>>> otification-02
>>>
>>> Based on the comments received, we will talk to our ADs about this
>>> piece of work.
>>>
>>> Thanks,
>>>
>>> Gonzalo
>>> DISPATCH co-chair
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>>        
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>      
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


From ian.elz@ericsson.com  Fri Feb  5 07:51:24 2010
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From: Ian Elz <ian.elz@ericsson.com>
To: Peter Musgrave <peter.musgrave@magorcorp.com>, Scott Lawrence <scottlawrenc@avaya.com>
Date: Fri, 5 Feb 2010 16:52:09 +0100
Thread-Topic: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Cc: "dispatch@ietf.org" <dispatch@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
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The contents of the two drafts mentioned below is basically similar. The or=
iginal draft contained more detail (and was perhaps harder to read as a res=
ult) but the proposals are the same.

The solution proposed is only applicable managing mapped dialog identities =
in the headers which reference other dialogs. The problem should be expande=
d to include bodies which reference other dialogs. This was mentioned in dr=
aft-marjou but a solution was not explored.

The problem with using a solution as proposed in these drafts is that to pe=
rform the dialog mapping in the bodies requires every node which modifies a=
 dialog identity parameter (call-id,to and from tags) to be able to underst=
and every body which may contain a reference to other dialogs. I don't beli=
eve that this is practical. To require that every B2BUA should be able to d=
ecode, modify, and re-encode every body which may contain a dialog referenc=
e is impractical.

Hadriel's secure call id draft goes some way to solving the problem of node=
s mapping the call-id but it will not solve the issue. B2BUAs also modify t=
he to and from tags in the dialogs on either side of the B2BUA. Just using =
an unchanged secure call id will not solve the issue when the tags are also=
 mapped.

The original Session-id proposal to use an end-to-end identifier which woul=
d be passed transparently has more promise of a final solution. It was a sh=
ame that the usage of this proposed header was deprecated to just tracing b=
efore the full opportunities could be explored. I understand that there are=
 issues associated with forking and 3PCC but these should be resolvable.

As for guidance then this is probably a good thing but it will only ever be=
 guidance. There are too many nodes deployed which will not follow any guid=
ance to ensure that they all follow ant rules which are written now. It is =
also easier to insert a B2BUA to overcome a specific problem as you can mov=
e outside the defined protocol rules inside the node.

Ian Elz

________________________________
From: Peter Musgrave [mailto:peter.musgrave@magorcorp.com]
Sent: 05 February 2010 13:48
To: Scott Lawrence
Cc: dispatch@ietf.org; Hadriel Kaplan
Subject: Re: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.t=
xt


I wonder whether or not it's time that we bit the bullet and try to
actually comprehensively write down what the correct and harmful
behaviors for non-proxy middleboxes are?

I think there is merit is providing guidance. SIP middle boxes are here to =
stay.

There is some stuff along these line in the expired http://tools.ietf.org/h=
tml/draft-marjou-sipping-b2bua-01

(although I Just noticed Dale's new draft  http://tools.ietf.org/html/draft=
-worley-sipcore-b2bua-passthru-00 and I'm not sure if there is any overlap)=
.

Peter


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dus-ascii" http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18865"></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The contents of the two drafts mentioned below is bas=
ically=20
similar. The original draft contained more detail (and was perhaps harder t=
o=20
read as a result) but the proposals&nbsp;are the same.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The solution proposed is only applicable managing map=
ped=20
dialog identities in the headers which reference other dialogs. The problem=
=20
should be expanded to include bodies which reference other dialogs. This wa=
s=20
mentioned in draft-marjou but a solution was not explored.</FONT></SPAN></D=
IV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The problem with using a solution as proposed in thes=
e drafts=20
is that to perform the dialog mapping in the bodies requires every node whi=
ch=20
modifies a dialog identity parameter (call-id,to and from tags) to be able =
to=20
understand every body which may contain a reference to other dialogs. I don=
't=20
believe that this is practical. To require that every B2BUA should be able =
to=20
decode, modify, and re-encode every body which may contain a dialog referen=
ce is=20
impractical.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Hadriel's secure call id draft goes some way to solvi=
ng the=20
problem of nodes mapping the call-id but it will not solve the issue. B2BUA=
s=20
also modify the to and from tags in the dialogs on either side of the B2BUA=
.=20
Just using an unchanged secure call id will not solve the issue when the ta=
gs=20
are also mapped.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>The original Session-id proposal to use an end-to-end=
=20
identifier which would be passed transparently has more promise of a final=
=20
solution. It was a shame that the usage of this proposed header was depreca=
ted=20
to just tracing before the full opportunities could be explored. I understa=
nd=20
that there are issues associated with forking and 3PCC but these should be=
=20
resolvable.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>As for guidance then this is probably a good thing bu=
t it will=20
only ever be guidance. There are too many nodes deployed which will not fol=
low=20
any guidance to ensure that they all follow ant rules which are written now=
. It=20
is also easier to insert a B2BUA to overcome a specific problem as you can =
move=20
outside the defined protocol rules inside the node.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D270563515-05022010><FONT color=3D=
#0000ff=20
size=3D2 face=3DArial>Ian Elz</FONT></SPAN></DIV><BR>
<BLOCKQUOTE style=3D"MARGIN-RIGHT: 0px">
  <DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT size=3D2 face=3DTahoma><B>From:</B> Peter Musgrave=20
  [mailto:peter.musgrave@magorcorp.com] <BR><B>Sent:</B> 05 February 2010=20
  13:48<BR><B>To:</B> Scott Lawrence<BR><B>Cc:</B> dispatch@ietf.org; Hadri=
el=20
  Kaplan<BR><B>Subject:</B> Re: [dispatch] FW:=20
  I-DAction:draft-kaplan-dispatch-session-id-00.txt<BR></FONT><BR></DIV>
  <DIV></DIV><BR>
  <DIV class=3Dgmail_quote>
  <BLOCKQUOTE=20
  style=3D"BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-=
LEFT: 1ex"=20
  class=3Dgmail_quote>I wonder whether or not it's time that we bit the bul=
let=20
    and try to<BR>actually comprehensively write down what the correct and=
=20
    harmful<BR>behaviors for non-proxy middleboxes are?<BR></BLOCKQUOTE>
  <DIV><BR></DIV>
  <DIV>I think there is merit is providing guidance. SIP middle boxes are h=
ere=20
  to stay.</DIV>
  <DIV><BR></DIV>
  <DIV>There is some stuff along these line in the expired <A=20
  href=3D"http://tools.ietf.org/html/draft-marjou-sipping-b2bua-01">http://=
tools.ietf.org/html/draft-marjou-sipping-b2bua-01</A></DIV>
  <DIV><BR></DIV>
  <DIV>(although I Just noticed Dale's new draft &nbsp;<A=20
  href=3D"http://tools.ietf.org/html/draft-worley-sipcore-b2bua-passthru-00=
">http://tools.ietf.org/html/draft-worley-sipcore-b2bua-passthru-00</A>=20
  and I'm not sure if there is any overlap).</DIV>
  <DIV><BR></DIV>
  <DIV>Peter</DIV></DIV><BR></BLOCKQUOTE></BODY></HTML>

--_000_D63DB3A4C85587439C3BA6A32D1D0943036A054080ESESSCMS0352e_--

From L.Liess@telekom.de  Fri Feb  5 08:21:53 2010
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From: <L.Liess@telekom.de>
To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>, <spencer@wonderhamster.org>
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Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com, HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de
Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Peter,

My understanding is that session-id and debug-id cover different sets of =
requirements.=20

Session-id is in every message. If people want to find out what happened =
two days before, they can look at the old logs and the session-id is =
there in every message to help them to correlate e2e. For the debug-id =
to be there you have to know in advance that and which e2e traffic you =
would like to have correlated. The debug-id is activated "on-demand" and =
has no performance impact when it is not active, session-id is active =
all the time. The advantage of the debug-id is that there is no =
performance impact when debugging is not needed. Is my understanding =
correct?=20
=20
I like the debug-id stuff, it's  really smart and we looked at it. But =
our troubleshooting and security guys want the session-id, for the =
reason above. I assume every service provider has its own needs and =
requirements.          =20

Spencer is right.  BOF, CLF whatever...we have to find some way to come =
forward with the e2e correlation.=20

Thanks a lot
Laura



> -----Original Message-----
> From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com]=20
> Sent: Friday, February 05, 2010 4:00 PM
> To: dispatch@ietf.org
> Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;=20
> HKaplan@acmepacket.com; H=FClsemann, Martin; Pinker, Gerold
> Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> Hello,
> Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03=20
> +0200) as
> "1) We have Hadriel's draft, 2) We also have the following=20
> draft, which
> eventually led to the disaggregated media work:
> http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
> -correlati
> on-01.txt 3) We also have the SIP-XMPP work, which may also need to
> correlate sessions somehow."
>=20
> I would like to add
> http://tools.ietf.org/html/draft-dawes-sipping-debug-02 to this
> discussion of use cases for some form of correlation identifier. This
> draft relates to debugging/troubleshooting to meet documented
> requirements from 3GPP and OMA, and proposes a kind of correlation
> identifier in a new header.
>=20
> Regards,
> Peter
>=20
>=20
> Message: 2
> Date: Fri, 22 Jan 2010 11:05:57 -0500
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Subject: Re: [dispatch] FW:
> 	I-DAction:draft-kaplan-dispatch-session-id-00.txt
> To: L.Liess@telekom.de
> Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
> 	HKaplan@acmepacket.com,	Martin.Huelsemann@telekom.de,
> 	Gerold.Pinker@telekom.de
> Message-ID: <4B59CCE5.2010202@cisco.com>
> Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed
>=20
>=20
>=20
> L.Liess@telekom.de wrote:
> > Paul,
> >=20
> > Thank you for the examples. The scenarios can be really complex. I
> hope we can get these correlation identifiers, whatever their=20
> names, to
> work correctly in all cases....=20
> >=20
> > I am not sure I understood your message correctly. Do you say we
> should have two different identifiers because we use them for two
> different things?=20
>=20
> I think there may be more than two distinct usages that will require=20
> different ones.
>=20
> IMO use cases should be gathered and then bucketed into sets=20
> that can be
>=20
> satisfied by a single mechanism, without assuming initially how many=20
> buckets there will be.
>=20
> (But I'm still on record as not being convinced that a=20
> correlation id is
>=20
> needed *at all* for the case when multiple sip UAs are=20
> participating on=20
> behalf of a single user in a call.)
>=20
> > Thanks a lot
> > Laura
> >=20
>=20

From victor.pascual.avila@gmail.com  Fri Feb  5 09:57:56 2010
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Date: Fri, 5 Feb 2010 18:58:39 +0100
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: Scott Lawrence <scottlawrenc@avaya.com>
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Cc: dispatch@ietf.org, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [dispatch] FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Scott,

On Fri, Feb 5, 2010 at 2:35 PM, Scott Lawrence <scottlawrenc@avaya.com> wro=
te:
> I wonder whether or not it's time that we bit the bullet and try to
> actually comprehensively write down what the correct and harmful
> behaviors for non-proxy middleboxes are?

Please check draft-elwell-sip-e2e-identity-important, Section 8:
Analysis of changes to SIP requests by B2BUAs

http://tools.ietf.org/html/draft-elwell-sip-e2e-identity-important-03#secti=
on-8

I believe folks may have a divergence of opinion on what is considered
correct and what is considered harmful

Cheers,
--=20
Victor Pascual =C3=81vila

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From: "Dale Worley" <dworley@avaya.com>
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Date: Fri, 05 Feb 2010 14:56:45 -0500
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Subject: Re: [dispatch] FW:	I-DAction:draft-kaplan-dispatch-session-id-00.txt
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On Fri, 2010-02-05 at 14:56 +0100, Elwell, John wrote:
> If somebody is able to put together a draft, that would be good.
> However, bear in mind that some non-proxy middle-boxes carry out 3PCC,
> and this can lead to situations where the two ends of a call can
> comprise components of two former calls, and therefore will not have
> the same call-ID even if the two former calls had the same call-ID
> end-to-end.

This is an important point about any proposed correlation mechanism:  In
various situations, as Paul describes, an element discovers that two
dialogs are correlated *after* they have been in existence some time,
which is after any putative correlation identifier has been assigned to
each call.

The advantage of Session-Id is that it addresses what is really a very
narrow application scenario, a call proceeding through an SBC, but one
that is very common.  Because of its simplicity, it has a higher chance
of being adopted.  It is not a complete solution, but it provides a lot
of value anyway.

On the other hand, if one wants a correlation mechanism that is
effective for a wide range of scenarios, one usually runs up against the
correlation-post-facto problem.  That more general case can't be solved
by assigning each dialog a unique correlation identifier.  You have to
have some method of declaring the correlation of two pre-existing
dialogs that share no identifiers.  (That's the approach taken in
http://tools.ietf.org/html/draft-worley-references.)

Dale



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To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
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Subject: Re: [dispatch] Comments requestedondraft-avasarala-dispatch-comm-div-notification
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inline...

Avasarala Ranjit-A20990 wrote:

>> Assuming sip:alice@office.domain.com is not an AOR, but rather is a 
>> registered contact to sip:alice@domain.com, where in the routing of 
>> the call does the server that evaluates this rule get control?
>>
>> <Ranjit> The server looks for the incoming caller identity and the 
>> destination identity to see if there is a matching rule configured.
> 
> I guess my question wasn't clear.
> Where in the routing sequence does this server reside?
> 
> <Ranjit-Jan27> The server could be the Application Server where the
> diversion rules reside and get executed.
> 
> If it is positioned before the contact routing has occurred, then it
> can't know which contact was chosen.
> <Ranjit-Jan27th> No it would be before that. 

> Since this proposal is coming, more or less, from IMS I assume you are
> imagining this is an IMS Application Server. IIRC, those get control
> *before* contact routing is done by the S-CSCF. So how would the AS know
> if a particular contact was chosen. (I take some issue with calling
> contact routing "diversion", but lets not discuss that right now.)
> 
> <Ranjit-Jan27th> Yes you are right. This proposal is more from a IMS and
> 3GPP perspective and I am trying to generalize it for IETF.

OK. So once again I have to ask for a description of the system 
architecture(s) in which this event package makes sense - where the 
notifier would have the knowledge to make the proposed kind of 
notifications.

>> If the registered 
>>          contact is not an AoR, then the call cannot be routed to it 
>> and
> 
> Huh? Whether the registered contact is an AOR has no bearing on whether
> its possible to route to it. And, its in general impossible to determine
> by looking whether a URI is an AOR or not.
> 
> <Ranjit-Jan27th> Agree with you on this.
> 
>> hence will be routed to the main AoR i.e sip:alice@domain.com and the 
>> rule would not be executed.
> 
> That statement makes me even more confused. I have no idea what it
> means.
> 
> <Ranjit-Jan27th> Here I mean, the notifications would be sent to the
> main AoR with the details of "diverting-user", "diverted-to-user" being
> part
> Of the XML package. The notifications need not be sent to the registered
> contact. For e.g if sip:alice@domain.com is the AoR and diversions are
> taking
> Place for sip:alice@office.domain.com, then notifications will be sent
> to sip:alice@domain.com only and not to sip:alice@domain.com. This will
> simplify things I feel !!

I'm am getting more confused, not less.

I think the draft needs some significant rewriting in terms of concepts 
and terminology, as well as assumed architecture, in order to be 
comprehensible in any context other that pure IMS. And frankly I don't 
understand it even if I restrict the context to IMS.

>> But if sip:alice@office.domain.com is an AoR (e.g. GRUU), then the 
>> call could be routed and if there is adiversion rule there,
>>          the diversion rule would get executed and a notification 
>> generated towards that AoR.
> 
> This is getting more and more confusing. So I have even more desire for
> at least a detailed use case that shows all the players, their addresses
> and relationships to one another, and the call flow.
> 
> <Ranjit-Jan27th> as I mentioned above, to simplify things, lets assume
> that notifications always go to the main AoR i.e the Public User
> Identity (in IMS)

> And not to any of the registered AoR irrespective of who the diverting
> user is. So here even tough the divering user is sip:alice@office or
> sip:alice@home, the notifications always go to sip:alice@domain.com. 

How does that simplify anything? If anything that makes it worse. 
Notifications go to the subscriber, and the content of that notification 
should in general have nothing to do with who the subscriber is. (But 
may be influenced by authorization policy.) Notifications are always 
in-dialog, to the remote target of that dialog, and hence pretty much 
never go to an AOR.

>> How would it work in cases where the address for Alice's office phone 
>> is a dynamically assigned ip address rather than a dns name?
>>
>> <Ranjit> Since the notification gets generated dynamically when ever 
>> the diversion occurs, the new ip address is taken prior to sending the
> 
>> notification.
> 
> Again it seems my question wasn't clear.
> You are showing rules that contain the contact addresses of particular
> devices. If the contact URIs registered by those devices contain
> dynamically assigned addresses, how would a subscriber know how to
> construct the rule that selects a particular device? Or if it is
> receiving all the diversions, how would it relate the reported results
> to particular devices?
> 
> <Ranjit-Jan27th> if the notifications are sent to PUI, then the contact
> addresses are specified as URLs. So there is no need to send any
> notifications to individual contact adsresses

You are missing my point entirely. I'm not asking about where the 
notifications are *sent* (which is irrelevant). I'm asking about the 
*content* of those notifications, and of the filters that select which 
ones the subscriber is interested in.

Assume for the moment that we have AOR sip:alice@SSP.com.
And registered with that AOR are contact addresses 
sip:alice@mobile123.SSP.com and sip:alice@office.domain.com.

Some UA (I'll not say which) is concerned about diversions related to 
that AOR. It subscribes to the comm-div package that sip:alice@SSP.com. 
It may include some sort of filter in the subscription.

If a request comes in to the AOR, and neither of the contacts responds, 
a server serving that AOR (perhaps an AS) may eventually decide to 
forward the request to a VM server. I can see how this would be 
considered a real diversion, and how the server would know about it. So 
it can generate a NOTIFY reporting this diversion.

Next, its possible that Alice, from her mobile phone, establishes a 
forwarding rule, that sends all incoming calls to the AOR to 
sip:bob@SSP.com. The server would apply this rule, as policy, to all 
incoming requests to the AOR, bypassing contact routing. Again its 
unremarkable that it would be able to generate a notification when 
carrying out such a retargeting.

None of that requires any filter on the address at which diversion takes 
place, because it takes place exactly on diversion of requests to the 
same URI to which the subscription was made.

Your filter package implies that the notifier can generate notifications 
for cases where the diversion takes place from some URI *other* than the 
one for which the subscriptions is made. I am guessing you mean cases 
when the diversion takes place after the call has been retargeted to one 
of the contacts. I have two problems with that:

1) how would the server for sip:alice@SSP.com know that a diversion has 
occurred on a request targetted to sip:alice@office.domain.com?

2) even if it had a way to discover that, how would a subscriber know 
the URI sip:alice@office.domain.com in order to place it in a filter?

But I find I am repeating myself (see below from before.)
There must be some fundamental difference in perspective that is 
preventing us from understanding one another.

> This relates, in a way, to whether contact routing is diversion or not. 
> Typically what I think of as real diversion would be to a URI that *is*
> an AOR, and hence stable and knowable by those formulating rules. But
> typical contact routing would involve URIs that aren't very meaningful.
> 
> If you want to be able to write rules like: "let me know when the call
> is routed to the phone in my office rather than the one in my home" 
> (where both have the same "phone number"), then you are going to have to
> find a way to give those stable URIs, like sip:alice@office.domain.com
> and sip:alice@home.domain.com. AFAIK that isn't the norm, so if you are
> assuming it then please make that clear.

	Thanks,
	Paul

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References: <4B4F2403.7010200@ericsson.com><A444A0F8084434499206E78C106220CA6A158B09@MCHP058A.global-ad.net><750BBC72E178114F9DC4872EBFF29A5B0A245729@ZMY16EXM66.ds.mot.com><4873a6aacb123154f96af62bb241016d.squirrel@www.softarmor.com><750BBC72E178114F9DC4872EBFF29A5B0A28A86E@ZMY16EXM66.ds.mot.com><A444A0F8084434499206E78C106220CA6A1C3DC7@MCHP058A.global-ad.net><750BBC72E178114F9DC4872EBFF29A5B0A28AA39@ZMY16EXM66.ds.mot.com> <A444A0F8084434499206E78C106220CA6A1C488E@MCHP058A.global-ad.net> <4B55F6B7.60002@cisco.com> <C1C95024-5687-4C69-9468-895BD2C9AA71@softarmor.com> <750BBC72E178114F9DC4872EBFF29A5B0A28AB6C@ZMY16EXM66.ds.mot.com> <4B574722.9070108@cisco.com> <A444A0F8084434499206E78C106220CA6A233EFB@MCHP058A.global-ad.net> <750BBC72E178114F9DC4872EBFF29A5B0A28ADD5@ZMY16EXM66.ds.mot.com> <4B59BA06.40608@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A28AF42@ZMY16EXM66.ds.mot.com> <4B5DA8E3.7050306@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A28B1FD@ZMY16EXM66.ds.mot.com> <4B6F4 742.4010 500@cisco.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Comments requestedondraft-avasarala-dispatch-comm-div-notification
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Hi Paul

My comments <Ranjit-Feb08>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Monday, February 08, 2010 4:36 AM
To: Avasarala Ranjit-A20990
Cc: Elwell, John; DISPATCH
Subject: Re: [dispatch] Comments
requestedondraft-avasarala-dispatch-comm-div-notification

inline...

Avasarala Ranjit-A20990 wrote:

>> Assuming sip:alice@office.domain.com is not an AOR, but rather is a=20
>> registered contact to sip:alice@domain.com, where in the routing of=20
>> the call does the server that evaluates this rule get control?
>>
>> <Ranjit> The server looks for the incoming caller identity and the=20
>> destination identity to see if there is a matching rule configured.
>=20
> I guess my question wasn't clear.
> Where in the routing sequence does this server reside?
>=20
> <Ranjit-Jan27> The server could be the Application Server where the=20
> diversion rules reside and get executed.
>=20
> If it is positioned before the contact routing has occurred, then it=20
> can't know which contact was chosen.
> <Ranjit-Jan27th> No it would be before that.=20

> Since this proposal is coming, more or less, from IMS I assume you are

> imagining this is an IMS Application Server. IIRC, those get control
> *before* contact routing is done by the S-CSCF. So how would the AS=20
> know if a particular contact was chosen. (I take some issue with=20
> calling contact routing "diversion", but lets not discuss that right=20
> now.)
>=20
> <Ranjit-Jan27th> Yes you are right. This proposal is more from a IMS=20
> and 3GPP perspective and I am trying to generalize it for IETF.

OK. So once again I have to ask for a description of the system
architecture(s) in which this event package makes sense - where the
notifier would have the knowledge to make the proposed kind of
notifications.

<Ranjit-Feb08> The proposed I-D is actually for registering the proposed
event package for the communication diversion service described in 3GPP
TS 24.604. So here the notifier would the AS where the communication
diversion service is running. So yes it would have knowledge of on whose
behalf it is
Doing the diversion and to whom it should notify.

>> If the registered=20
>>          contact is not an AoR, then the call cannot be routed to it=20
>> and
>=20
> Huh? Whether the registered contact is an AOR has no bearing on=20
> whether its possible to route to it. And, its in general impossible to

> determine by looking whether a URI is an AOR or not.
>=20
> <Ranjit-Jan27th> Agree with you on this.
>=20
>> hence will be routed to the main AoR i.e sip:alice@domain.com and the

>> rule would not be executed.
>=20
> That statement makes me even more confused. I have no idea what it=20
> means.
>=20
> <Ranjit-Jan27th> Here I mean, the notifications would be sent to the=20
> main AoR with the details of "diverting-user", "diverted-to-user"=20
> being part Of the XML package. The notifications need not be sent to=20
> the registered contact. For e.g if sip:alice@domain.com is the AoR and

> diversions are taking Place for sip:alice@office.domain.com, then=20
> notifications will be sent to sip:alice@domain.com only and not to=20
> sip:alice@domain.com. This will simplify things I feel !!

I'm am getting more confused, not less.

I think the draft needs some significant rewriting in terms of concepts
and terminology, as well as assumed architecture, in order to be
comprehensible in any context other that pure IMS. And frankly I don't
understand it even if I restrict the context to IMS.

<Ranjit-Feb08> Actually the terminology and the service in general is
explained in 24.604 (Section 4.10). So we felt that this I-D would just=20
Register the proposed event package.

>> But if sip:alice@office.domain.com is an AoR (e.g. GRUU), then the=20
>> call could be routed and if there is adiversion rule there,
>>          the diversion rule would get executed and a notification=20
>> generated towards that AoR.
>=20
> This is getting more and more confusing. So I have even more desire=20
> for at least a detailed use case that shows all the players, their=20
> addresses and relationships to one another, and the call flow.
>=20
> <Ranjit-Jan27th> as I mentioned above, to simplify things, lets assume

> that notifications always go to the main AoR i.e the Public User Bu
> Identity (in IMS)

> And not to any of the registered AoR irrespective of who the diverting

> user is. So here even tough the divering user is sip:alice@office or=20
> sip:alice@home, the notifications always go to sip:alice@domain.com.

How does that simplify anything? If anything that makes it worse.=20
Notifications go to the subscriber, and the content of that notification
should in general have nothing to do with who the subscriber is. (But
may be influenced by authorization policy.) Notifications are always
in-dialog, to the remote target of that dialog, and hence pretty much
never go to an AOR.

<Ranjit-Feb08> True. Notification go to the subscribed entity as shown
in the example in 24.604.=20

>> How would it work in cases where the address for Alice's office phone

>> is a dynamically assigned ip address rather than a dns name?
>>
>> <Ranjit> Since the notification gets generated dynamically when ever=20
>> the diversion occurs, the new ip address is taken prior to sending=20
>> the
>=20
>> notification.
>=20
> Again it seems my question wasn't clear.
> You are showing rules that contain the contact addresses of particular

> devices. If the contact URIs registered by those devices contain=20
> dynamically assigned addresses, how would a subscriber know how to=20
> construct the rule that selects a particular device? Or if it is=20
> receiving all the diversions, how would it relate the reported results

> to particular devices?
>=20
> <Ranjit-Jan27th> if the notifications are sent to PUI, then the=20
> contact addresses are specified as URLs. So there is no need to send=20
> any notifications to individual contact adsresses

You are missing my point entirely. I'm not asking about where the
notifications are *sent* (which is irrelevant). I'm asking about the
*content* of those notifications, and of the filters that select which
ones the subscriber is interested in.

<Ranjit-Feb08> The content of notifications are the elements specified
in the event pacakge. The example shows both a sample notification and
the elements that can be used as filters to control the content of
notifications. E.g. Section 4.10.1.1.1.2 of 24.604

Assume for the moment that we have AOR sip:alice@SSP.com.
And registered with that AOR are contact addresses
sip:alice@mobile123.SSP.com and sip:alice@office.domain.com.

Some UA (I'll not say which) is concerned about diversions related to
that AOR. It subscribes to the comm-div package that sip:alice@SSP.com.=20
It may include some sort of filter in the subscription.

If a request comes in to the AOR, and neither of the contacts responds,
a server serving that AOR (perhaps an AS) may eventually decide to
forward the request to a VM server. I can see how this would be
considered a real diversion, and how the server would know about it. So
it can generate a NOTIFY reporting this diversion.

<Ranjit-Feb08> True. So here the notification would be towards
sip:alice@SSP.com with entity=3D"sip:alice@mobile123.SSP.com OR
entity=3Dsip:alice@office.SSP.com on whose behlaf the diversion took =
place
or where the call landed.=20

Next, its possible that Alice, from her mobile phone, establishes a
forwarding rule, that sends all incoming calls to the AOR to
sip:bob@SSP.com. The server would apply this rule, as policy, to all
incoming requests to the AOR, bypassing contact routing. Again its
unremarkable that it would be able to generate a notification when
carrying out such a retargeting.

<Ranjit-Feb08> so here if Alice gets notified whenever a incoming call
to her gets forwarded to Bob. If she sets a filter criteria like
timeofday, then only incoming calls during that time get notified and
not others. So the notifications depend on the target of the incoming
call and other specified criteria.=20

None of that requires any filter on the address at which diversion takes
place, because it takes place exactly on diversion of requests to the
same URI to which the subscription was made.

<Ranjit-Feb08> why not? Filters can be set based on time, range of time
or particular day or context (alice is OOO, etc)

Your filter package implies that the notifier can generate notifications
for cases where the diversion takes place from some URI *other* than the
one for which the subscriptions is made. I am guessing you mean cases
when the diversion takes place after the call has been retargeted to one
of the contacts. I have two problems with that:

<Ranjit-Feb08> No No. Notifier generates notifications only for
diversions that take place on behalf of the subscribed entity and not
others. So here sip:alice@SSP.com subscribes to diversions taking place
at any of its contacts and get notified based on filter criteria. The
filter critera can specify which contacts the notification is required
for and which of those notifications are not required.=20

1) how would the server for sip:alice@SSP.com know that a diversion has
occurred on a request targetted to sip:alice@office.domain.com?

2) even if it had a way to discover that, how would a subscriber know
the URI sip:alice@office.domain.com in order to place it in a filter?

But I find I am repeating myself (see below from before.) There must be
some fundamental difference in perspective that is preventing us from
understanding one another.

> This relates, in a way, to whether contact routing is diversion or
not.=20
> Typically what I think of as real diversion would be to a URI that=20
> *is* an AOR, and hence stable and knowable by those formulating rules.

> But typical contact routing would involve URIs that aren't very
meaningful.
>=20
> If you want to be able to write rules like: "let me know when the call

> is routed to the phone in my office rather than the one in my home"
> (where both have the same "phone number"), then you are going to have=20
> to find a way to give those stable URIs, like=20
> sip:alice@office.domain.com and sip:alice@home.domain.com. AFAIK that=20
> isn't the norm, so if you are assuming it then please make that clear.

	Thanks,
	Paul

Regards
Ranjit

From drage@alcatel-lucent.com  Mon Feb  8 05:42:58 2010
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 8 Feb 2010 14:43:09 +0100
Thread-Topic: [dispatch] Comments requestedondraft-avasarala-dispatch-comm-div-notification
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References: <4B4F2403.7010200@ericsson.com><A444A0F8084434499206E78C106220CA6A158B09@MCHP058A.global-ad.net><750BBC72E178114F9DC4872EBFF29A5B0A245729@ZMY16EXM66.ds.mot.com><4873a6aacb123154f96af62bb241016d.squirrel@www.softarmor.com><750BBC72E178114F9DC4872EBFF29A5B0A28A86E@ZMY16EXM66.ds.mot.com><A444A0F8084434499206E78C106220CA6A1C3DC7@MCHP058A.global-ad.net><750BBC72E178114F9DC4872EBFF29A5B0A28AA39@ZMY16EXM66.ds.mot.com> <A444A0F8084434499206E78C106220CA6A1C488E@MCHP058A.global-ad.net> <4B55F6B7.60002@cisco.com> <C1C95024-5687-4C69-9468-895BD2C9AA71@softarmor.com> <750BBC72E178114F9DC4872EBFF29A5B0A28AB6C@ZMY16EXM66.ds.mot.com> <4B574722.9070108@cisco.com> <A444A0F8084434499206E78C106220CA6A233EFB@MCHP058A.global-ad.net> <750BBC72E178114F9DC4872EBFF29A5B0A28ADD5@ZMY16EXM66.ds.mot.com> <4B59BA06.40608@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A28AF42@ZMY16EXM66.ds.mot.com> <4B5DA8E3.7050306@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A28B1FD@ZMY16EXM66.ds.mot.com>	<4B6F4 742.4010 500@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A2CD16D@ZMY16EXM66.ds.mot.com>
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Subject: Re: [dispatch] Comments	requestedondraft-avasarala-dispatch-comm-div-notification
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Can I understand where this discussion is going?

The relevant paragraph of http://tools.ietf.org/id/draft-peterson-rai-rfc34=
27bis-04.txt is:

   Individuals may wish to publish SIP Event packages that they believe
   fall outside the scope of any chartered work currently in RAI.
   Individual proposals for registration of a SIP event package MUST
   first be published as Internet-drafts for review by the DISPATCH
   Working Group, or the working group, mailing list, or expert
   designated by the RAI Area Directors if the DISPATCH Working Group
   has closed.  Proposals should include a strong motivational section,
   a thorough description of the proposed syntax and semantics, event
   package considerations, security considerations, and examples of
   usage.  Authors should submit their proposals as individual Internet-
   Drafts, and post an announcement to the working group mailing list to
   begin discussion.  The DISPATCH Working Group will determine if a
   proposed package is a) an appropriate usage of SIP which should be
   spun into a new effort, b) applicable to SIP but not sufficiently
   interesting, general, or in-scope to adopt as a working group effort,
   c) contrary to similar work chartered in an existing effort, or d)
   recommended to be adopted as or merged with chartered work elsewhere
   in RAI.=20

Are we still trying to prove somehow that this is generally applicable, for=
 which I strongly suspect the answer is NO and it never will be no matter h=
ow much manipulation it requires.

Are are we trying to identify that there is some error in the documentation=
 that needs correction?

regards

Keith

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Avasarala=20
> Ranjit-A20990
> Sent: Monday, February 08, 2010 10:09 AM
> To: Paul Kyzivat
> Cc: DISPATCH
> Subject: Re: [dispatch] Comments=20
> requestedondraft-avasarala-dispatch-comm-div-notification
>=20
> Hi Paul
>=20
> My comments <Ranjit-Feb08>=20
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, February 08, 2010 4:36 AM
> To: Avasarala Ranjit-A20990
> Cc: Elwell, John; DISPATCH
> Subject: Re: [dispatch] Comments
> requestedondraft-avasarala-dispatch-comm-div-notification
>=20
> inline...
>=20
> Avasarala Ranjit-A20990 wrote:
>=20
> >> Assuming sip:alice@office.domain.com is not an AOR, but=20
> rather is a=20
> >> registered contact to sip:alice@domain.com, where in the=20
> routing of=20
> >> the call does the server that evaluates this rule get control?
> >>
> >> <Ranjit> The server looks for the incoming caller identity and the=20
> >> destination identity to see if there is a matching rule configured.
> >=20
> > I guess my question wasn't clear.
> > Where in the routing sequence does this server reside?
> >=20
> > <Ranjit-Jan27> The server could be the Application Server where the=20
> > diversion rules reside and get executed.
> >=20
> > If it is positioned before the contact routing has=20
> occurred, then it=20
> > can't know which contact was chosen.
> > <Ranjit-Jan27th> No it would be before that.=20
>=20
> > Since this proposal is coming, more or less, from IMS I=20
> assume you are
>=20
> > imagining this is an IMS Application Server. IIRC, those get control
> > *before* contact routing is done by the S-CSCF. So how would the AS=20
> > know if a particular contact was chosen. (I take some issue with=20
> > calling contact routing "diversion", but lets not discuss that right
> > now.)
> >=20
> > <Ranjit-Jan27th> Yes you are right. This proposal is more=20
> from a IMS=20
> > and 3GPP perspective and I am trying to generalize it for IETF.
>=20
> OK. So once again I have to ask for a description of the system
> architecture(s) in which this event package makes sense -=20
> where the notifier would have the knowledge to make the=20
> proposed kind of notifications.
>=20
> <Ranjit-Feb08> The proposed I-D is actually for registering=20
> the proposed event package for the communication diversion=20
> service described in 3GPP TS 24.604. So here the notifier=20
> would the AS where the communication diversion service is=20
> running. So yes it would have knowledge of on whose behalf it=20
> is Doing the diversion and to whom it should notify.
>=20
> >> If the registered=20
> >>          contact is not an AoR, then the call cannot be=20
> routed to it=20
> >> and
> >=20
> > Huh? Whether the registered contact is an AOR has no bearing on=20
> > whether its possible to route to it. And, its in general=20
> impossible to
>=20
> > determine by looking whether a URI is an AOR or not.
> >=20
> > <Ranjit-Jan27th> Agree with you on this.
> >=20
> >> hence will be routed to the main AoR i.e=20
> sip:alice@domain.com and the
>=20
> >> rule would not be executed.
> >=20
> > That statement makes me even more confused. I have no idea what it=20
> > means.
> >=20
> > <Ranjit-Jan27th> Here I mean, the notifications would be=20
> sent to the=20
> > main AoR with the details of "diverting-user", "diverted-to-user"
> > being part Of the XML package. The notifications need not=20
> be sent to=20
> > the registered contact. For e.g if sip:alice@domain.com is=20
> the AoR and
>=20
> > diversions are taking Place for sip:alice@office.domain.com, then=20
> > notifications will be sent to sip:alice@domain.com only and not to=20
> > sip:alice@domain.com. This will simplify things I feel !!
>=20
> I'm am getting more confused, not less.
>=20
> I think the draft needs some significant rewriting in terms=20
> of concepts and terminology, as well as assumed architecture,=20
> in order to be comprehensible in any context other that pure=20
> IMS. And frankly I don't understand it even if I restrict the=20
> context to IMS.
>=20
> <Ranjit-Feb08> Actually the terminology and the service in=20
> general is explained in 24.604 (Section 4.10). So we felt=20
> that this I-D would just Register the proposed event package.
>=20
> >> But if sip:alice@office.domain.com is an AoR (e.g. GRUU), then the=20
> >> call could be routed and if there is adiversion rule there,
> >>          the diversion rule would get executed and a notification=20
> >> generated towards that AoR.
> >=20
> > This is getting more and more confusing. So I have even more desire=20
> > for at least a detailed use case that shows all the players, their=20
> > addresses and relationships to one another, and the call flow.
> >=20
> > <Ranjit-Jan27th> as I mentioned above, to simplify things,=20
> lets assume
>=20
> > that notifications always go to the main AoR i.e the Public User Bu=20
> > Identity (in IMS)
>=20
> > And not to any of the registered AoR irrespective of who=20
> the diverting
>=20
> > user is. So here even tough the divering user is=20
> sip:alice@office or=20
> > sip:alice@home, the notifications always go to sip:alice@domain.com.
>=20
> How does that simplify anything? If anything that makes it worse.=20
> Notifications go to the subscriber, and the content of that=20
> notification should in general have nothing to do with who=20
> the subscriber is. (But may be influenced by authorization=20
> policy.) Notifications are always in-dialog, to the remote=20
> target of that dialog, and hence pretty much never go to an AOR.
>=20
> <Ranjit-Feb08> True. Notification go to the subscribed entity=20
> as shown in the example in 24.604.=20
>=20
> >> How would it work in cases where the address for Alice's=20
> office phone
>=20
> >> is a dynamically assigned ip address rather than a dns name?
> >>
> >> <Ranjit> Since the notification gets generated dynamically=20
> when ever=20
> >> the diversion occurs, the new ip address is taken prior to sending=20
> >> the
> >=20
> >> notification.
> >=20
> > Again it seems my question wasn't clear.
> > You are showing rules that contain the contact addresses of=20
> particular
>=20
> > devices. If the contact URIs registered by those devices contain=20
> > dynamically assigned addresses, how would a subscriber know how to=20
> > construct the rule that selects a particular device? Or if it is=20
> > receiving all the diversions, how would it relate the=20
> reported results
>=20
> > to particular devices?
> >=20
> > <Ranjit-Jan27th> if the notifications are sent to PUI, then the=20
> > contact addresses are specified as URLs. So there is no=20
> need to send=20
> > any notifications to individual contact adsresses
>=20
> You are missing my point entirely. I'm not asking about where=20
> the notifications are *sent* (which is irrelevant). I'm=20
> asking about the
> *content* of those notifications, and of the filters that=20
> select which ones the subscriber is interested in.
>=20
> <Ranjit-Feb08> The content of notifications are the elements=20
> specified in the event pacakge. The example shows both a=20
> sample notification and the elements that can be used as=20
> filters to control the content of notifications. E.g. Section=20
> 4.10.1.1.1.2 of 24.604
>=20
> Assume for the moment that we have AOR sip:alice@SSP.com.
> And registered with that AOR are contact addresses=20
> sip:alice@mobile123.SSP.com and sip:alice@office.domain.com.
>=20
> Some UA (I'll not say which) is concerned about diversions=20
> related to that AOR. It subscribes to the comm-div package=20
> that sip:alice@SSP.com.=20
> It may include some sort of filter in the subscription.
>=20
> If a request comes in to the AOR, and neither of the contacts=20
> responds, a server serving that AOR (perhaps an AS) may=20
> eventually decide to forward the request to a VM server. I=20
> can see how this would be considered a real diversion, and=20
> how the server would know about it. So it can generate a=20
> NOTIFY reporting this diversion.
>=20
> <Ranjit-Feb08> True. So here the notification would be=20
> towards sip:alice@SSP.com with=20
> entity=3D"sip:alice@mobile123.SSP.com OR=20
> entity=3Dsip:alice@office.SSP.com on whose behlaf the diversion=20
> took place or where the call landed.=20
>=20
> Next, its possible that Alice, from her mobile phone,=20
> establishes a forwarding rule, that sends all incoming calls=20
> to the AOR to sip:bob@SSP.com. The server would apply this=20
> rule, as policy, to all incoming requests to the AOR,=20
> bypassing contact routing. Again its unremarkable that it=20
> would be able to generate a notification when carrying out=20
> such a retargeting.
>=20
> <Ranjit-Feb08> so here if Alice gets notified whenever a=20
> incoming call to her gets forwarded to Bob. If she sets a=20
> filter criteria like timeofday, then only incoming calls=20
> during that time get notified and not others. So the=20
> notifications depend on the target of the incoming call and=20
> other specified criteria.=20
>=20
> None of that requires any filter on the address at which=20
> diversion takes place, because it takes place exactly on=20
> diversion of requests to the same URI to which the=20
> subscription was made.
>=20
> <Ranjit-Feb08> why not? Filters can be set based on time,=20
> range of time or particular day or context (alice is OOO, etc)
>=20
> Your filter package implies that the notifier can generate=20
> notifications for cases where the diversion takes place from=20
> some URI *other* than the one for which the subscriptions is=20
> made. I am guessing you mean cases when the diversion takes=20
> place after the call has been retargeted to one of the=20
> contacts. I have two problems with that:
>=20
> <Ranjit-Feb08> No No. Notifier generates notifications only=20
> for diversions that take place on behalf of the subscribed=20
> entity and not others. So here sip:alice@SSP.com subscribes=20
> to diversions taking place at any of its contacts and get=20
> notified based on filter criteria. The filter critera can=20
> specify which contacts the notification is required for and=20
> which of those notifications are not required.=20
>=20
> 1) how would the server for sip:alice@SSP.com know that a=20
> diversion has occurred on a request targetted to=20
> sip:alice@office.domain.com?
>=20
> 2) even if it had a way to discover that, how would a=20
> subscriber know the URI sip:alice@office.domain.com in order=20
> to place it in a filter?
>=20
> But I find I am repeating myself (see below from before.)=20
> There must be some fundamental difference in perspective that=20
> is preventing us from understanding one another.
>=20
> > This relates, in a way, to whether contact routing is diversion or
> not.=20
> > Typically what I think of as real diversion would be to a URI that
> > *is* an AOR, and hence stable and knowable by those=20
> formulating rules.
>=20
> > But typical contact routing would involve URIs that aren't very
> meaningful.
> >=20
> > If you want to be able to write rules like: "let me know=20
> when the call
>=20
> > is routed to the phone in my office rather than the one in my home"
> > (where both have the same "phone number"), then you are=20
> going to have=20
> > to find a way to give those stable URIs, like=20
> > sip:alice@office.domain.com and sip:alice@home.domain.com.=20
> AFAIK that=20
> > isn't the norm, so if you are assuming it then please make=20
> that clear.
>=20
> 	Thanks,
> 	Paul
>=20
> Regards
> Ranjit
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

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Subject: Re: [dispatch] Comments	requestedondraft-avasarala-dispatch-comm-div-notification
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DRAGE, Keith (Keith) wrote:
> Can I understand where this discussion is going?
> 
> The relevant paragraph of http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-04.txt is:
> 
>    Individuals may wish to publish SIP Event packages that they believe
>    fall outside the scope of any chartered work currently in RAI.
>    Individual proposals for registration of a SIP event package MUST
>    first be published as Internet-drafts for review by the DISPATCH
>    Working Group, or the working group, mailing list, or expert
>    designated by the RAI Area Directors if the DISPATCH Working Group
>    has closed.  Proposals should include a strong motivational section,
>    a thorough description of the proposed syntax and semantics, event
>    package considerations, security considerations, and examples of
>    usage.  Authors should submit their proposals as individual Internet-
>    Drafts, and post an announcement to the working group mailing list to
>    begin discussion.  The DISPATCH Working Group will determine if a
>    proposed package is a) an appropriate usage of SIP which should be
>    spun into a new effort, b) applicable to SIP but not sufficiently
>    interesting, general, or in-scope to adopt as a working group effort,
>    c) contrary to similar work chartered in an existing effort, or d)
>    recommended to be adopted as or merged with chartered work elsewhere
>    in RAI. 
> 
> Are we still trying to prove somehow that this is generally applicable, for which I strongly suspect the answer is NO and it never will be no matter how much manipulation it requires.

I started out with the working assumption that this could be of general 
interest. I still think something addressing the same general area might 
be. But I am reaching the conclusion that the authors have no particular 
interest in generalizing it.

So I think it is probably better viewed as something specific for the 
IMS community. If I understand the procedures, it would be ok to publish 
this as such, but think the text should be very clear about the area of 
applicability. As currently worded I think people outside the IMS 
community might attempt to implement it without the guidance of 
additional IMS documents. IMO if they did so the likelihood of 
interoperable implementations would be slim to none.

	Thanks,
	Paul

> Are are we trying to identify that there is some error in the documentation that needs correction?
> 
> regards
> 
> Keith
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Avasarala 
>> Ranjit-A20990
>> Sent: Monday, February 08, 2010 10:09 AM
>> To: Paul Kyzivat
>> Cc: DISPATCH
>> Subject: Re: [dispatch] Comments 
>> requestedondraft-avasarala-dispatch-comm-div-notification
>>
>> Hi Paul
>>
>> My comments <Ranjit-Feb08> 
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, February 08, 2010 4:36 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Elwell, John; DISPATCH
>> Subject: Re: [dispatch] Comments
>> requestedondraft-avasarala-dispatch-comm-div-notification
>>
>> inline...
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>>> Assuming sip:alice@office.domain.com is not an AOR, but 
>> rather is a 
>>>> registered contact to sip:alice@domain.com, where in the 
>> routing of 
>>>> the call does the server that evaluates this rule get control?
>>>>
>>>> <Ranjit> The server looks for the incoming caller identity and the 
>>>> destination identity to see if there is a matching rule configured.
>>> I guess my question wasn't clear.
>>> Where in the routing sequence does this server reside?
>>>
>>> <Ranjit-Jan27> The server could be the Application Server where the 
>>> diversion rules reside and get executed.
>>>
>>> If it is positioned before the contact routing has 
>> occurred, then it 
>>> can't know which contact was chosen.
>>> <Ranjit-Jan27th> No it would be before that. 
>>> Since this proposal is coming, more or less, from IMS I 
>> assume you are
>>
>>> imagining this is an IMS Application Server. IIRC, those get control
>>> *before* contact routing is done by the S-CSCF. So how would the AS 
>>> know if a particular contact was chosen. (I take some issue with 
>>> calling contact routing "diversion", but lets not discuss that right
>>> now.)
>>>
>>> <Ranjit-Jan27th> Yes you are right. This proposal is more 
>> from a IMS 
>>> and 3GPP perspective and I am trying to generalize it for IETF.
>> OK. So once again I have to ask for a description of the system
>> architecture(s) in which this event package makes sense - 
>> where the notifier would have the knowledge to make the 
>> proposed kind of notifications.
>>
>> <Ranjit-Feb08> The proposed I-D is actually for registering 
>> the proposed event package for the communication diversion 
>> service described in 3GPP TS 24.604. So here the notifier 
>> would the AS where the communication diversion service is 
>> running. So yes it would have knowledge of on whose behalf it 
>> is Doing the diversion and to whom it should notify.
>>
>>>> If the registered 
>>>>          contact is not an AoR, then the call cannot be 
>> routed to it 
>>>> and
>>> Huh? Whether the registered contact is an AOR has no bearing on 
>>> whether its possible to route to it. And, its in general 
>> impossible to
>>
>>> determine by looking whether a URI is an AOR or not.
>>>
>>> <Ranjit-Jan27th> Agree with you on this.
>>>
>>>> hence will be routed to the main AoR i.e 
>> sip:alice@domain.com and the
>>
>>>> rule would not be executed.
>>> That statement makes me even more confused. I have no idea what it 
>>> means.
>>>
>>> <Ranjit-Jan27th> Here I mean, the notifications would be 
>> sent to the 
>>> main AoR with the details of "diverting-user", "diverted-to-user"
>>> being part Of the XML package. The notifications need not 
>> be sent to 
>>> the registered contact. For e.g if sip:alice@domain.com is 
>> the AoR and
>>
>>> diversions are taking Place for sip:alice@office.domain.com, then 
>>> notifications will be sent to sip:alice@domain.com only and not to 
>>> sip:alice@domain.com. This will simplify things I feel !!
>> I'm am getting more confused, not less.
>>
>> I think the draft needs some significant rewriting in terms 
>> of concepts and terminology, as well as assumed architecture, 
>> in order to be comprehensible in any context other that pure 
>> IMS. And frankly I don't understand it even if I restrict the 
>> context to IMS.
>>
>> <Ranjit-Feb08> Actually the terminology and the service in 
>> general is explained in 24.604 (Section 4.10). So we felt 
>> that this I-D would just Register the proposed event package.
>>
>>>> But if sip:alice@office.domain.com is an AoR (e.g. GRUU), then the 
>>>> call could be routed and if there is adiversion rule there,
>>>>          the diversion rule would get executed and a notification 
>>>> generated towards that AoR.
>>> This is getting more and more confusing. So I have even more desire 
>>> for at least a detailed use case that shows all the players, their 
>>> addresses and relationships to one another, and the call flow.
>>>
>>> <Ranjit-Jan27th> as I mentioned above, to simplify things, 
>> lets assume
>>
>>> that notifications always go to the main AoR i.e the Public User Bu 
>>> Identity (in IMS)
>>> And not to any of the registered AoR irrespective of who 
>> the diverting
>>
>>> user is. So here even tough the divering user is 
>> sip:alice@office or 
>>> sip:alice@home, the notifications always go to sip:alice@domain.com.
>> How does that simplify anything? If anything that makes it worse. 
>> Notifications go to the subscriber, and the content of that 
>> notification should in general have nothing to do with who 
>> the subscriber is. (But may be influenced by authorization 
>> policy.) Notifications are always in-dialog, to the remote 
>> target of that dialog, and hence pretty much never go to an AOR.
>>
>> <Ranjit-Feb08> True. Notification go to the subscribed entity 
>> as shown in the example in 24.604. 
>>
>>>> How would it work in cases where the address for Alice's 
>> office phone
>>
>>>> is a dynamically assigned ip address rather than a dns name?
>>>>
>>>> <Ranjit> Since the notification gets generated dynamically 
>> when ever 
>>>> the diversion occurs, the new ip address is taken prior to sending 
>>>> the
>>>> notification.
>>> Again it seems my question wasn't clear.
>>> You are showing rules that contain the contact addresses of 
>> particular
>>
>>> devices. If the contact URIs registered by those devices contain 
>>> dynamically assigned addresses, how would a subscriber know how to 
>>> construct the rule that selects a particular device? Or if it is 
>>> receiving all the diversions, how would it relate the 
>> reported results
>>
>>> to particular devices?
>>>
>>> <Ranjit-Jan27th> if the notifications are sent to PUI, then the 
>>> contact addresses are specified as URLs. So there is no 
>> need to send 
>>> any notifications to individual contact adsresses
>> You are missing my point entirely. I'm not asking about where 
>> the notifications are *sent* (which is irrelevant). I'm 
>> asking about the
>> *content* of those notifications, and of the filters that 
>> select which ones the subscriber is interested in.
>>
>> <Ranjit-Feb08> The content of notifications are the elements 
>> specified in the event pacakge. The example shows both a 
>> sample notification and the elements that can be used as 
>> filters to control the content of notifications. E.g. Section 
>> 4.10.1.1.1.2 of 24.604
>>
>> Assume for the moment that we have AOR sip:alice@SSP.com.
>> And registered with that AOR are contact addresses 
>> sip:alice@mobile123.SSP.com and sip:alice@office.domain.com.
>>
>> Some UA (I'll not say which) is concerned about diversions 
>> related to that AOR. It subscribes to the comm-div package 
>> that sip:alice@SSP.com. 
>> It may include some sort of filter in the subscription.
>>
>> If a request comes in to the AOR, and neither of the contacts 
>> responds, a server serving that AOR (perhaps an AS) may 
>> eventually decide to forward the request to a VM server. I 
>> can see how this would be considered a real diversion, and 
>> how the server would know about it. So it can generate a 
>> NOTIFY reporting this diversion.
>>
>> <Ranjit-Feb08> True. So here the notification would be 
>> towards sip:alice@SSP.com with 
>> entity="sip:alice@mobile123.SSP.com OR 
>> entity=sip:alice@office.SSP.com on whose behlaf the diversion 
>> took place or where the call landed. 
>>
>> Next, its possible that Alice, from her mobile phone, 
>> establishes a forwarding rule, that sends all incoming calls 
>> to the AOR to sip:bob@SSP.com. The server would apply this 
>> rule, as policy, to all incoming requests to the AOR, 
>> bypassing contact routing. Again its unremarkable that it 
>> would be able to generate a notification when carrying out 
>> such a retargeting.
>>
>> <Ranjit-Feb08> so here if Alice gets notified whenever a 
>> incoming call to her gets forwarded to Bob. If she sets a 
>> filter criteria like timeofday, then only incoming calls 
>> during that time get notified and not others. So the 
>> notifications depend on the target of the incoming call and 
>> other specified criteria. 
>>
>> None of that requires any filter on the address at which 
>> diversion takes place, because it takes place exactly on 
>> diversion of requests to the same URI to which the 
>> subscription was made.
>>
>> <Ranjit-Feb08> why not? Filters can be set based on time, 
>> range of time or particular day or context (alice is OOO, etc)
>>
>> Your filter package implies that the notifier can generate 
>> notifications for cases where the diversion takes place from 
>> some URI *other* than the one for which the subscriptions is 
>> made. I am guessing you mean cases when the diversion takes 
>> place after the call has been retargeted to one of the 
>> contacts. I have two problems with that:
>>
>> <Ranjit-Feb08> No No. Notifier generates notifications only 
>> for diversions that take place on behalf of the subscribed 
>> entity and not others. So here sip:alice@SSP.com subscribes 
>> to diversions taking place at any of its contacts and get 
>> notified based on filter criteria. The filter critera can 
>> specify which contacts the notification is required for and 
>> which of those notifications are not required. 
>>
>> 1) how would the server for sip:alice@SSP.com know that a 
>> diversion has occurred on a request targetted to 
>> sip:alice@office.domain.com?
>>
>> 2) even if it had a way to discover that, how would a 
>> subscriber know the URI sip:alice@office.domain.com in order 
>> to place it in a filter?
>>
>> But I find I am repeating myself (see below from before.) 
>> There must be some fundamental difference in perspective that 
>> is preventing us from understanding one another.
>>
>>> This relates, in a way, to whether contact routing is diversion or
>> not. 
>>> Typically what I think of as real diversion would be to a URI that
>>> *is* an AOR, and hence stable and knowable by those 
>> formulating rules.
>>
>>> But typical contact routing would involve URIs that aren't very
>> meaningful.
>>> If you want to be able to write rules like: "let me know 
>> when the call
>>
>>> is routed to the phone in my office rather than the one in my home"
>>> (where both have the same "phone number"), then you are 
>> going to have 
>>> to find a way to give those stable URIs, like 
>>> sip:alice@office.domain.com and sip:alice@home.domain.com. 
>> AFAIK that 
>>> isn't the norm, so if you are assuming it then please make 
>> that clear.
>>
>> 	Thanks,
>> 	Paul
>>
>> Regards
>> Ranjit
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From Peter.Dawes@vodafone.com  Mon Feb  8 07:57:23 2010
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Thread-Topic: Re: I-DAction:draft-kaplan-dispatch-session-id-00.txt
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From: "Dawes, Peter, VF-Group" <Peter.Dawes@vodafone.com>
To: <dispatch@ietf.org>
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Cc: "Dawes, Peter, VF-Group" <Peter.Dawes@vodafone.com>, gonzalo.camarillo@ericsson.com, HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de, Gerold.Pinker@telekom.de
Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Laura,
What you describe below about debug-id is the correct understanding. I
am not convinced that using session-id for troubleshooting is practical
because it implies that everything has to be logged everywhere in the
network all the time so that you can, for example, look at any
signalling that happened two days ago. The debug-id is inserted as
required, and also policed at a couple of entities (first proxy,
registrar) to take the burden of deciding whether or not to log away
from the rest of the network. The debug-id is also based on 3GPP and OMA
requirements, as I mentioned, but these requirements are, I believe,
necessary to minimize burden on the network.=20

Starting to read through the threads that consider the idea from Paul
Kyzivat:

> IMO use cases should be gathered and then bucketed into sets that can=20

> be

>=20

> satisfied by a single mechanism, without assuming initially how many=20

> buckets there will be.

the beginning of a list of use cases seems to be 3rd-party CC,
correlating multiple parallel dialogs, forked requests, requests that
pass through B2BUAs. I guess that if an unambiguous test can be
expressed for whether a particular session belongs to a set of related
sessions, then one header to hold the id that correlates them is enough,
even if populating it is difficult for some use cases. For example, I
imagine that the debug-id draft could be re-written to use session-id
(if session-id is chosen as the single "correlation id" header) to hold
the "correlation id" with an added header field parameter to indicate
the extra functionality for reducing the network burden of
troubleshooting. This would satisfy the troubleshooting and 3GPP/OMA
requirements without limiting the meaning of the "correlation id" to
troubleshooting.=20

Like you, I would like to find some way forward with the e2e correlation
topic (Spencer is right. BOF, CLF whatever...we have to find some way to
come forward with the e2e correlation.). Also, our network guys consider
session-id to be useful, but I would also like to satisfy some OMA/3GPP
requirements that it does not cover. =20

Best Regards,
Peter






Message: 3

Date: Fri, 5 Feb 2010 17:21:39 +0100

From: <L.Liess@telekom.de>

Subject: Re: [dispatch]

I-DAction:draft-kaplan-dispatch-session-id-00.txt

To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>,

<spencer@wonderhamster.org>

Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,

HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de

Message-ID:

<40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>

Content-Type: text/plain; charset=3D"iso-8859-1"

Hi Peter,

My understanding is that session-id and debug-id cover different sets of
requirements.=20

Session-id is in every message. If people want to find out what happened
two days before, they can look at the old logs and the session-id is
there in every message to help them to correlate e2e. For the debug-id
to be there you have to know in advance that and which e2e traffic you
would like to have correlated. The debug-id is activated "on-demand" and
has no performance impact when it is not active, session-id is active
all the time. The advantage of the debug-id is that there is no
performance impact when debugging is not needed. Is my understanding
correct?=20

I like the debug-id stuff, it's really smart and we looked at it. But
our troubleshooting and security guys want the session-id, for the
reason above. I assume every service provider has its own needs and
requirements.=20

Spencer is right. BOF, CLF whatever...we have to find some way to come
forward with the e2e correlation.=20

Thanks a lot

Laura

=20

=20

> -----Original Message-----

> From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
<mailto:Peter.Dawes@vodafone.com> ]

> Sent: Friday, February 05, 2010 4:00 PM

> To: dispatch@ietf.org

> Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;=20

> HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold

> Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt

>=20

> Hello,

> Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03

> +0200) as

> "1) We have Hadriel's draft, 2) We also have the following draft,=20

> which eventually led to the disaggregated media work:

> http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
<http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>=20

> -correlati

> on-01.txt 3) We also have the SIP-XMPP work, which may also need to=20

> correlate sessions somehow."

>=20

> I would like to add

> http://tools.ietf.org/html/draft-dawes-sipping-debug-02
<http://tools.ietf.org/html/draft-dawes-sipping-debug-02>  to this=20

> discussion of use cases for some form of correlation identifier. This=20

> draft relates to debugging/troubleshooting to meet documented=20

> requirements from 3GPP and OMA, and proposes a kind of correlation=20

> identifier in a new header.

>=20

> Regards,

> Peter

>=20

>=20

> Message: 2

> Date: Fri, 22 Jan 2010 11:05:57 -0500

> From: Paul Kyzivat <pkyzivat@cisco.com>

> Subject: Re: [dispatch] FW:

> I-DAction:draft-kaplan-dispatch-session-id-00.txt

> To: L.Liess@telekom.de

> Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,

> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,

> Gerold.Pinker@telekom.de

> Message-ID: <4B59CCE5.2010202@cisco.com>

> Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed

>=20

>=20

>=20

> L.Liess@telekom.de wrote:

> > Paul,

> >=20

> > Thank you for the examples. The scenarios can be really complex. I

> hope we can get these correlation identifiers, whatever their names,=20

> to work correctly in all cases....

> >=20

> > I am not sure I understood your message correctly. Do you say we

> should have two different identifiers because we use them for two=20

> different things?

>=20

> I think there may be more than two distinct usages that will require=20

> different ones.

>=20

> IMO use cases should be gathered and then bucketed into sets that can=20

> be

>=20

> satisfied by a single mechanism, without assuming initially how many=20

> buckets there will be.

>=20

> (But I'm still on record as not being convinced that a correlation id=20

> is

>=20

> needed *at all* for the case when multiple sip UAs are participating=20

> on behalf of a single user in a call.)

>=20

> > Thanks a lot

> > Laura

> >=20

>=20

=20
=20

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
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Subject: Re: [dispatch] Comments	requestedondraft-avasarala-dispatch-comm-div-notification
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Hi Paul, Keith

True. This event package is intended for CDIV procedures outlined in
24.604. The specific CDIV procedures and the event package are already
described in 24.604. The schema too is well explained. But 3GPP wanted
the event package and the schema to be registered through a
informational or standards track RFC as per IETF.

So we submitted an I-D proposing the CDIV notification event package
putting reference to 24.604.=20

Thanks=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Monday, February 08, 2010 7:40 PM
To: DRAGE, Keith (Keith)
Cc: Avasarala Ranjit-A20990; DISPATCH
Subject: Re: [dispatch] Comments
requestedondraft-avasarala-dispatch-comm-div-notification



DRAGE, Keith (Keith) wrote:
> Can I understand where this discussion is going?
>=20
> The relevant paragraph of
http://tools.ietf.org/id/draft-peterson-rai-rfc3427bis-04.txt is:
>=20
>    Individuals may wish to publish SIP Event packages that they
believe
>    fall outside the scope of any chartered work currently in RAI.
>    Individual proposals for registration of a SIP event package MUST
>    first be published as Internet-drafts for review by the DISPATCH
>    Working Group, or the working group, mailing list, or expert
>    designated by the RAI Area Directors if the DISPATCH Working Group
>    has closed.  Proposals should include a strong motivational
section,
>    a thorough description of the proposed syntax and semantics, event
>    package considerations, security considerations, and examples of
>    usage.  Authors should submit their proposals as individual
Internet-
>    Drafts, and post an announcement to the working group mailing list
to
>    begin discussion.  The DISPATCH Working Group will determine if a
>    proposed package is a) an appropriate usage of SIP which should be
>    spun into a new effort, b) applicable to SIP but not sufficiently
>    interesting, general, or in-scope to adopt as a working group
effort,
>    c) contrary to similar work chartered in an existing effort, or d)
>    recommended to be adopted as or merged with chartered work
elsewhere
>    in RAI.=20
>=20
> Are we still trying to prove somehow that this is generally
applicable, for which I strongly suspect the answer is NO and it never
will be no matter how much manipulation it requires.

I started out with the working assumption that this could be of general
interest. I still think something addressing the same general area might
be. But I am reaching the conclusion that the authors have no particular
interest in generalizing it.

So I think it is probably better viewed as something specific for the
IMS community. If I understand the procedures, it would be ok to publish
this as such, but think the text should be very clear about the area of
applicability. As currently worded I think people outside the IMS
community might attempt to implement it without the guidance of
additional IMS documents. IMO if they did so the likelihood of
interoperable implementations would be slim to none.

	Thanks,
	Paul

> Are are we trying to identify that there is some error in the
documentation that needs correction?
>=20
> regards
>=20
> Keith
>=20
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Avasarala=20
>> Ranjit-A20990
>> Sent: Monday, February 08, 2010 10:09 AM
>> To: Paul Kyzivat
>> Cc: DISPATCH
>> Subject: Re: [dispatch] Comments
>> requestedondraft-avasarala-dispatch-comm-div-notification
>>
>> Hi Paul
>>
>> My comments <Ranjit-Feb08>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Monday, February 08, 2010 4:36 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Elwell, John; DISPATCH
>> Subject: Re: [dispatch] Comments
>> requestedondraft-avasarala-dispatch-comm-div-notification
>>
>> inline...
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>>> Assuming sip:alice@office.domain.com is not an AOR, but
>> rather is a
>>>> registered contact to sip:alice@domain.com, where in the
>> routing of
>>>> the call does the server that evaluates this rule get control?
>>>>
>>>> <Ranjit> The server looks for the incoming caller identity and the=20
>>>> destination identity to see if there is a matching rule configured.
>>> I guess my question wasn't clear.
>>> Where in the routing sequence does this server reside?
>>>
>>> <Ranjit-Jan27> The server could be the Application Server where the=20
>>> diversion rules reside and get executed.
>>>
>>> If it is positioned before the contact routing has
>> occurred, then it
>>> can't know which contact was chosen.
>>> <Ranjit-Jan27th> No it would be before that.=20
>>> Since this proposal is coming, more or less, from IMS I
>> assume you are
>>
>>> imagining this is an IMS Application Server. IIRC, those get control
>>> *before* contact routing is done by the S-CSCF. So how would the AS=20
>>> know if a particular contact was chosen. (I take some issue with=20
>>> calling contact routing "diversion", but lets not discuss that right
>>> now.)
>>>
>>> <Ranjit-Jan27th> Yes you are right. This proposal is more
>> from a IMS
>>> and 3GPP perspective and I am trying to generalize it for IETF.
>> OK. So once again I have to ask for a description of the system
>> architecture(s) in which this event package makes sense - where the=20
>> notifier would have the knowledge to make the proposed kind of=20
>> notifications.
>>
>> <Ranjit-Feb08> The proposed I-D is actually for registering the=20
>> proposed event package for the communication diversion service=20
>> described in 3GPP TS 24.604. So here the notifier would the AS where=20
>> the communication diversion service is running. So yes it would have=20
>> knowledge of on whose behalf it is Doing the diversion and to whom it

>> should notify.
>>
>>>> If the registered=20
>>>>          contact is not an AoR, then the call cannot be
>> routed to it
>>>> and
>>> Huh? Whether the registered contact is an AOR has no bearing on=20
>>> whether its possible to route to it. And, its in general
>> impossible to
>>
>>> determine by looking whether a URI is an AOR or not.
>>>
>>> <Ranjit-Jan27th> Agree with you on this.
>>>
>>>> hence will be routed to the main AoR i.e
>> sip:alice@domain.com and the
>>
>>>> rule would not be executed.
>>> That statement makes me even more confused. I have no idea what it=20
>>> means.
>>>
>>> <Ranjit-Jan27th> Here I mean, the notifications would be
>> sent to the
>>> main AoR with the details of "diverting-user", "diverted-to-user"
>>> being part Of the XML package. The notifications need not
>> be sent to
>>> the registered contact. For e.g if sip:alice@domain.com is
>> the AoR and
>>
>>> diversions are taking Place for sip:alice@office.domain.com, then=20
>>> notifications will be sent to sip:alice@domain.com only and not to=20
>>> sip:alice@domain.com. This will simplify things I feel !!
>> I'm am getting more confused, not less.
>>
>> I think the draft needs some significant rewriting in terms of=20
>> concepts and terminology, as well as assumed architecture, in order=20
>> to be comprehensible in any context other that pure IMS. And frankly=20
>> I don't understand it even if I restrict the context to IMS.
>>
>> <Ranjit-Feb08> Actually the terminology and the service in general is

>> explained in 24.604 (Section 4.10). So we felt that this I-D would=20
>> just Register the proposed event package.
>>
>>>> But if sip:alice@office.domain.com is an AoR (e.g. GRUU), then the=20
>>>> call could be routed and if there is adiversion rule there,
>>>>          the diversion rule would get executed and a notification=20
>>>> generated towards that AoR.
>>> This is getting more and more confusing. So I have even more desire=20
>>> for at least a detailed use case that shows all the players, their=20
>>> addresses and relationships to one another, and the call flow.
>>>
>>> <Ranjit-Jan27th> as I mentioned above, to simplify things,
>> lets assume
>>
>>> that notifications always go to the main AoR i.e the Public User Bu=20
>>> Identity (in IMS) And not to any of the registered AoR irrespective=20
>>> of who
>> the diverting
>>
>>> user is. So here even tough the divering user is
>> sip:alice@office or
>>> sip:alice@home, the notifications always go to sip:alice@domain.com.
>> How does that simplify anything? If anything that makes it worse.=20
>> Notifications go to the subscriber, and the content of that=20
>> notification should in general have nothing to do with who the=20
>> subscriber is. (But may be influenced by authorization
>> policy.) Notifications are always in-dialog, to the remote target of=20
>> that dialog, and hence pretty much never go to an AOR.
>>
>> <Ranjit-Feb08> True. Notification go to the subscribed entity as=20
>> shown in the example in 24.604.
>>
>>>> How would it work in cases where the address for Alice's
>> office phone
>>
>>>> is a dynamically assigned ip address rather than a dns name?
>>>>
>>>> <Ranjit> Since the notification gets generated dynamically
>> when ever
>>>> the diversion occurs, the new ip address is taken prior to sending=20
>>>> the notification.
>>> Again it seems my question wasn't clear.
>>> You are showing rules that contain the contact addresses of
>> particular
>>
>>> devices. If the contact URIs registered by those devices contain=20
>>> dynamically assigned addresses, how would a subscriber know how to=20
>>> construct the rule that selects a particular device? Or if it is=20
>>> receiving all the diversions, how would it relate the
>> reported results
>>
>>> to particular devices?
>>>
>>> <Ranjit-Jan27th> if the notifications are sent to PUI, then the=20
>>> contact addresses are specified as URLs. So there is no
>> need to send
>>> any notifications to individual contact adsresses
>> You are missing my point entirely. I'm not asking about where the=20
>> notifications are *sent* (which is irrelevant). I'm asking about the
>> *content* of those notifications, and of the filters that select=20
>> which ones the subscriber is interested in.
>>
>> <Ranjit-Feb08> The content of notifications are the elements=20
>> specified in the event pacakge. The example shows both a sample=20
>> notification and the elements that can be used as filters to control=20
>> the content of notifications. E.g. Section
>> 4.10.1.1.1.2 of 24.604
>>
>> Assume for the moment that we have AOR sip:alice@SSP.com.
>> And registered with that AOR are contact addresses=20
>> sip:alice@mobile123.SSP.com and sip:alice@office.domain.com.
>>
>> Some UA (I'll not say which) is concerned about diversions related to

>> that AOR. It subscribes to the comm-div package that=20
>> sip:alice@SSP.com.
>> It may include some sort of filter in the subscription.
>>
>> If a request comes in to the AOR, and neither of the contacts=20
>> responds, a server serving that AOR (perhaps an AS) may eventually=20
>> decide to forward the request to a VM server. I can see how this=20
>> would be considered a real diversion, and how the server would know=20
>> about it. So it can generate a NOTIFY reporting this diversion.
>>
>> <Ranjit-Feb08> True. So here the notification would be towards=20
>> sip:alice@SSP.com with entity=3D"sip:alice@mobile123.SSP.com OR=20
>> entity=3Dsip:alice@office.SSP.com on whose behlaf the diversion took=20
>> place or where the call landed.
>>
>> Next, its possible that Alice, from her mobile phone, establishes a=20
>> forwarding rule, that sends all incoming calls to the AOR to=20
>> sip:bob@SSP.com. The server would apply this rule, as policy, to all=20
>> incoming requests to the AOR, bypassing contact routing. Again its=20
>> unremarkable that it would be able to generate a notification when=20
>> carrying out such a retargeting.
>>
>> <Ranjit-Feb08> so here if Alice gets notified whenever a incoming=20
>> call to her gets forwarded to Bob. If she sets a filter criteria like

>> timeofday, then only incoming calls during that time get notified and

>> not others. So the notifications depend on the target of the incoming

>> call and other specified criteria.
>>
>> None of that requires any filter on the address at which diversion=20
>> takes place, because it takes place exactly on diversion of requests=20
>> to the same URI to which the subscription was made.
>>
>> <Ranjit-Feb08> why not? Filters can be set based on time, range of=20
>> time or particular day or context (alice is OOO, etc)
>>
>> Your filter package implies that the notifier can generate=20
>> notifications for cases where the diversion takes place from some URI

>> *other* than the one for which the subscriptions is made. I am=20
>> guessing you mean cases when the diversion takes place after the call

>> has been retargeted to one of the contacts. I have two problems with=20
>> that:
>>
>> <Ranjit-Feb08> No No. Notifier generates notifications only for=20
>> diversions that take place on behalf of the subscribed entity and not

>> others. So here sip:alice@SSP.com subscribes to diversions taking=20
>> place at any of its contacts and get notified based on filter=20
>> criteria. The filter critera can specify which contacts the=20
>> notification is required for and which of those notifications are not

>> required.
>>
>> 1) how would the server for sip:alice@SSP.com know that a diversion=20
>> has occurred on a request targetted to sip:alice@office.domain.com?
>>
>> 2) even if it had a way to discover that, how would a subscriber know

>> the URI sip:alice@office.domain.com in order to place it in a filter?
>>
>> But I find I am repeating myself (see below from before.) There must=20
>> be some fundamental difference in perspective that is preventing us=20
>> from understanding one another.
>>
>>> This relates, in a way, to whether contact routing is diversion or
>> not.=20
>>> Typically what I think of as real diversion would be to a URI that
>>> *is* an AOR, and hence stable and knowable by those
>> formulating rules.
>>
>>> But typical contact routing would involve URIs that aren't very
>> meaningful.
>>> If you want to be able to write rules like: "let me know
>> when the call
>>
>>> is routed to the phone in my office rather than the one in my home"
>>> (where both have the same "phone number"), then you are
>> going to have
>>> to find a way to give those stable URIs, like=20
>>> sip:alice@office.domain.com and sip:alice@home.domain.com.
>> AFAIK that
>>> isn't the norm, so if you are assuming it then please make
>> that clear.
>>
>> 	Thanks,
>> 	Paul
>>
>> Regards
>> Ranjit
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>

From dean.willis@softarmor.com  Mon Feb  8 14:38:41 2010
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On Tue, 2010-02-09 at 00:45 +0800, Avasarala Ranjit-A20990 wrote:
> Hi Paul, Keith
> 
> True. This event package is intended for CDIV procedures outlined in
> 24.604. The specific CDIV procedures and the event package are already
> described in 24.604. The schema too is well explained. But 3GPP wanted
> the event package and the schema to be registered through a
> informational or standards track RFC as per IETF.
> 
> So we submitted an I-D proposing the CDIV notification event package
> putting reference to 24.604. 
> 

So I think we're jut arguing about the applicability statement.


Something like "Applicability Statement: The event package defined in
this specification is designed to work in the context of IMS, using
specifics thereof which are defined in 3GPP specification 24.604.",
along with appropriate bibliography entries.

Further, the AS should go on to state how the target environment, in
general, differs from the "general SIP environment", specifically
discussing the assumptions about which servers know about diversion and
can act as notifiers.

--
Dean


From gonzalo.camarillo@ericsson.com  Tue Feb  9 01:15:24 2010
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Subject: [dispatch] Prorgessing draft-avasarala-dispatch-comm-div-notification-02
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Hi,

there have been a set of messages on the list providing comments on the
draft below:

http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-notification-02

As you know, the process for defining SIP event packages is documented here:

http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#section-4.1

Based on the feedback received, the authors need to decide whether they
want to generalize their solution so that is is generally applicable to
the public Internet or if they want to (further) clarify that this
mechanism is intended to work only in IMS networks that provide a CDIV
service.

If the authors choose the latter approach, we (the DISPATCH chairs) will
choose a "Designated Expert" who will check the draft against the 7
points described in the document above.

Cheers,

Gonzalo
DISPATCH co-chair

From drage@alcatel-lucent.com  Tue Feb  9 07:05:01 2010
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>, DISPATCH <dispatch@ietf.org>
Date: Tue, 9 Feb 2010 16:05:08 +0100
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At the moment, I think those of us looking at this from the 3GPP way of doi=
ng things do not see a more general application of this.

I do not have a problem with doing it more generally, but I guess the peopl=
e who see a clear use case for that need to speak up and identify their use=
 cases.=20

Otherwise we end up designing a general event package that no one else ever=
 uses. As opposed to the alternative of clearly stating the restricted use =
to a particular applicability, approving it, and moving on to something els=
e with our restricted amount of resources.

regards

Keith=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> Sent: Tuesday, February 09, 2010 9:16 AM
> To: DISPATCH
> Cc: Mary Barnes
> Subject: [dispatch] Prorgessing=20
> draft-avasarala-dispatch-comm-div-notification-02
>=20
> Hi,
>=20
> there have been a set of messages on the list providing=20
> comments on the draft below:
>=20
> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
> otification-02
>=20
> As you know, the process for defining SIP event packages is=20
> documented here:
>=20
> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
> ction-4.1
>=20
> Based on the feedback received, the authors need to decide=20
> whether they want to generalize their solution so that is is=20
> generally applicable to the public Internet or if they want=20
> to (further) clarify that this mechanism is intended to work=20
> only in IMS networks that provide a CDIV service.
>=20
> If the authors choose the latter approach, we (the DISPATCH=20
> chairs) will choose a "Designated Expert" who will check the=20
> draft against the 7 points described in the document above.
>=20
> Cheers,
>=20
> Gonzalo
> DISPATCH co-chair
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From L.Liess@telekom.de  Tue Feb  9 07:45:33 2010
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Hi Peter,=20

I agree. We should put together a first draft describing the use cases =
and, I think, the requirements for each use case. We already have a lot =
of stuff in the existing drafts.=20
However, Salvatore tried this about two years ago and it didn't work =
then. I think there were two reasons for this (just my personal =
opinion):=20

2) People involved with the issue (Hadriel, Dale, myself) had other, =
more urgent things to do then. (I think you were not involved in the =
discussion then, but I am not sure.)

1) The draft included the disagregated media correlation, which is quite =
difficult to compare with the other stuff and this made things quite =
complicated. Two or three weeks ago we discussed again the issue =
session-id vs. disagregated media correlation and Salvatore and I came =
to the conclusion that they are different and we need different =
identifiers for them. I don't know if other people agreed to this view.=20

I am not sure it is realistic and very usefull to try submitting a new =
draft about the use cases/requirements draft before Anaheim. I am now =
quite busy myself and I think other people have the same problem. I =
could offer to do main editor for the use cases draft, but after =
Anaheim. Or maybe we could begin with Sal's expired use cases draft.=20

I would propose that before Anaheim people interested in the issue try =
to read the drafts existing on this issue. Depending on who will attend =
Anaheim, we could meet there and discuss first the existing drafts, =
overlapping requirements/use cases and the differences.


What do you think?  Do you intend to attend Anaheim?=20

Laura =20

      =20

=20


> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Dawes, Peter, VF-Group
> Sent: Monday, February 08, 2010 4:58 PM
> To: dispatch@ietf.org
> Cc: Dawes, Peter, VF-Group; gonzalo.camarillo@ericsson.com;=20
> HKaplan@acmepacket.com; H=FClsemann, Martin; Pinker, Gerold
> Subject: Re: [dispatch]=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> Hi Laura,
> What you describe below about debug-id is the correct understanding. I
> am not convinced that using session-id for troubleshooting is=20
> practical
> because it implies that everything has to be logged everywhere in the
> network all the time so that you can, for example, look at any
> signalling that happened two days ago. The debug-id is inserted as
> required, and also policed at a couple of entities (first proxy,
> registrar) to take the burden of deciding whether or not to log away
> from the rest of the network. The debug-id is also based on=20
> 3GPP and OMA
> requirements, as I mentioned, but these requirements are, I believe,
> necessary to minimize burden on the network.=20
>=20
> Starting to read through the threads that consider the idea from Paul
> Kyzivat:
>=20
> > IMO use cases should be gathered and then bucketed into=20
> sets that can=20
>=20
> > be
>=20
> >=20
>=20
> > satisfied by a single mechanism, without assuming initially=20
> how many=20
>=20
> > buckets there will be.
>=20
> the beginning of a list of use cases seems to be 3rd-party CC,
> correlating multiple parallel dialogs, forked requests, requests that
> pass through B2BUAs. I guess that if an unambiguous test can be
> expressed for whether a particular session belongs to a set of related
> sessions, then one header to hold the id that correlates them=20
> is enough,
> even if populating it is difficult for some use cases. For example, I
> imagine that the debug-id draft could be re-written to use session-id
> (if session-id is chosen as the single "correlation id"=20
> header) to hold
> the "correlation id" with an added header field parameter to indicate
> the extra functionality for reducing the network burden of
> troubleshooting. This would satisfy the troubleshooting and 3GPP/OMA
> requirements without limiting the meaning of the "correlation id" to
> troubleshooting.=20
>=20
> Like you, I would like to find some way forward with the e2e=20
> correlation
> topic (Spencer is right. BOF, CLF whatever...we have to find=20
> some way to
> come forward with the e2e correlation.). Also, our network=20
> guys consider
> session-id to be useful, but I would also like to satisfy=20
> some OMA/3GPP
> requirements that it does not cover. =20
>=20
> Best Regards,
> Peter
>=20
>=20
>=20
>=20
>=20
>=20
> Message: 3
>=20
> Date: Fri, 5 Feb 2010 17:21:39 +0100
>=20
> From: <L.Liess@telekom.de>
>=20
> Subject: Re: [dispatch]
>=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>,
>=20
> <spencer@wonderhamster.org>
>=20
> Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,
>=20
> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de
>=20
> Message-ID:
>=20
> <40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>
>=20
> Content-Type: text/plain; charset=3D"iso-8859-1"
>=20
> Hi Peter,
>=20
> My understanding is that session-id and debug-id cover=20
> different sets of
> requirements.=20
>=20
> Session-id is in every message. If people want to find out=20
> what happened
> two days before, they can look at the old logs and the session-id is
> there in every message to help them to correlate e2e. For the debug-id
> to be there you have to know in advance that and which e2e traffic you
> would like to have correlated. The debug-id is activated=20
> "on-demand" and
> has no performance impact when it is not active, session-id is active
> all the time. The advantage of the debug-id is that there is no
> performance impact when debugging is not needed. Is my understanding
> correct?=20
>=20
> I like the debug-id stuff, it's really smart and we looked at it. But
> our troubleshooting and security guys want the session-id, for the
> reason above. I assume every service provider has its own needs and
> requirements.=20
>=20
> Spencer is right. BOF, CLF whatever...we have to find some way to come
> forward with the e2e correlation.=20
>=20
> Thanks a lot
>=20
> Laura
>=20
> =20
>=20
> =20
>=20
> > -----Original Message-----
>=20
> > From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
> <mailto:Peter.Dawes@vodafone.com> ]
>=20
> > Sent: Friday, February 05, 2010 4:00 PM
>=20
> > To: dispatch@ietf.org
>=20
> > Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;=20
>=20
> > HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>=20
> > Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > Hello,
>=20
> > Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03
>=20
> > +0200) as
>=20
> > "1) We have Hadriel's draft, 2) We also have the following draft,=20
>=20
> > which eventually led to the disaggregated media work:
>=20
> > http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>=20
>=20
> > -correlati
>=20
> > on-01.txt 3) We also have the SIP-XMPP work, which may also need to=20
>=20
> > correlate sessions somehow."
>=20
> >=20
>=20
> > I would like to add
>=20
> > http://tools.ietf.org/html/draft-dawes-sipping-debug-02
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02>  to this=20
>=20
> > discussion of use cases for some form of correlation=20
> identifier. This=20
>=20
> > draft relates to debugging/troubleshooting to meet documented=20
>=20
> > requirements from 3GPP and OMA, and proposes a kind of correlation=20
>=20
> > identifier in a new header.
>=20
> >=20
>=20
> > Regards,
>=20
> > Peter
>=20
> >=20
>=20
> >=20
>=20
> > Message: 2
>=20
> > Date: Fri, 22 Jan 2010 11:05:57 -0500
>=20
> > From: Paul Kyzivat <pkyzivat@cisco.com>
>=20
> > Subject: Re: [dispatch] FW:
>=20
> > I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> > To: L.Liess@telekom.de
>=20
> > Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
>=20
> > HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,
>=20
> > Gerold.Pinker@telekom.de
>=20
> > Message-ID: <4B59CCE5.2010202@cisco.com>
>=20
> > Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> > L.Liess@telekom.de wrote:
>=20
> > > Paul,
>=20
> > >=20
>=20
> > > Thank you for the examples. The scenarios can be really complex. I
>=20
> > hope we can get these correlation identifiers, whatever=20
> their names,=20
>=20
> > to work correctly in all cases....
>=20
> > >=20
>=20
> > > I am not sure I understood your message correctly. Do you say we
>=20
> > should have two different identifiers because we use them for two=20
>=20
> > different things?
>=20
> >=20
>=20
> > I think there may be more than two distinct usages that=20
> will require=20
>=20
> > different ones.
>=20
> >=20
>=20
> > IMO use cases should be gathered and then bucketed into=20
> sets that can=20
>=20
> > be
>=20
> >=20
>=20
> > satisfied by a single mechanism, without assuming initially=20
> how many=20
>=20
> > buckets there will be.
>=20
> >=20
>=20
> > (But I'm still on record as not being convinced that a=20
> correlation id=20
>=20
> > is
>=20
> >=20
>=20
> > needed *at all* for the case when multiple sip UAs are=20
> participating=20
>=20
> > on behalf of a single user in a call.)
>=20
> >=20
>=20
> > > Thanks a lot
>=20
> > > Laura
>=20
> > >=20
>=20
> >=20
>=20
> =20
> =20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From L.Liess@telekom.de  Tue Feb  9 07:50:02 2010
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From: <L.Liess@telekom.de>
To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Peter,=20

One more question.=20

>For example, I
> imagine that the debug-id draft could be re-written to use session-id
> (if session-id is chosen as the single "correlation id"=20
> header) to hold
> the "correlation id" with an added header field parameter to indicate
> the extra functionality for reducing the network burden of
> troubleshooting.=20

I is not realy clear to me how you intend to use the new parameter =
field. Could you give me more details?

Thanks a lot,
Laura

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Dawes, Peter, VF-Group
> Sent: Monday, February 08, 2010 4:58 PM
> To: dispatch@ietf.org
> Cc: Dawes, Peter, VF-Group; gonzalo.camarillo@ericsson.com;=20
> HKaplan@acmepacket.com; H=FClsemann, Martin; Pinker, Gerold
> Subject: Re: [dispatch]=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> Hi Laura,
> What you describe below about debug-id is the correct understanding. I
> am not convinced that using session-id for troubleshooting is=20
> practical
> because it implies that everything has to be logged everywhere in the
> network all the time so that you can, for example, look at any
> signalling that happened two days ago. The debug-id is inserted as
> required, and also policed at a couple of entities (first proxy,
> registrar) to take the burden of deciding whether or not to log away
> from the rest of the network. The debug-id is also based on=20
> 3GPP and OMA
> requirements, as I mentioned, but these requirements are, I believe,
> necessary to minimize burden on the network.=20
>=20
> Starting to read through the threads that consider the idea from Paul
> Kyzivat:
>=20
> > IMO use cases should be gathered and then bucketed into=20
> sets that can=20
>=20
> > be
>=20
> >=20
>=20
> > satisfied by a single mechanism, without assuming initially=20
> how many=20
>=20
> > buckets there will be.
>=20
> the beginning of a list of use cases seems to be 3rd-party CC,
> correlating multiple parallel dialogs, forked requests, requests that
> pass through B2BUAs. I guess that if an unambiguous test can be
> expressed for whether a particular session belongs to a set of related
> sessions, then one header to hold the id that correlates them=20
> is enough,
> even if populating it is difficult for some use cases. For example, I
> imagine that the debug-id draft could be re-written to use session-id
> (if session-id is chosen as the single "correlation id"=20
> header) to hold
> the "correlation id" with an added header field parameter to indicate
> the extra functionality for reducing the network burden of
> troubleshooting. This would satisfy the troubleshooting and 3GPP/OMA
> requirements without limiting the meaning of the "correlation id" to
> troubleshooting.=20
>=20
> Like you, I would like to find some way forward with the e2e=20
> correlation
> topic (Spencer is right. BOF, CLF whatever...we have to find=20
> some way to
> come forward with the e2e correlation.). Also, our network=20
> guys consider
> session-id to be useful, but I would also like to satisfy=20
> some OMA/3GPP
> requirements that it does not cover. =20
>=20
> Best Regards,
> Peter
>=20
>=20
>=20
>=20
>=20
>=20
> Message: 3
>=20
> Date: Fri, 5 Feb 2010 17:21:39 +0100
>=20
> From: <L.Liess@telekom.de>
>=20
> Subject: Re: [dispatch]
>=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>,
>=20
> <spencer@wonderhamster.org>
>=20
> Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,
>=20
> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de
>=20
> Message-ID:
>=20
> <40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>
>=20
> Content-Type: text/plain; charset=3D"iso-8859-1"
>=20
> Hi Peter,
>=20
> My understanding is that session-id and debug-id cover=20
> different sets of
> requirements.=20
>=20
> Session-id is in every message. If people want to find out=20
> what happened
> two days before, they can look at the old logs and the session-id is
> there in every message to help them to correlate e2e. For the debug-id
> to be there you have to know in advance that and which e2e traffic you
> would like to have correlated. The debug-id is activated=20
> "on-demand" and
> has no performance impact when it is not active, session-id is active
> all the time. The advantage of the debug-id is that there is no
> performance impact when debugging is not needed. Is my understanding
> correct?=20
>=20
> I like the debug-id stuff, it's really smart and we looked at it. But
> our troubleshooting and security guys want the session-id, for the
> reason above. I assume every service provider has its own needs and
> requirements.=20
>=20
> Spencer is right. BOF, CLF whatever...we have to find some way to come
> forward with the e2e correlation.=20
>=20
> Thanks a lot
>=20
> Laura
>=20
> =20
>=20
> =20
>=20
> > -----Original Message-----
>=20
> > From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
> <mailto:Peter.Dawes@vodafone.com> ]
>=20
> > Sent: Friday, February 05, 2010 4:00 PM
>=20
> > To: dispatch@ietf.org
>=20
> > Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;=20
>=20
> > HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>=20
> > Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > Hello,
>=20
> > Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03
>=20
> > +0200) as
>=20
> > "1) We have Hadriel's draft, 2) We also have the following draft,=20
>=20
> > which eventually led to the disaggregated media work:
>=20
> > http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>=20
>=20
> > -correlati
>=20
> > on-01.txt 3) We also have the SIP-XMPP work, which may also need to=20
>=20
> > correlate sessions somehow."
>=20
> >=20
>=20
> > I would like to add
>=20
> > http://tools.ietf.org/html/draft-dawes-sipping-debug-02
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02>  to this=20
>=20
> > discussion of use cases for some form of correlation=20
> identifier. This=20
>=20
> > draft relates to debugging/troubleshooting to meet documented=20
>=20
> > requirements from 3GPP and OMA, and proposes a kind of correlation=20
>=20
> > identifier in a new header.
>=20
> >=20
>=20
> > Regards,
>=20
> > Peter
>=20
> >=20
>=20
> >=20
>=20
> > Message: 2
>=20
> > Date: Fri, 22 Jan 2010 11:05:57 -0500
>=20
> > From: Paul Kyzivat <pkyzivat@cisco.com>
>=20
> > Subject: Re: [dispatch] FW:
>=20
> > I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> > To: L.Liess@telekom.de
>=20
> > Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
>=20
> > HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,
>=20
> > Gerold.Pinker@telekom.de
>=20
> > Message-ID: <4B59CCE5.2010202@cisco.com>
>=20
> > Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> > L.Liess@telekom.de wrote:
>=20
> > > Paul,
>=20
> > >=20
>=20
> > > Thank you for the examples. The scenarios can be really complex. I
>=20
> > hope we can get these correlation identifiers, whatever=20
> their names,=20
>=20
> > to work correctly in all cases....
>=20
> > >=20
>=20
> > > I am not sure I understood your message correctly. Do you say we
>=20
> > should have two different identifiers because we use them for two=20
>=20
> > different things?
>=20
> >=20
>=20
> > I think there may be more than two distinct usages that=20
> will require=20
>=20
> > different ones.
>=20
> >=20
>=20
> > IMO use cases should be gathered and then bucketed into=20
> sets that can=20
>=20
> > be
>=20
> >=20
>=20
> > satisfied by a single mechanism, without assuming initially=20
> how many=20
>=20
> > buckets there will be.
>=20
> >=20
>=20
> > (But I'm still on record as not being convinced that a=20
> correlation id=20
>=20
> > is
>=20
> >=20
>=20
> > needed *at all* for the case when multiple sip UAs are=20
> participating=20
>=20
> > on behalf of a single user in a call.)
>=20
> >=20
>=20
> > > Thanks a lot
>=20
> > > Laura
>=20
> > >=20
>=20
> >=20
>=20
> =20
> =20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From pkyzivat@cisco.com  Tue Feb  9 07:56:06 2010
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Cc: Mary Barnes <mbarnes@nortelnetworks.com>, DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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I don't have a problem with it being IMS-specific.
I just want it to *say* that it is IMS-specific.

	Thanks,
	Paul

DRAGE, Keith (Keith) wrote:
> At the moment, I think those of us looking at this from the 3GPP way of doing things do not see a more general application of this.
> 
> I do not have a problem with doing it more generally, but I guess the people who see a clear use case for that need to speak up and identify their use cases. 
> 
> Otherwise we end up designing a general event package that no one else ever uses. As opposed to the alternative of clearly stating the restricted use to a particular applicability, approving it, and moving on to something else with our restricted amount of resources.
> 
> regards
> 
> Keith 
> 
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org 
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>> Sent: Tuesday, February 09, 2010 9:16 AM
>> To: DISPATCH
>> Cc: Mary Barnes
>> Subject: [dispatch] Prorgessing 
>> draft-avasarala-dispatch-comm-div-notification-02
>>
>> Hi,
>>
>> there have been a set of messages on the list providing 
>> comments on the draft below:
>>
>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>> otification-02
>>
>> As you know, the process for defining SIP event packages is 
>> documented here:
>>
>> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
>> ction-4.1
>>
>> Based on the feedback received, the authors need to decide 
>> whether they want to generalize their solution so that is is 
>> generally applicable to the public Internet or if they want 
>> to (further) clarify that this mechanism is intended to work 
>> only in IMS networks that provide a CDIV service.
>>
>> If the authors choose the latter approach, we (the DISPATCH 
>> chairs) will choose a "Designated Expert" who will check the 
>> draft against the 7 points described in the document above.
>>
>> Cheers,
>>
>> Gonzalo
>> DISPATCH co-chair
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From Peter.Dawes@vodafone.com  Tue Feb  9 08:14:52 2010
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Laura,
As I understand it, the session-id header field is included in all
requests. I think it would be possible to define a header field
parameter for session-id, say "; debug", that is added to session-id by
a UA iff certain conditions are met. This is similar to including the
debug-id header field iff pre-supplied trigger conditions are matched in
the UA. The first proxy can then police sesssion-id header fields
containing this parameter (and let others go through unchecked), this
parameter can trigger logging in downsteam SIP entities and so on to
provide all functions of the debug draft. It might even be possible to
use Proxy-Require based on this new parameter to make sure that the
logging happens.=20
I am not sure that I want to re-write my draft this way, but it could be
a way to separate correlation-id in general from a specific use case
(troubleshooting).

Best Regards,
Peter





Message: 5

Date: Tue, 9 Feb 2010 16:51:03 +0100

From: <L.Liess@telekom.de>

Subject: Re: [dispatch]

I-DAction:draft-kaplan-dispatch-session-id-00.txt

To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>

Cc: gonzalo.camarillo@ericsson.com, HKaplan@acmepacket.com,

Martin.Huelsemann@telekom.de

Message-ID:

<40FB0FFB97588246A1BEFB05759DC8A003EFE9AB@S4DE9JSAANI.ost.t-com.de>

Content-Type: text/plain; charset=3D"iso-8859-1"

Hi Peter,=20

One more question.=20

>For example, I

> imagine that the debug-id draft could be re-written to use session-id=20

>(if session-id is chosen as the single "correlation id"

> header) to hold

> the "correlation id" with an added header field parameter to indicate=20

>the extra functionality for reducing the network burden of=20

>troubleshooting.

I is not realy clear to me how you intend to use the new parameter
field. Could you give me more details?

Thanks a lot,

Laura

> -----Original Message-----

> From: dispatch-bounces@ietf.org

> [mailto:dispatch-bounces@ietf.org <mailto:dispatch-bounces@ietf.org> ]
On Behalf Of Dawes, Peter, VF-Group

> Sent: Monday, February 08, 2010 4:58 PM

> To: dispatch@ietf.org

> Cc: Dawes, Peter, VF-Group; gonzalo.camarillo@ericsson.com;=20

> HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold

> Subject: Re: [dispatch]

> I-DAction:draft-kaplan-dispatch-session-id-00.txt

>=20

> Hi Laura,

> What you describe below about debug-id is the correct understanding. I


> am not convinced that using session-id for troubleshooting is=20

> practical because it implies that everything has to be logged=20

> everywhere in the network all the time so that you can, for example,=20

> look at any signalling that happened two days ago. The debug-id is=20

> inserted as required, and also policed at a couple of entities (first=20

> proxy,

> registrar) to take the burden of deciding whether or not to log away=20

> from the rest of the network. The debug-id is also based on 3GPP and=20

> OMA requirements, as I mentioned, but these requirements are, I=20

> believe, necessary to minimize burden on the network.

>=20

> Starting to read through the threads that consider the idea from Paul

> Kyzivat:

>=20

> > IMO use cases should be gathered and then bucketed into

> sets that can

>=20

> > be

>=20

> >=20

>=20

> > satisfied by a single mechanism, without assuming initially

> how many

>=20

> > buckets there will be.

>=20

> the beginning of a list of use cases seems to be 3rd-party CC,=20

> correlating multiple parallel dialogs, forked requests, requests that=20

> pass through B2BUAs. I guess that if an unambiguous test can be=20

> expressed for whether a particular session belongs to a set of related


> sessions, then one header to hold the id that correlates them is=20

> enough, even if populating it is difficult for some use cases. For=20

> example, I imagine that the debug-id draft could be re-written to use=20

> session-id (if session-id is chosen as the single "correlation id"

> header) to hold

> the "correlation id" with an added header field parameter to indicate=20

> the extra functionality for reducing the network burden of=20

> troubleshooting. This would satisfy the troubleshooting and 3GPP/OMA=20

> requirements without limiting the meaning of the "correlation id" to=20

> troubleshooting.

>=20

> Like you, I would like to find some way forward with the e2e=20

> correlation topic (Spencer is right. BOF, CLF whatever...we have to=20

> find some way to come forward with the e2e correlation.). Also, our=20

> network guys consider session-id to be useful, but I would also like=20

> to satisfy some OMA/3GPP requirements that it does not cover.

>=20

> Best Regards,

> Peter

>=20

>=20

>=20

>=20

>=20

>=20

> Message: 3

>=20

> Date: Fri, 5 Feb 2010 17:21:39 +0100

>=20

> From: <L.Liess@telekom.de>

>=20

> Subject: Re: [dispatch]

>=20

> I-DAction:draft-kaplan-dispatch-session-id-00.txt

>=20

> To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>,

>=20

> <spencer@wonderhamster.org>

>=20

> Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,

>=20

> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de

>=20

> Message-ID:

>=20

> <40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>

>=20

> Content-Type: text/plain; charset=3D"iso-8859-1"

>=20

> Hi Peter,

>=20

> My understanding is that session-id and debug-id cover different sets=20

> of requirements.

>=20

> Session-id is in every message. If people want to find out what=20

> happened two days before, they can look at the old logs and the=20

> session-id is there in every message to help them to correlate e2e.=20

> For the debug-id to be there you have to know in advance that and=20

> which e2e traffic you would like to have correlated. The debug-id is=20

> activated "on-demand" and has no performance impact when it is not=20

> active, session-id is active all the time. The advantage of the=20

> debug-id is that there is no performance impact when debugging is not=20

> needed. Is my understanding correct?

>=20

> I like the debug-id stuff, it's really smart and we looked at it. But=20

> our troubleshooting and security guys want the session-id, for the=20

> reason above. I assume every service provider has its own needs and=20

> requirements.

>=20

> Spencer is right. BOF, CLF whatever...we have to find some way to come


> forward with the e2e correlation.

>=20

> Thanks a lot

>=20

> Laura

>=20

>=20

>=20

>=20

>=20

> > -----Original Message-----

>=20

> > From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
<mailto:Peter.Dawes@vodafone.com>=20

> <mailto:Peter.Dawes@vodafone.com <mailto:Peter.Dawes@vodafone.com> > ]

>=20

> > Sent: Friday, February 05, 2010 4:00 PM

>=20

> > To: dispatch@ietf.org

>=20

> > Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;

>=20

> > HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold

>=20

> > Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt

>=20

> >=20

>=20

> > Hello,

>=20

> > Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03

>=20

> > +0200) as

>=20

> > "1) We have Hadriel's draft, 2) We also have the following draft,

>=20

> > which eventually led to the disaggregated media work:

>=20

> > http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
<http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>=20

> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
<http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog> >

>=20

> > -correlati

>=20

> > on-01.txt 3) We also have the SIP-XMPP work, which may also need to

>=20

> > correlate sessions somehow."

>=20

> >=20

>=20

> > I would like to add

>=20

> > http://tools.ietf.org/html/draft-dawes-sipping-debug-02
<http://tools.ietf.org/html/draft-dawes-sipping-debug-02>=20

> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02
<http://tools.ietf.org/html/draft-dawes-sipping-debug-02> > to this

>=20

> > discussion of use cases for some form of correlation

> identifier. This

>=20

> > draft relates to debugging/troubleshooting to meet documented

>=20

> > requirements from 3GPP and OMA, and proposes a kind of correlation

>=20

> > identifier in a new header.

>=20

> >=20

>=20

> > Regards,

>=20

> > Peter

>=20

> >=20

>=20

> >=20

>=20

> > Message: 2

>=20

> > Date: Fri, 22 Jan 2010 11:05:57 -0500

>=20

> > From: Paul Kyzivat <pkyzivat@cisco.com>

>=20

> > Subject: Re: [dispatch] FW:

>=20

> > I-DAction:draft-kaplan-dispatch-session-id-00.txt

>=20

> > To: L.Liess@telekom.de

>=20

> > Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,

>=20

> > HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,

>=20

> > Gerold.Pinker@telekom.de

>=20

> > Message-ID: <4B59CCE5.2010202@cisco.com>

>=20

> > Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed

>=20

> >=20

>=20

> >=20

>=20

> >=20

>=20

> > L.Liess@telekom.de wrote:

>=20

> > > Paul,

>=20

> > >=20

>=20

> > > Thank you for the examples. The scenarios can be really complex. I

>=20

> > hope we can get these correlation identifiers, whatever

> their names,

>=20

> > to work correctly in all cases....

>=20

> > >=20

>=20

> > > I am not sure I understood your message correctly. Do you say we

>=20

> > should have two different identifiers because we use them for two

>=20

> > different things?

>=20

> >=20

>=20

> > I think there may be more than two distinct usages that

> will require

>=20

> > different ones.

>=20

> >=20

>=20

> > IMO use cases should be gathered and then bucketed into

> sets that can

>=20

> > be

>=20

> >=20

>=20

> > satisfied by a single mechanism, without assuming initially

> how many

>=20

> > buckets there will be.

>=20

> >=20

>=20

> > (But I'm still on record as not being convinced that a

> correlation id

>=20

> > is

>=20

> >=20

>=20

> > needed *at all* for the case when multiple sip UAs are

> participating

>=20

> > on behalf of a single user in a call.)

>=20

> >=20

>=20

> > > Thanks a lot

>=20

> > > Laura

>=20

> > >=20

>=20

> >=20

>=20

>=20

>=20

> _______________________________________________

> dispatch mailing list

> dispatch@ietf.org

> https://www.ietf.org/mailman/listinfo/dispatch
<https://www.ietf.org/mailman/listinfo/dispatch>=20

>=20

=20

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

_______________________________________________

dispatch mailing list

dispatch@ietf.org

https://www.ietf.org/mailman/listinfo/dispatch
<https://www.ietf.org/mailman/listinfo/dispatch>=20

=20

End of dispatch Digest, Vol 11, Issue 21

****************************************

=20
=20

From drage@alcatel-lucent.com  Tue Feb  9 08:53:32 2010
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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 9 Feb 2010 17:53:57 +0100
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Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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I think that is a given and if that is the way to go, I'm sure we'll all ma=
ke sure the words on the front cover go that way.

But there seemed to be some prior responses trying to take potential use ca=
ses beyond that, and I wanted to make sure that if they existed, they were =
explicitly declared.


regards

Keith=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Tuesday, February 09, 2010 3:57 PM
> To: DRAGE, Keith (Keith)
> Cc: Gonzalo Camarillo; DISPATCH; Mary Barnes
> Subject: Re: [dispatch] Prorgessing=20
> draft-avasarala-dispatch-comm-div-notification-02
>=20
> I don't have a problem with it being IMS-specific.
> I just want it to *say* that it is IMS-specific.
>=20
> 	Thanks,
> 	Paul
>=20
> DRAGE, Keith (Keith) wrote:
> > At the moment, I think those of us looking at this from the=20
> 3GPP way of doing things do not see a more general=20
> application of this.
> >=20
> > I do not have a problem with doing it more generally, but I=20
> guess the people who see a clear use case for that need to=20
> speak up and identify their use cases.=20
> >=20
> > Otherwise we end up designing a general event package that=20
> no one else ever uses. As opposed to the alternative of=20
> clearly stating the restricted use to a particular=20
> applicability, approving it, and moving on to something else=20
> with our restricted amount of resources.
> >=20
> > regards
> >=20
> > Keith
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> >> Sent: Tuesday, February 09, 2010 9:16 AM
> >> To: DISPATCH
> >> Cc: Mary Barnes
> >> Subject: [dispatch] Prorgessing
> >> draft-avasarala-dispatch-comm-div-notification-02
> >>
> >> Hi,
> >>
> >> there have been a set of messages on the list providing=20
> comments on=20
> >> the draft below:
> >>
> >> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
> >> otification-02
> >>
> >> As you know, the process for defining SIP event packages is=20
> >> documented here:
> >>
> >> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
> >> ction-4.1
> >>
> >> Based on the feedback received, the authors need to decide whether=20
> >> they want to generalize their solution so that is is generally=20
> >> applicable to the public Internet or if they want to (further)=20
> >> clarify that this mechanism is intended to work only in=20
> IMS networks=20
> >> that provide a CDIV service.
> >>
> >> If the authors choose the latter approach, we (the DISPATCH
> >> chairs) will choose a "Designated Expert" who will check the draft=20
> >> against the 7 points described in the document above.
> >>
> >> Cheers,
> >>
> >> Gonzalo
> >> DISPATCH co-chair
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> =

From ranjit@motorola.com  Tue Feb  9 09:01:05 2010
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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Hi all

Sure. I am updating the I-D with the comments and modifying the
applicability statement to reflect that it is specific to IMS. Will
submit the updated (-03) version of the I-D and then take it forward
with Option-2 as suggested by Camarillo.

Thanks=20


Regards
Ranjit

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of DRAGE, Keith (Keith)
Sent: Tuesday, February 09, 2010 10:24 PM
To: Paul Kyzivat
Cc: Mary Barnes; DISPATCH
Subject: Re: [dispatch]Prorgessing
draft-avasarala-dispatch-comm-div-notification-02

I think that is a given and if that is the way to go, I'm sure we'll all
make sure the words on the front cover go that way.

But there seemed to be some prior responses trying to take potential use
cases beyond that, and I wanted to make sure that if they existed, they
were explicitly declared.


regards

Keith=20

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, February 09, 2010 3:57 PM
> To: DRAGE, Keith (Keith)
> Cc: Gonzalo Camarillo; DISPATCH; Mary Barnes
> Subject: Re: [dispatch] Prorgessing
> draft-avasarala-dispatch-comm-div-notification-02
>=20
> I don't have a problem with it being IMS-specific.
> I just want it to *say* that it is IMS-specific.
>=20
> 	Thanks,
> 	Paul
>=20
> DRAGE, Keith (Keith) wrote:
> > At the moment, I think those of us looking at this from the
> 3GPP way of doing things do not see a more general application of=20
> this.
> >=20
> > I do not have a problem with doing it more generally, but I
> guess the people who see a clear use case for that need to speak up=20
> and identify their use cases.
> >=20
> > Otherwise we end up designing a general event package that
> no one else ever uses. As opposed to the alternative of clearly=20
> stating the restricted use to a particular applicability, approving=20
> it, and moving on to something else with our restricted amount of=20
> resources.
> >=20
> > regards
> >=20
> > Keith
> >=20
> >> -----Original Message-----
> >> From: dispatch-bounces@ietf.org
> >> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
> >> Sent: Tuesday, February 09, 2010 9:16 AM
> >> To: DISPATCH
> >> Cc: Mary Barnes
> >> Subject: [dispatch] Prorgessing
> >> draft-avasarala-dispatch-comm-div-notification-02
> >>
> >> Hi,
> >>
> >> there have been a set of messages on the list providing
> comments on
> >> the draft below:
> >>
> >> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
> >> otification-02
> >>
> >> As you know, the process for defining SIP event packages is=20
> >> documented here:
> >>
> >> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
> >> ction-4.1
> >>
> >> Based on the feedback received, the authors need to decide whether=20
> >> they want to generalize their solution so that is is generally=20
> >> applicable to the public Internet or if they want to (further)=20
> >> clarify that this mechanism is intended to work only in
> IMS networks
> >> that provide a CDIV service.
> >>
> >> If the authors choose the latter approach, we (the DISPATCH
> >> chairs) will choose a "Designated Expert" who will check the draft=20
> >> against the 7 points described in the document above.
> >>
> >> Cheers,
> >>
> >> Gonzalo
> >> DISPATCH co-chair
> >> _______________________________________________
> >> dispatch mailing list
> >> dispatch@ietf.org
> >> https://www.ietf.org/mailman/listinfo/dispatch
> >>
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From salvatore.loreto@ericsson.com  Tue Feb  9 09:26:50 2010
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Peter,

I think your proposal to use a header parameter like "debug"
is a reasonable solution.

However I would suggest to start from requirements and then arrive to 
the actual solution.

/Sal

On 02/09/2010 06:15 PM, Dawes, Peter, VF-Group wrote:
> Hi Laura,
> As I understand it, the session-id header field is included in all
> requests. I think it would be possible to define a header field
> parameter for session-id, say "; debug", that is added to session-id by
> a UA iff certain conditions are met. This is similar to including the
> debug-id header field iff pre-supplied trigger conditions are matched in
> the UA. The first proxy can then police sesssion-id header fields
> containing this parameter (and let others go through unchecked), this
> parameter can trigger logging in downsteam SIP entities and so on to
> provide all functions of the debug draft. It might even be possible to
> use Proxy-Require based on this new parameter to make sure that the
> logging happens.
> I am not sure that I want to re-write my draft this way, but it could be
> a way to separate correlation-id in general from a specific use case
> (troubleshooting).
>
> Best Regards,
> Peter
>
>
>
>
>
> Message: 5
>
> Date: Tue, 9 Feb 2010 16:51:03 +0100
>
> From:<L.Liess@telekom.de>
>
> Subject: Re: [dispatch]
>
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>
> To:<Peter.Dawes@vodafone.com>,<dispatch@ietf.org>
>
> Cc: gonzalo.camarillo@ericsson.com, HKaplan@acmepacket.com,
>
> Martin.Huelsemann@telekom.de
>
> Message-ID:
>
> <40FB0FFB97588246A1BEFB05759DC8A003EFE9AB@S4DE9JSAANI.ost.t-com.de>
>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Peter,
>
> One more question.
>
>    
>> For example, I
>>      
>    
>> imagine that the debug-id draft could be re-written to use session-id
>>      
>    
>> (if session-id is chosen as the single "correlation id"
>>      
>    
>> header) to hold
>>      
>    
>> the "correlation id" with an added header field parameter to indicate
>>      
>    
>> the extra functionality for reducing the network burden of
>>      
>    
>> troubleshooting.
>>      
> I is not realy clear to me how you intend to use the new parameter
> field. Could you give me more details?
>
> Thanks a lot,
>
> Laura
>
>    
>> -----Original Message-----
>>      
>    
>> From: dispatch-bounces@ietf.org
>>      
>    
>> [mailto:dispatch-bounces@ietf.org<mailto:dispatch-bounces@ietf.org>  ]
>>      
> On Behalf Of Dawes, Peter, VF-Group
>
>    
>> Sent: Monday, February 08, 2010 4:58 PM
>>      
>    
>> To: dispatch@ietf.org
>>      
>    
>> Cc: Dawes, Peter, VF-Group; gonzalo.camarillo@ericsson.com;
>>      
>    
>> HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>>      
>    
>> Subject: Re: [dispatch]
>>      
>    
>> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>>      
>    
>>      
>    
>> Hi Laura,
>>      
>    
>> What you describe below about debug-id is the correct understanding. I
>>      
>
>    
>> am not convinced that using session-id for troubleshooting is
>>      
>    
>> practical because it implies that everything has to be logged
>>      
>    
>> everywhere in the network all the time so that you can, for example,
>>      
>    
>> look at any signalling that happened two days ago. The debug-id is
>>      
>    
>> inserted as required, and also policed at a couple of entities (first
>>      
>    
>> proxy,
>>      
>    
>> registrar) to take the burden of deciding whether or not to log away
>>      
>    
>> from the rest of the network. The debug-id is also based on 3GPP and
>>      
>    
>> OMA requirements, as I mentioned, but these requirements are, I
>>      
>    
>> believe, necessary to minimize burden on the network.
>>      
>    
>>      
>    
>> Starting to read through the threads that consider the idea from Paul
>>      
>    
>> Kyzivat:
>>      
>    
>>      
>    
>>> IMO use cases should be gathered and then bucketed into
>>>        
>    
>> sets that can
>>      
>    
>>      
>    
>>> be
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> satisfied by a single mechanism, without assuming initially
>>>        
>    
>> how many
>>      
>    
>>      
>    
>>> buckets there will be.
>>>        
>    
>>      
>    
>> the beginning of a list of use cases seems to be 3rd-party CC,
>>      
>    
>> correlating multiple parallel dialogs, forked requests, requests that
>>      
>    
>> pass through B2BUAs. I guess that if an unambiguous test can be
>>      
>    
>> expressed for whether a particular session belongs to a set of related
>>      
>
>    
>> sessions, then one header to hold the id that correlates them is
>>      
>    
>> enough, even if populating it is difficult for some use cases. For
>>      
>    
>> example, I imagine that the debug-id draft could be re-written to use
>>      
>    
>> session-id (if session-id is chosen as the single "correlation id"
>>      
>    
>> header) to hold
>>      
>    
>> the "correlation id" with an added header field parameter to indicate
>>      
>    
>> the extra functionality for reducing the network burden of
>>      
>    
>> troubleshooting. This would satisfy the troubleshooting and 3GPP/OMA
>>      
>    
>> requirements without limiting the meaning of the "correlation id" to
>>      
>    
>> troubleshooting.
>>      
>    
>>      
>    
>> Like you, I would like to find some way forward with the e2e
>>      
>    
>> correlation topic (Spencer is right. BOF, CLF whatever...we have to
>>      
>    
>> find some way to come forward with the e2e correlation.). Also, our
>>      
>    
>> network guys consider session-id to be useful, but I would also like
>>      
>    
>> to satisfy some OMA/3GPP requirements that it does not cover.
>>      
>    
>>      
>    
>> Best Regards,
>>      
>    
>> Peter
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>> Message: 3
>>      
>    
>>      
>    
>> Date: Fri, 5 Feb 2010 17:21:39 +0100
>>      
>    
>>      
>    
>> From:<L.Liess@telekom.de>
>>      
>    
>>      
>    
>> Subject: Re: [dispatch]
>>      
>    
>>      
>    
>> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>>      
>    
>>      
>    
>> To:<Peter.Dawes@vodafone.com>,<dispatch@ietf.org>,
>>      
>    
>>      
>    
>> <spencer@wonderhamster.org>
>>      
>    
>>      
>    
>> Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,
>>      
>    
>>      
>    
>> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de
>>      
>    
>>      
>    
>> Message-ID:
>>      
>    
>>      
>    
>> <40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>
>>      
>    
>>      
>    
>> Content-Type: text/plain; charset="iso-8859-1"
>>      
>    
>>      
>    
>> Hi Peter,
>>      
>    
>>      
>    
>> My understanding is that session-id and debug-id cover different sets
>>      
>    
>> of requirements.
>>      
>    
>>      
>    
>> Session-id is in every message. If people want to find out what
>>      
>    
>> happened two days before, they can look at the old logs and the
>>      
>    
>> session-id is there in every message to help them to correlate e2e.
>>      
>    
>> For the debug-id to be there you have to know in advance that and
>>      
>    
>> which e2e traffic you would like to have correlated. The debug-id is
>>      
>    
>> activated "on-demand" and has no performance impact when it is not
>>      
>    
>> active, session-id is active all the time. The advantage of the
>>      
>    
>> debug-id is that there is no performance impact when debugging is not
>>      
>    
>> needed. Is my understanding correct?
>>      
>    
>>      
>    
>> I like the debug-id stuff, it's really smart and we looked at it. But
>>      
>    
>> our troubleshooting and security guys want the session-id, for the
>>      
>    
>> reason above. I assume every service provider has its own needs and
>>      
>    
>> requirements.
>>      
>    
>>      
>    
>> Spencer is right. BOF, CLF whatever...we have to find some way to come
>>      
>
>    
>> forward with the e2e correlation.
>>      
>    
>>      
>    
>> Thanks a lot
>>      
>    
>>      
>    
>> Laura
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>      
>    
>>> -----Original Message-----
>>>        
>    
>>      
>    
>>> From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
>>>        
> <mailto:Peter.Dawes@vodafone.com>
>
>    
>> <mailto:Peter.Dawes@vodafone.com<mailto:Peter.Dawes@vodafone.com>  >  ]
>>      
>    
>>      
>    
>>> Sent: Friday, February 05, 2010 4:00 PM
>>>        
>    
>>      
>    
>>> To: dispatch@ietf.org
>>>        
>    
>>      
>    
>>> Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;
>>>        
>    
>>      
>    
>>> HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>>>        
>    
>>      
>    
>>> Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> Hello,
>>>        
>    
>>      
>    
>>> Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03
>>>        
>    
>>      
>    
>>> +0200) as
>>>        
>    
>>      
>    
>>> "1) We have Hadriel's draft, 2) We also have the following draft,
>>>        
>    
>>      
>    
>>> which eventually led to the disaggregated media work:
>>>        
>    
>>      
>    
>>> http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
>>>        
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>
>
>    
>> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
>>      
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>  >
>
>    
>>      
>    
>>> -correlati
>>>        
>    
>>      
>    
>>> on-01.txt 3) We also have the SIP-XMPP work, which may also need to
>>>        
>    
>>      
>    
>>> correlate sessions somehow."
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> I would like to add
>>>        
>    
>>      
>    
>>> http://tools.ietf.org/html/draft-dawes-sipping-debug-02
>>>        
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02>
>
>    
>> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02
>>      
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02>  >  to this
>
>    
>>      
>    
>>> discussion of use cases for some form of correlation
>>>        
>    
>> identifier. This
>>      
>    
>>      
>    
>>> draft relates to debugging/troubleshooting to meet documented
>>>        
>    
>>      
>    
>>> requirements from 3GPP and OMA, and proposes a kind of correlation
>>>        
>    
>>      
>    
>>> identifier in a new header.
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> Regards,
>>>        
>    
>>      
>    
>>> Peter
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> Message: 2
>>>        
>    
>>      
>    
>>> Date: Fri, 22 Jan 2010 11:05:57 -0500
>>>        
>    
>>      
>    
>>> From: Paul Kyzivat<pkyzivat@cisco.com>
>>>        
>    
>>      
>    
>>> Subject: Re: [dispatch] FW:
>>>        
>    
>>      
>    
>>> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>>>        
>    
>>      
>    
>>> To: L.Liess@telekom.de
>>>        
>    
>>      
>    
>>> Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
>>>        
>    
>>      
>    
>>> HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,
>>>        
>    
>>      
>    
>>> Gerold.Pinker@telekom.de
>>>        
>    
>>      
>    
>>> Message-ID:<4B59CCE5.2010202@cisco.com>
>>>        
>    
>>      
>    
>>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> L.Liess@telekom.de wrote:
>>>        
>    
>>      
>    
>>>> Paul,
>>>>          
>    
>>      
>    
>>>>          
>    
>>      
>    
>>>> Thank you for the examples. The scenarios can be really complex. I
>>>>          
>    
>>      
>    
>>> hope we can get these correlation identifiers, whatever
>>>        
>    
>> their names,
>>      
>    
>>      
>    
>>> to work correctly in all cases....
>>>        
>    
>>      
>    
>>>>          
>    
>>      
>    
>>>> I am not sure I understood your message correctly. Do you say we
>>>>          
>    
>>      
>    
>>> should have two different identifiers because we use them for two
>>>        
>    
>>      
>    
>>> different things?
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> I think there may be more than two distinct usages that
>>>        
>    
>> will require
>>      
>    
>>      
>    
>>> different ones.
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> IMO use cases should be gathered and then bucketed into
>>>        
>    
>> sets that can
>>      
>    
>>      
>    
>>> be
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> satisfied by a single mechanism, without assuming initially
>>>        
>    
>> how many
>>      
>    
>>      
>    
>>> buckets there will be.
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> (But I'm still on record as not being convinced that a
>>>        
>    
>> correlation id
>>      
>    
>>      
>    
>>> is
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>> needed *at all* for the case when multiple sip UAs are
>>>        
>    
>> participating
>>      
>    
>>      
>    
>>> on behalf of a single user in a call.)
>>>        
>    
>>      
>    
>>>        
>    
>>      
>    
>>>> Thanks a lot
>>>>          
>    
>>      
>    
>>>> Laura
>>>>          
>    
>>      
>    
>>>>          
>    
>>      
>    
>>>        
>    
>>      
>    
>>      
>    
>>      
>    
>> _______________________________________________
>>      
>    
>> dispatch mailing list
>>      
>    
>> dispatch@ietf.org
>>      
>    
>> https://www.ietf.org/mailman/listinfo/dispatch
>>      
> <https://www.ietf.org/mailman/listinfo/dispatch>
>
>    
>>      
>
>
> ------------------------------
>
> _______________________________________________
>
> dispatch mailing list
>
> dispatch@ietf.org
>
> https://www.ietf.org/mailman/listinfo/dispatch
> <https://www.ietf.org/mailman/listinfo/dispatch>
>
>
>
> End of dispatch Digest, Vol 11, Issue 21
>
> ****************************************
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


From ankriste@cisco.com  Tue Feb  9 12:03:06 2010
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From: Anders Kristensen <ankriste@cisco.com>
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Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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IIUC the IMS use case (or one of them) is that users may configure call 
forwarding services and forget that they've done so. The CDIV feature 
then allows them to learn of calls being forwarded and take corrective 
action. This is not at all IMS specific - the same use case can be said 
to exist in non-IMS deployments.

I suspect addressing this problem in a general way is no more work than 
to do it in an IMS specific way but would make for a better standard. 
What's more, if I'm right and this is inherently not an IMS specific 
problem then the solution is likely to be picked up and used in other 
contexts regardless of whatever stern wording is on the front page of 
the RFC.

Thanks,
Anders

On 2/9/2010 8:53 AM, DRAGE, Keith (Keith) wrote:
> I think that is a given and if that is the way to go, I'm sure we'll all make sure the words on the front cover go that way.
>
> But there seemed to be some prior responses trying to take potential use cases beyond that, and I wanted to make sure that if they existed, they were explicitly declared.
>
>
> regards
>
> Keith
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, February 09, 2010 3:57 PM
>> To: DRAGE, Keith (Keith)
>> Cc: Gonzalo Camarillo; DISPATCH; Mary Barnes
>> Subject: Re: [dispatch] Prorgessing
>> draft-avasarala-dispatch-comm-div-notification-02
>>
>> I don't have a problem with it being IMS-specific.
>> I just want it to *say* that it is IMS-specific.
>>
>> 	Thanks,
>> 	Paul
>>
>> DRAGE, Keith (Keith) wrote:
>>> At the moment, I think those of us looking at this from the
>> 3GPP way of doing things do not see a more general
>> application of this.
>>>
>>> I do not have a problem with doing it more generally, but I
>> guess the people who see a clear use case for that need to
>> speak up and identify their use cases.
>>>
>>> Otherwise we end up designing a general event package that
>> no one else ever uses. As opposed to the alternative of
>> clearly stating the restricted use to a particular
>> applicability, approving it, and moving on to something else
>> with our restricted amount of resources.
>>>
>>> regards
>>>
>>> Keith
>>>
>>>> -----Original Message-----
>>>> From: dispatch-bounces@ietf.org
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>>>> Sent: Tuesday, February 09, 2010 9:16 AM
>>>> To: DISPATCH
>>>> Cc: Mary Barnes
>>>> Subject: [dispatch] Prorgessing
>>>> draft-avasarala-dispatch-comm-div-notification-02
>>>>
>>>> Hi,
>>>>
>>>> there have been a set of messages on the list providing
>> comments on
>>>> the draft below:
>>>>
>>>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>>>> otification-02
>>>>
>>>> As you know, the process for defining SIP event packages is
>>>> documented here:
>>>>
>>>> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
>>>> ction-4.1
>>>>
>>>> Based on the feedback received, the authors need to decide whether
>>>> they want to generalize their solution so that is is generally
>>>> applicable to the public Internet or if they want to (further)
>>>> clarify that this mechanism is intended to work only in
>> IMS networks
>>>> that provide a CDIV service.
>>>>
>>>> If the authors choose the latter approach, we (the DISPATCH
>>>> chairs) will choose a "Designated Expert" who will check the draft
>>>> against the 7 points described in the document above.
>>>>
>>>> Cheers,
>>>>
>>>> Gonzalo
>>>> DISPATCH co-chair
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From pkyzivat@cisco.com  Tue Feb  9 12:50:13 2010
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Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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Anders,

It was also my assumption going in that this is potentially a problem 
area of general interest.

But the existing solution (as best I understand it) is wed to a 
particular system architecture - not one that we would likely wish to 
see enshrined in a more general standard.

To get beyond that we would need somebody who is interested in doing the 
work to get something more general. I'm not seeing anybody stand up to 
do that. (I know I don't have the time to do it.)

Lacking that, I see no justification to block their getting their event 
package as long as it is suitably scoped.

	Thanks,
	Paul

Anders Kristensen wrote:
> IIUC the IMS use case (or one of them) is that users may configure call 
> forwarding services and forget that they've done so. The CDIV feature 
> then allows them to learn of calls being forwarded and take corrective 
> action. This is not at all IMS specific - the same use case can be said 
> to exist in non-IMS deployments.
> 
> I suspect addressing this problem in a general way is no more work than 
> to do it in an IMS specific way but would make for a better standard. 
> What's more, if I'm right and this is inherently not an IMS specific 
> problem then the solution is likely to be picked up and used in other 
> contexts regardless of whatever stern wording is on the front page of 
> the RFC.
> 
> Thanks,
> Anders
> 
> On 2/9/2010 8:53 AM, DRAGE, Keith (Keith) wrote:
>> I think that is a given and if that is the way to go, I'm sure we'll 
>> all make sure the words on the front cover go that way.
>>
>> But there seemed to be some prior responses trying to take potential 
>> use cases beyond that, and I wanted to make sure that if they existed, 
>> they were explicitly declared.
>>
>>
>> regards
>>
>> Keith
>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>> Sent: Tuesday, February 09, 2010 3:57 PM
>>> To: DRAGE, Keith (Keith)
>>> Cc: Gonzalo Camarillo; DISPATCH; Mary Barnes
>>> Subject: Re: [dispatch] Prorgessing
>>> draft-avasarala-dispatch-comm-div-notification-02
>>>
>>> I don't have a problem with it being IMS-specific.
>>> I just want it to *say* that it is IMS-specific.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>> DRAGE, Keith (Keith) wrote:
>>>> At the moment, I think those of us looking at this from the
>>> 3GPP way of doing things do not see a more general
>>> application of this.
>>>>
>>>> I do not have a problem with doing it more generally, but I
>>> guess the people who see a clear use case for that need to
>>> speak up and identify their use cases.
>>>>
>>>> Otherwise we end up designing a general event package that
>>> no one else ever uses. As opposed to the alternative of
>>> clearly stating the restricted use to a particular
>>> applicability, approving it, and moving on to something else
>>> with our restricted amount of resources.
>>>>
>>>> regards
>>>>
>>>> Keith
>>>>
>>>>> -----Original Message-----
>>>>> From: dispatch-bounces@ietf.org
>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>>>>> Sent: Tuesday, February 09, 2010 9:16 AM
>>>>> To: DISPATCH
>>>>> Cc: Mary Barnes
>>>>> Subject: [dispatch] Prorgessing
>>>>> draft-avasarala-dispatch-comm-div-notification-02
>>>>>
>>>>> Hi,
>>>>>
>>>>> there have been a set of messages on the list providing
>>> comments on
>>>>> the draft below:
>>>>>
>>>>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>>>>> otification-02
>>>>>
>>>>> As you know, the process for defining SIP event packages is
>>>>> documented here:
>>>>>
>>>>> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
>>>>> ction-4.1
>>>>>
>>>>> Based on the feedback received, the authors need to decide whether
>>>>> they want to generalize their solution so that is is generally
>>>>> applicable to the public Internet or if they want to (further)
>>>>> clarify that this mechanism is intended to work only in
>>> IMS networks
>>>>> that provide a CDIV service.
>>>>>
>>>>> If the authors choose the latter approach, we (the DISPATCH
>>>>> chairs) will choose a "Designated Expert" who will check the draft
>>>>> against the 7 points described in the document above.
>>>>>
>>>>> Cheers,
>>>>>
>>>>> Gonzalo
>>>>> DISPATCH co-chair
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>>>> _______________________________________________
>>>> dispatch mailing list
>>>> dispatch@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From gonzalo.camarillo@ericsson.com  Wed Feb 10 00:32:20 2010
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Date: Wed, 10 Feb 2010 10:33:25 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: Anders Kristensen <ankriste@cisco.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: Re: [dispatch] Prorgessing	draft-avasarala-dispatch-comm-div-notification-02
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Hi,

as I stated in my previous email, Paul's description of the situation
below is accurate.

Cheers,

Gonzalo

Paul Kyzivat wrote:
> Anders,
> 
> It was also my assumption going in that this is potentially a problem 
> area of general interest.
> 
> But the existing solution (as best I understand it) is wed to a 
> particular system architecture - not one that we would likely wish to 
> see enshrined in a more general standard.
> 
> To get beyond that we would need somebody who is interested in doing the 
> work to get something more general. I'm not seeing anybody stand up to 
> do that. (I know I don't have the time to do it.)
> 
> Lacking that, I see no justification to block their getting their event 
> package as long as it is suitably scoped.
> 
> 	Thanks,
> 	Paul
> 
> Anders Kristensen wrote:
>> IIUC the IMS use case (or one of them) is that users may configure call 
>> forwarding services and forget that they've done so. The CDIV feature 
>> then allows them to learn of calls being forwarded and take corrective 
>> action. This is not at all IMS specific - the same use case can be said 
>> to exist in non-IMS deployments.
>>
>> I suspect addressing this problem in a general way is no more work than 
>> to do it in an IMS specific way but would make for a better standard. 
>> What's more, if I'm right and this is inherently not an IMS specific 
>> problem then the solution is likely to be picked up and used in other 
>> contexts regardless of whatever stern wording is on the front page of 
>> the RFC.
>>
>> Thanks,
>> Anders
>>
>> On 2/9/2010 8:53 AM, DRAGE, Keith (Keith) wrote:
>>> I think that is a given and if that is the way to go, I'm sure we'll 
>>> all make sure the words on the front cover go that way.
>>>
>>> But there seemed to be some prior responses trying to take potential 
>>> use cases beyond that, and I wanted to make sure that if they existed, 
>>> they were explicitly declared.
>>>
>>>
>>> regards
>>>
>>> Keith
>>>
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Sent: Tuesday, February 09, 2010 3:57 PM
>>>> To: DRAGE, Keith (Keith)
>>>> Cc: Gonzalo Camarillo; DISPATCH; Mary Barnes
>>>> Subject: Re: [dispatch] Prorgessing
>>>> draft-avasarala-dispatch-comm-div-notification-02
>>>>
>>>> I don't have a problem with it being IMS-specific.
>>>> I just want it to *say* that it is IMS-specific.
>>>>
>>>>     Thanks,
>>>>     Paul
>>>>
>>>> DRAGE, Keith (Keith) wrote:
>>>>> At the moment, I think those of us looking at this from the
>>>> 3GPP way of doing things do not see a more general
>>>> application of this.
>>>>> I do not have a problem with doing it more generally, but I
>>>> guess the people who see a clear use case for that need to
>>>> speak up and identify their use cases.
>>>>> Otherwise we end up designing a general event package that
>>>> no one else ever uses. As opposed to the alternative of
>>>> clearly stating the restricted use to a particular
>>>> applicability, approving it, and moving on to something else
>>>> with our restricted amount of resources.
>>>>> regards
>>>>>
>>>>> Keith
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: dispatch-bounces@ietf.org
>>>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gonzalo Camarillo
>>>>>> Sent: Tuesday, February 09, 2010 9:16 AM
>>>>>> To: DISPATCH
>>>>>> Cc: Mary Barnes
>>>>>> Subject: [dispatch] Prorgessing
>>>>>> draft-avasarala-dispatch-comm-div-notification-02
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> there have been a set of messages on the list providing
>>>> comments on
>>>>>> the draft below:
>>>>>>
>>>>>> http://tools.ietf.org/html/draft-avasarala-dispatch-comm-div-n
>>>>>> otification-02
>>>>>>
>>>>>> As you know, the process for defining SIP event packages is
>>>>>> documented here:
>>>>>>
>>>>>> http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#se
>>>>>> ction-4.1
>>>>>>
>>>>>> Based on the feedback received, the authors need to decide whether
>>>>>> they want to generalize their solution so that is is generally
>>>>>> applicable to the public Internet or if they want to (further)
>>>>>> clarify that this mechanism is intended to work only in
>>>> IMS networks
>>>>>> that provide a CDIV service.
>>>>>>
>>>>>> If the authors choose the latter approach, we (the DISPATCH
>>>>>> chairs) will choose a "Designated Expert" who will check the draft
>>>>>> against the 7 points described in the document above.
>>>>>>
>>>>>> Cheers,
>>>>>>
>>>>>> Gonzalo
>>>>>> DISPATCH co-chair
>>>>>> _______________________________________________
>>>>>> dispatch mailing list
>>>>>> dispatch@ietf.org
>>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>>
>>>>> _______________________________________________
>>>>> dispatch mailing list
>>>>> dispatch@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>>>
>>> _______________________________________________
>>> dispatch mailing list
>>> dispatch@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dispatch
>>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


From L.Liess@telekom.de  Wed Feb 10 05:07:20 2010
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Subject: Re: [dispatch] I-DAction:draft-kaplan-dispatch-session-id-00.txt
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Hi Peter,

Thank you. This sounds great to me.=20

Sal is right that we have to define the requirements and use cases =
first, but I think having a feeling of how the solution could look like =
helps a lot.=20

Kind regards
Laura
=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Dawes, Peter, VF-Group
> Sent: Tuesday, February 09, 2010 5:16 PM
> To: Liess, Laura; dispatch@ietf.org
> Cc: gonzalo.camarillo@ericsson.com; HKaplan@acmepacket.com;=20
> H=FClsemann, Martin
> Subject: Re: [dispatch]=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> Hi Laura,
> As I understand it, the session-id header field is included in all
> requests. I think it would be possible to define a header field
> parameter for session-id, say "; debug", that is added to=20
> session-id by
> a UA iff certain conditions are met. This is similar to including the
> debug-id header field iff pre-supplied trigger conditions are=20
> matched in
> the UA. The first proxy can then police sesssion-id header fields
> containing this parameter (and let others go through unchecked), this
> parameter can trigger logging in downsteam SIP entities and so on to
> provide all functions of the debug draft. It might even be possible to
> use Proxy-Require based on this new parameter to make sure that the
> logging happens.=20
> I am not sure that I want to re-write my draft this way, but=20
> it could be
> a way to separate correlation-id in general from a specific use case
> (troubleshooting).
>=20
> Best Regards,
> Peter
>=20
>=20
>=20
>=20
>=20
> Message: 5
>=20
> Date: Tue, 9 Feb 2010 16:51:03 +0100
>=20
> From: <L.Liess@telekom.de>
>=20
> Subject: Re: [dispatch]
>=20
> I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>
>=20
> Cc: gonzalo.camarillo@ericsson.com, HKaplan@acmepacket.com,
>=20
> Martin.Huelsemann@telekom.de
>=20
> Message-ID:
>=20
> <40FB0FFB97588246A1BEFB05759DC8A003EFE9AB@S4DE9JSAANI.ost.t-com.de>
>=20
> Content-Type: text/plain; charset=3D"iso-8859-1"
>=20
> Hi Peter,=20
>=20
> One more question.=20
>=20
> >For example, I
>=20
> > imagine that the debug-id draft could be re-written to use=20
> session-id=20
>=20
> >(if session-id is chosen as the single "correlation id"
>=20
> > header) to hold
>=20
> > the "correlation id" with an added header field parameter=20
> to indicate=20
>=20
> >the extra functionality for reducing the network burden of=20
>=20
> >troubleshooting.
>=20
> I is not realy clear to me how you intend to use the new parameter
> field. Could you give me more details?
>=20
> Thanks a lot,
>=20
> Laura
>=20
> > -----Original Message-----
>=20
> > From: dispatch-bounces@ietf.org
>=20
> > [mailto:dispatch-bounces@ietf.org=20
> <mailto:dispatch-bounces@ietf.org> ]
> On Behalf Of Dawes, Peter, VF-Group
>=20
> > Sent: Monday, February 08, 2010 4:58 PM
>=20
> > To: dispatch@ietf.org
>=20
> > Cc: Dawes, Peter, VF-Group; gonzalo.camarillo@ericsson.com;=20
>=20
> > HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>=20
> > Subject: Re: [dispatch]
>=20
> > I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > Hi Laura,
>=20
> > What you describe below about debug-id is the correct=20
> understanding. I
>=20
>=20
> > am not convinced that using session-id for troubleshooting is=20
>=20
> > practical because it implies that everything has to be logged=20
>=20
> > everywhere in the network all the time so that you can, for=20
> example,=20
>=20
> > look at any signalling that happened two days ago. The debug-id is=20
>=20
> > inserted as required, and also policed at a couple of=20
> entities (first=20
>=20
> > proxy,
>=20
> > registrar) to take the burden of deciding whether or not to=20
> log away=20
>=20
> > from the rest of the network. The debug-id is also based on=20
> 3GPP and=20
>=20
> > OMA requirements, as I mentioned, but these requirements are, I=20
>=20
> > believe, necessary to minimize burden on the network.
>=20
> >=20
>=20
> > Starting to read through the threads that consider the idea=20
> from Paul
>=20
> > Kyzivat:
>=20
> >=20
>=20
> > > IMO use cases should be gathered and then bucketed into
>=20
> > sets that can
>=20
> >=20
>=20
> > > be
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > satisfied by a single mechanism, without assuming initially
>=20
> > how many
>=20
> >=20
>=20
> > > buckets there will be.
>=20
> >=20
>=20
> > the beginning of a list of use cases seems to be 3rd-party CC,=20
>=20
> > correlating multiple parallel dialogs, forked requests,=20
> requests that=20
>=20
> > pass through B2BUAs. I guess that if an unambiguous test can be=20
>=20
> > expressed for whether a particular session belongs to a set=20
> of related
>=20
>=20
> > sessions, then one header to hold the id that correlates them is=20
>=20
> > enough, even if populating it is difficult for some use cases. For=20
>=20
> > example, I imagine that the debug-id draft could be=20
> re-written to use=20
>=20
> > session-id (if session-id is chosen as the single "correlation id"
>=20
> > header) to hold
>=20
> > the "correlation id" with an added header field parameter=20
> to indicate=20
>=20
> > the extra functionality for reducing the network burden of=20
>=20
> > troubleshooting. This would satisfy the troubleshooting and=20
> 3GPP/OMA=20
>=20
> > requirements without limiting the meaning of the=20
> "correlation id" to=20
>=20
> > troubleshooting.
>=20
> >=20
>=20
> > Like you, I would like to find some way forward with the e2e=20
>=20
> > correlation topic (Spencer is right. BOF, CLF whatever...we have to=20
>=20
> > find some way to come forward with the e2e correlation.). Also, our=20
>=20
> > network guys consider session-id to be useful, but I would=20
> also like=20
>=20
> > to satisfy some OMA/3GPP requirements that it does not cover.
>=20
> >=20
>=20
> > Best Regards,
>=20
> > Peter
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> > Message: 3
>=20
> >=20
>=20
> > Date: Fri, 5 Feb 2010 17:21:39 +0100
>=20
> >=20
>=20
> > From: <L.Liess@telekom.de>
>=20
> >=20
>=20
> > Subject: Re: [dispatch]
>=20
> >=20
>=20
> > I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > To: <Peter.Dawes@vodafone.com>, <dispatch@ietf.org>,
>=20
> >=20
>=20
> > <spencer@wonderhamster.org>
>=20
> >=20
>=20
> > Cc: Gerold.Pinker@telekom.de, gonzalo.camarillo@ericsson.com,
>=20
> >=20
>=20
> > HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de
>=20
> >=20
>=20
> > Message-ID:
>=20
> >=20
>=20
> > <40FB0FFB97588246A1BEFB05759DC8A003EFE040@S4DE9JSAANI.ost.t-com.de>
>=20
> >=20
>=20
> > Content-Type: text/plain; charset=3D"iso-8859-1"
>=20
> >=20
>=20
> > Hi Peter,
>=20
> >=20
>=20
> > My understanding is that session-id and debug-id cover=20
> different sets=20
>=20
> > of requirements.
>=20
> >=20
>=20
> > Session-id is in every message. If people want to find out what=20
>=20
> > happened two days before, they can look at the old logs and the=20
>=20
> > session-id is there in every message to help them to correlate e2e.=20
>=20
> > For the debug-id to be there you have to know in advance that and=20
>=20
> > which e2e traffic you would like to have correlated. The=20
> debug-id is=20
>=20
> > activated "on-demand" and has no performance impact when it is not=20
>=20
> > active, session-id is active all the time. The advantage of the=20
>=20
> > debug-id is that there is no performance impact when=20
> debugging is not=20
>=20
> > needed. Is my understanding correct?
>=20
> >=20
>=20
> > I like the debug-id stuff, it's really smart and we looked=20
> at it. But=20
>=20
> > our troubleshooting and security guys want the session-id, for the=20
>=20
> > reason above. I assume every service provider has its own needs and=20
>=20
> > requirements.
>=20
> >=20
>=20
> > Spencer is right. BOF, CLF whatever...we have to find some=20
> way to come
>=20
>=20
> > forward with the e2e correlation.
>=20
> >=20
>=20
> > Thanks a lot
>=20
> >=20
>=20
> > Laura
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> > > -----Original Message-----
>=20
> >=20
>=20
> > > From: Dawes, Peter, VF-Group [mailto:Peter.Dawes@vodafone.com
> <mailto:Peter.Dawes@vodafone.com>=20
>=20
> > <mailto:Peter.Dawes@vodafone.com=20
> <mailto:Peter.Dawes@vodafone.com> > ]
>=20
> >=20
>=20
> > > Sent: Friday, February 05, 2010 4:00 PM
>=20
> >=20
>=20
> > > To: dispatch@ietf.org
>=20
> >=20
>=20
> > > Cc: Liess, Laura; gonzalo.camarillo@ericsson.com;
>=20
> >=20
>=20
> > > HKaplan@acmepacket.com; H?lsemann, Martin; Pinker, Gerold
>=20
> >=20
>=20
> > > Subject: FW: I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > Hello,
>=20
> >=20
>=20
> > > Gonzallo listed related work (Date: Thu, 14 Jan 2010 17:08:03
>=20
> >=20
>=20
> > > +0200) as
>=20
> >=20
>=20
> > > "1) We have Hadriel's draft, 2) We also have the following draft,
>=20
> >=20
>=20
> > > which eventually led to the disaggregated media work:
>=20
> >=20
>=20
> > > http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog>=20
>=20
> > <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog
> <http://www.watersprings.org/pub/id/draft-loreto-sipping-dialog> >
>=20
> >=20
>=20
> > > -correlati
>=20
> >=20
>=20
> > > on-01.txt 3) We also have the SIP-XMPP work, which may=20
> also need to
>=20
> >=20
>=20
> > > correlate sessions somehow."
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > I would like to add
>=20
> >=20
>=20
> > > http://tools.ietf.org/html/draft-dawes-sipping-debug-02
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02>=20
>=20
> > <http://tools.ietf.org/html/draft-dawes-sipping-debug-02
> <http://tools.ietf.org/html/draft-dawes-sipping-debug-02> > to this
>=20
> >=20
>=20
> > > discussion of use cases for some form of correlation
>=20
> > identifier. This
>=20
> >=20
>=20
> > > draft relates to debugging/troubleshooting to meet documented
>=20
> >=20
>=20
> > > requirements from 3GPP and OMA, and proposes a kind of correlation
>=20
> >=20
>=20
> > > identifier in a new header.
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > Regards,
>=20
> >=20
>=20
> > > Peter
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > Message: 2
>=20
> >=20
>=20
> > > Date: Fri, 22 Jan 2010 11:05:57 -0500
>=20
> >=20
>=20
> > > From: Paul Kyzivat <pkyzivat@cisco.com>
>=20
> >=20
>=20
> > > Subject: Re: [dispatch] FW:
>=20
> >=20
>=20
> > > I-DAction:draft-kaplan-dispatch-session-id-00.txt
>=20
> >=20
>=20
> > > To: L.Liess@telekom.de
>=20
> >=20
>=20
> > > Cc: dispatch@ietf.org, gonzalo.camarillo@ericsson.com,
>=20
> >=20
>=20
> > > HKaplan@acmepacket.com, Martin.Huelsemann@telekom.de,
>=20
> >=20
>=20
> > > Gerold.Pinker@telekom.de
>=20
> >=20
>=20
> > > Message-ID: <4B59CCE5.2010202@cisco.com>
>=20
> >=20
>=20
> > > Content-Type: text/plain; charset=3DISO-8859-1; format=3Dflowed
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > >=20
>=20
> >=20
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> > >=20
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> >=20
>=20
> > > L.Liess@telekom.de wrote:
>=20
> >=20
>=20
> > > > Paul,
>=20
> >=20
>=20
> > > >=20
>=20
> >=20
>=20
> > > > Thank you for the examples. The scenarios can be really=20
> complex. I
>=20
> >=20
>=20
> > > hope we can get these correlation identifiers, whatever
>=20
> > their names,
>=20
> >=20
>=20
> > > to work correctly in all cases....
>=20
> >=20
>=20
> > > >=20
>=20
> >=20
>=20
> > > > I am not sure I understood your message correctly. Do you say we
>=20
> >=20
>=20
> > > should have two different identifiers because we use them for two
>=20
> >=20
>=20
> > > different things?
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > I think there may be more than two distinct usages that
>=20
> > will require
>=20
> >=20
>=20
> > > different ones.
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > IMO use cases should be gathered and then bucketed into
>=20
> > sets that can
>=20
> >=20
>=20
> > > be
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > satisfied by a single mechanism, without assuming initially
>=20
> > how many
>=20
> >=20
>=20
> > > buckets there will be.
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > (But I'm still on record as not being convinced that a
>=20
> > correlation id
>=20
> >=20
>=20
> > > is
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > needed *at all* for the case when multiple sip UAs are
>=20
> > participating
>=20
> >=20
>=20
> > > on behalf of a single user in a call.)
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> > > > Thanks a lot
>=20
> >=20
>=20
> > > > Laura
>=20
> >=20
>=20
> > > >=20
>=20
> >=20
>=20
> > >=20
>=20
> >=20
>=20
> >=20
>=20
> >=20
>=20
> > _______________________________________________
>=20
> > dispatch mailing list
>=20
> > dispatch@ietf.org
>=20
> > https://www.ietf.org/mailman/listinfo/dispatch
> <https://www.ietf.org/mailman/listinfo/dispatch>=20
>=20
> >=20
>=20
> =20
>=20
> ------------------------------
>=20
> _______________________________________________
>=20
> dispatch mailing list
>=20
> dispatch@ietf.org
>=20
> https://www.ietf.org/mailman/listinfo/dispatch
> <https://www.ietf.org/mailman/listinfo/dispatch>=20
>=20
> =20
>=20
> End of dispatch Digest, Vol 11, Issue 21
>=20
> ****************************************
>=20
> =20
> =20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From fluffy@cisco.com  Wed Feb 10 13:28:40 2010
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To: DISPATCH <dispatch@ietf.org>
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Subject: [dispatch] Name for Session Recording WG
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In IESG review, it got pointed out this name conflicts with SRP (Securer =
Remote Password) in the security context. Could folks propose some other =
name for the WG be early next week.=20

Thanks, Cullen




From stpeter@stpeter.im  Wed Feb 10 13:31:02 2010
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From: Peter Saint-Andre <stpeter@stpeter.im>
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On 2/10/10 2:29 PM, Cullen Jennings wrote:
>=20
> In IESG review, it got pointed out this name conflicts with SRP
> (Securer Remote Password) in the security context. Could folks
> propose some other name for the WG be early next week.

I'd suggest "Memorex" but there might be trademark issues. :)

Peter

--=20
Peter Saint-Andre
https://stpeter.im/




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--------------ms010500030109030703080101--

From emcho@sip-communicator.org  Wed Feb 10 13:32:03 2010
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Date: Wed, 10 Feb 2010 22:33:07 +0100
From: Emil Ivov <emcho@sip-communicator.org>
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To: Simo.Veikkolainen@nokia.com
References: <B23B311878A0B4438F5F09F47E01AAE04DC322A03F@NOK-EUMSG-01.mgdnok.nokia.com>	<1260472418.25475.1915.camel@scott>	<8647CF2D344C40F19ABD162093605F46@china.huawei.com>	<4B214EA7.1090405@sip-communicator.org>	<787991D583944628AC6F8AB34A67EF4F@china.huawei.com>	<4B215477.7000003@sip-communicator.org>	<B23B311878A0B4438F5F09F47E01AAE04DC33957F8@NOK-EUMSG-01.mgdnok.nokia.com>	<4B26233D.60104@sip-communicator.org> <4B26386C.5040704@cisco.com>	<01fa01ca7cc3$73e8e430$5bbaac90$%roni@huawei.com>	<B23B311878A0B4438F5F09F47E01AAE04DC3395C81@NOK-EUMSG-01.mgdnok.nokia.com>	<4B2658FB.50506@sip-communicator.org>	<027001ca7d11$5577c8b0$00675a10$%roni@huawei.com>	<B23B311878A0B4438F5F09F47E01AAE04DC342C3A8@NOK-EUMSG-01.mgdnok.nokia.com>	<038c01ca7e4d$f7828c70$e687a550$%roni@huawei.com>	<B23B311878A0B4438F5F09F47E01@NOK-EUMSG-01.mgdnok.nokia.com> <03fd01ca7f18$8772c9b0$96585d10$%roni@huawei.com> <B23B311878A0B4438F5F09F47E01AAE04E2B49A911@NOK-EUMSG-01.mgdnok.nokia.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] Revised proposed charter for Combining SIP and XMPP
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Hey all,

Just wondering where this charter currently stands. Is there any
decision as to what will be happening with it?

Cheers,
Emil

Simo.Veikkolainen@nokia.com =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=B0:
> Hello,
>=20
> I had an offline email exchange with Roni during which he mentioned
> that he would be fine with the charter text as it currently stands.
>=20
> The other concern that was raised by Emil was on using shared
> credentials for SIP and XMPP authentication. My conclusion seemed to
> be that the current charter text would be ok, but we can still
> document such a feature during the protocol specification phase if
> that is deemed useful.
>=20
> If my understanding is correct, and unless there are other comments
> to the charter proposal, it is good to go. Please let us know asap if
> there is still something you would like to comment.
>=20
> Regards, Simo
>=20
>> -----Original Message----- From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Roni Even Sent:
>> 17 December, 2009 14:58 To: Veikkolainen Simo (Nokia-D/Helsinki);
>> emcho@sip-communicator.org Cc: dispatch@ietf.org Subject: Re:
>> [dispatch] Revised proposed charter for Combining SIP and XMPP
>>=20
>> Simo, The other user has two endpoints, one is a video conferencing
>> appliance and the other is a PC running XMPP client. The user will
>> register his XMPP and SIP capability using the two endpoints in the
>> infrastructure Roni
>>=20
>>> -----Original Message----- From: dispatch-bounces@ietf.org
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of
>>> Simo.Veikkolainen@nokia.com Sent: Thursday, December 17, 2009
>>> 2:34 PM To: Even.roni@huawei.com; emcho@sip-communicator.org Cc:
>>> dispatch@ietf.org Subject: Re: [dispatch] Revised proposed
>>> charter for Combining SIP and XMPP
>>>=20
>>> Roni,
>>>=20
>>>>> A "dual stack" SIP/XMPP client should be able to talk SIP to
>>>>> a SIP-
>>>> only
>>>>> endpoint and XMPP to an XMPP-only endpoint. What is that you
>>>>> need
>> in
>>>>> addition to this?
>>>> The dual stack client need to know that these two clients are
>>>> the
>> same
>>>> user. For example start a XMPP chat and add the SIP video call.
>>>> This
>>> is
>>>> not advertised by the remote XMPP client but by the
>>>> infrastructure.
>>>>=20
>>>> From the other direction (two systems) it will involve starting
>>>> chat from XMPP and adding SIP video to the call by the XMPP
>>>> client for example by telling the other side or the SIP server
>>>> to call the his
>>> SIP
>>>> client on the appliance.
>>>>=20
>>> Let me see if I understood you correctly. It sounds you are
>>> looking
>> for
>>> two use cases:
>>>=20
>>> 1) to add SIP voice/video or XMPP chat from an integrated
>>> endpoint
>> even
>>> though the other endpoint is SIP-only or XMPP-only
>>>=20
>>> 2) to add SIP voice/video or XMPP chat from a XMPP-only or
>>> SIP-only endpoint, respectively.
>>>=20
>>> For 2), I don't see how that is possible without changes in the
>>> client side.
>>>=20
>>> For 1), you would need to be able, from a "dual stack" endpoint,
>>> to tell the XMPP server to start an XMPP session, or the SIP
>>> server to make a SIP call to the same user you already have a SIP
>>> or XMPP
>> session
>>> with. While probably doable, this is not what we had in mind for
>>> this work.
>>>=20
>>> Simo _______________________________________________ dispatch
>>> mailing list dispatch@ietf.org=20
>>> https://www.ietf.org/mailman/listinfo/dispatch
>> _______________________________________________ dispatch mailing
>> list dispatch@ietf.org=20
>> https://www.ietf.org/mailman/listinfo/dispatch
>=20

--=20
Emil Ivov, Ph.D.                               67000 Strasbourg,
Project Lead                                   France
SIP Communicator
emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
http://sip-communicator.org                    FAX:   +33.1.77.62.47.31


From Leon.Portman@nice.com  Wed Feb 10 14:20:38 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: "'fluffy@cisco.com'" <fluffy@cisco.com>, "'dispatch@ietf.org'" <dispatch@ietf.org>
Date: Thu, 11 Feb 2010 00:21:45 +0200
Thread-Topic: [dispatch] Name for Session Recording WG
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Possible names:

RECON - recording control
REX - recording extensions
INTER - Interactions Recording
MMREC - multimedia recording


Regards

Leon


----- Original Message -----
From: dispatch-bounces@ietf.org <dispatch-bounces@ietf.org>
To: DISPATCH <dispatch@ietf.org>
Sent: Wed Feb 10 23:29:50 2010=0A=
Subject: [dispatch] Name for Session Recording WG


In IESG review, it got pointed out this name conflicts with SRP (Securer Re=
mote Password) in the security context. Could folks propose some other name=
 for the WG be early next week.=20

Thanks, Cullen



_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

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I would suggest "Network Enabled diVERsion of Multimedia On Recording 
Equipment," or "NEVERMORE."

/a

On 2/10/10 15:29, Feb 10, Cullen Jennings wrote:
> In IESG review, it got pointed out this name conflicts with SRP (Securer Remote Password) in the security context. Could folks propose some other name for the WG be early next week.
>
> Thanks, Cullen
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>    


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Subject: Re: [dispatch] Name for Session Recording WG
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MARE - Multimedia Assets REcording

It's an italian word... it stands for "sea"

Tobia




Quoting Cullen Jennings <fluffy@cisco.com>:

>
> In IESG review, it got pointed out this name conflicts with SRP   
> (Securer Remote Password) in the security context. Could folks   
> propose some other name for the WG be early next week.
>
> Thanks, Cullen
>
>
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>




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Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Name for Session Recording WG
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On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
> In IESG review, it got pointed out this name conflicts with SRP
> (Securer Remote Password) in the security context. Could folks propose
> some other name for the WG be early next week. 

Unified Session Archival In Datastore (USAID)



From christer.holmberg@ericsson.com  Thu Feb 11 04:38:40 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Scott Lawrence <scottlawrenc@avaya.com>, Cullen Jennings <fluffy@cisco.com>
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Intermediate Media Storage (IMS)=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
> Sent: 11. helmikuuta 2010 14:36
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>=20
> On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
> > In IESG review, it got pointed out this name conflicts with SRP=20
> > (Securer Remote Password) in the security context. Could=20
> folks propose=20
> > some other name for the WG be early next week.
>=20
> Unified Session Archival In Datastore (USAID)
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From lorenzo@meetecho.com  Thu Feb 11 04:43:27 2010
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From: Lorenzo Miniero <lorenzo@meetecho.com>
To: Adam Roach <adam@nostrum.com>
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I like this name, and the fact there's a metal band with the same name
is totally irrelevant! ;)

L.


On Wed, 10 Feb 2010 16:50:13 -0600
Adam Roach <adam@nostrum.com> wrote:

> I would suggest "Network Enabled diVERsion of Multimedia On Recording 
> Equipment," or "NEVERMORE."
> 
> /a
> 
> On 2/10/10 15:29, Feb 10, Cullen Jennings wrote:
> > In IESG review, it got pointed out this name conflicts with SRP (Securer Remote Password) in the security context. Could folks propose some other name for the WG be early next week.
> >
> > Thanks, Cullen
> >
> >
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >    
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 
Lorenzo Miniero
Meetecho s.r.l.
http://www.meetecho.com/

From Markus.Isomaki@nokia.com  Thu Feb 11 05:06:23 2010
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From: <Markus.Isomaki@nokia.com>
To: <emcho@sip-communicator.org>, <Simo.Veikkolainen@nokia.com>
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Subject: Re: [dispatch] Revised proposed charter for Combining SIP and XMPP
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Hi Emil,

We have gotten the charter proposal as far as the RAI ADs, but we haven't g=
otten an approval yet.

Simo and me are anyway working on a draft (based on our earlier draft) that=
 would crystallize the use cases and requirements, and would fit as the fir=
st deliverable according to the charter. Once we get that out, I propose to=
 focus the discussion on that to make sure we agree on that part of the wor=
k.=20

The technical solution work can definitely progress in parallel, but right =
now at least Simo and me don't have much to add to what we have already sub=
mitted and what was discussed in Hiroshima and on the list. It would be use=
ful to put together a presentation of the different options (the ones outli=
ned in our current draft and those proposed by others) together before Anah=
eim also to see how easy it will be to find consensus on those. Would someo=
ne volunteer to take that task? (If not, Simo and me can take a stab on tha=
t too, but I remember a few hands were raised at the BOF :-).

If we manage to get a slot at Anaheim I suppose these would be the things o=
n the agenda as well:
1. Use cases and requirements --> are we happy to adopt a draft as a WG ite=
m to work towards the first deliverable
2. Go through the different technical approaches --> can we find consensus =
and also can we find an editor for the draft or drafts defining the extensi=
ons we want to standardize

Regarding the combined client provisioning/configuration issue that you, Em=
il, brought up: I was thinking that it might be wise to actually write a sh=
ort draft (a BCP) about the best ways how to arrange the autoconfig of a co=
mbined SIP/XMPP client. On SIP side we finally a pretty decent approach wit=
h the recent SIP Forum UA-config work (Scott Lawrence, John Elwell et al) a=
nd there are ways for XMPP as well, so we could take those as the baseline.=
 Perhaps we could write something about the case that a single service prov=
ider offers both SIP and XMPP accounts and what synergies might exist for s=
uch a case. Do you Emil think that would be a useful thing to have?

Markus


>-----Original Message-----
>From: dispatch-bounces@ietf.org=20
>[mailto:dispatch-bounces@ietf.org] On Behalf Of ext Emil Ivov
>Sent: 10 February, 2010 23:33
>To: Veikkolainen Simo (Nokia-D/Helsinki)
>Cc: dispatch@ietf.org
>Subject: Re: [dispatch] Revised proposed charter for Combining=20
>SIP and XMPP
>
>Hey all,
>
>Just wondering where this charter currently stands. Is there=20
>any decision as to what will be happening with it?
>
>Cheers,
>Emil
>
>Simo.Veikkolainen@nokia.com =CE=C1=D0=C9=D3=C1:
>> Hello,
>>=20
>> I had an offline email exchange with Roni during which he mentioned=20
>> that he would be fine with the charter text as it currently stands.
>>=20
>> The other concern that was raised by Emil was on using shared=20
>> credentials for SIP and XMPP authentication. My conclusion seemed to=20
>> be that the current charter text would be ok, but we can still=20
>> document such a feature during the protocol specification phase if=20
>> that is deemed useful.
>>=20
>> If my understanding is correct, and unless there are other=20
>comments to=20
>> the charter proposal, it is good to go. Please let us know asap if=20
>> there is still something you would like to comment.
>>=20
>> Regards, Simo
>>=20
>>> -----Original Message----- From: dispatch-bounces@ietf.org=20
>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of ext Roni Even Sent:
>>> 17 December, 2009 14:58 To: Veikkolainen Simo (Nokia-D/Helsinki);=20
>>> emcho@sip-communicator.org Cc: dispatch@ietf.org Subject: Re:
>>> [dispatch] Revised proposed charter for Combining SIP and XMPP
>>>=20
>>> Simo, The other user has two endpoints, one is a video conferencing=20
>>> appliance and the other is a PC running XMPP client. The user will=20
>>> register his XMPP and SIP capability using the two endpoints in the=20
>>> infrastructure Roni
>>>=20
>>>> -----Original Message----- From: dispatch-bounces@ietf.org=20
>>>> [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
>>>> Simo.Veikkolainen@nokia.com Sent: Thursday, December 17, 2009
>>>> 2:34 PM To: Even.roni@huawei.com; emcho@sip-communicator.org Cc:
>>>> dispatch@ietf.org Subject: Re: [dispatch] Revised proposed charter=20
>>>> for Combining SIP and XMPP
>>>>=20
>>>> Roni,
>>>>=20
>>>>>> A "dual stack" SIP/XMPP client should be able to talk SIP to a=20
>>>>>> SIP-
>>>>> only
>>>>>> endpoint and XMPP to an XMPP-only endpoint. What is that you need
>>> in
>>>>>> addition to this?
>>>>> The dual stack client need to know that these two clients are the
>>> same
>>>>> user. For example start a XMPP chat and add the SIP video call.
>>>>> This
>>>> is
>>>>> not advertised by the remote XMPP client but by the=20
>infrastructure.
>>>>>=20
>>>>> From the other direction (two systems) it will involve starting=20
>>>>> chat from XMPP and adding SIP video to the call by the=20
>XMPP client=20
>>>>> for example by telling the other side or the SIP server=20
>to call the=20
>>>>> his
>>>> SIP
>>>>> client on the appliance.
>>>>>=20
>>>> Let me see if I understood you correctly. It sounds you are looking
>>> for
>>>> two use cases:
>>>>=20
>>>> 1) to add SIP voice/video or XMPP chat from an integrated endpoint
>>> even
>>>> though the other endpoint is SIP-only or XMPP-only
>>>>=20
>>>> 2) to add SIP voice/video or XMPP chat from a XMPP-only or=20
>SIP-only=20
>>>> endpoint, respectively.
>>>>=20
>>>> For 2), I don't see how that is possible without changes in the=20
>>>> client side.
>>>>=20
>>>> For 1), you would need to be able, from a "dual stack"=20
>endpoint, to=20
>>>> tell the XMPP server to start an XMPP session, or the SIP=20
>server to=20
>>>> make a SIP call to the same user you already have a SIP or XMPP
>>> session
>>>> with. While probably doable, this is not what we had in mind for=20
>>>> this work.
>>>>=20
>>>> Simo _______________________________________________ dispatch=20
>>>> mailing list dispatch@ietf.org=20
>>>> https://www.ietf.org/mailman/listinfo/dispatch
>>> _______________________________________________ dispatch=20
>mailing list=20
>>> dispatch@ietf.org https://www.ietf.org/mailman/listinfo/dispatch
>>=20
>
>--=20
>Emil Ivov, Ph.D.                               67000 Strasbourg,
>Project Lead                                   France
>SIP Communicator
>emcho@sip-communicator.org                     PHONE: +33.1.77.62.43.30
>http://sip-communicator.org                    FAX:   +33.1.77.62.47.31
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch
>=

From steve.logalbo@cybertech-na.com  Thu Feb 11 05:59:34 2010
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From: "Steven LoGalbo" <steve.logalbo@cybertech-na.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, "Scott Lawrence" <scottlawrenc@avaya.com>, "Cullen Jennings" <fluffy@cisco.com>
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SRIP - Session Recording Initiation Protocol



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Christer Holmberg
Sent: Thursday, February 11, 2010 7:40 AM
To: Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG


Intermediate Media Storage (IMS)=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
> Sent: 11. helmikuuta 2010 14:36
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>=20
> On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
> > In IESG review, it got pointed out this name conflicts with SRP=20
> > (Securer Remote Password) in the security context. Could=20
> folks propose=20
> > some other name for the WG be early next week.
>=20
> Unified Session Archival In Datastore (USAID)
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
_______________________________________________
dispatch mailing list
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https://www.ietf.org/mailman/listinfo/dispatch

From andrew.hutton@siemens-enterprise.com  Thu Feb 11 06:06:15 2010
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To: Steven LoGalbo <steve.logalbo@cybertech-na.com>, Christer Holmberg <christer.holmberg@ericsson.com>, Scott Lawrence <scottlawrenc@avaya.com>, Cullen Jennings <fluffy@cisco.com>
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I quite like SRIP but I don't think we should call it a protocol.

I suggest "Media Recording using the Session Initiation Protocol" or "MRSIP=
".

Regards
Andy
=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Steven LoGalbo
Sent: 11 February 2010 13:57
To: Christer Holmberg; Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

SRIP - Session Recording Initiation Protocol



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Christer Holmberg
Sent: Thursday, February 11, 2010 7:40 AM
To: Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG


Intermediate Media Storage (IMS)=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
> Sent: 11. helmikuuta 2010 14:36
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>=20
> On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
> > In IESG review, it got pointed out this name conflicts with SRP=20
> > (Securer Remote Password) in the security context. Could=20
> folks propose=20
> > some other name for the WG be early next week.
>=20
> Unified Session Archival In Datastore (USAID)
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
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From steve.logalbo@cybertech-na.com  Thu Feb 11 06:23:12 2010
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From: "Steven LoGalbo" <steve.logalbo@cybertech-na.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, "Christer Holmberg" <christer.holmberg@ericsson.com>, "Scott Lawrence" <scottlawrenc@avaya.com>, "Cullen Jennings" <fluffy@cisco.com>
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SRI - Session Recording Interface?
IPSR - IP Session Recording?

Regards,

Steve LoGalbo
steve.logalbo@cybertech-na.com
914-269-4099



-----Original Message-----
From: Hutton, Andrew [mailto:andrew.hutton@siemens-enterprise.com]=20
Sent: Thursday, February 11, 2010 9:07 AM
To: Steven LoGalbo; Christer Holmberg; Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: RE: [dispatch] Name for Session Recording WG


I quite like SRIP but I don't think we should call it a protocol.

I suggest "Media Recording using the Session Initiation Protocol" or
"MRSIP".

Regards
Andy
=20

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Steven LoGalbo
Sent: 11 February 2010 13:57
To: Christer Holmberg; Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

SRIP - Session Recording Initiation Protocol



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Christer Holmberg
Sent: Thursday, February 11, 2010 7:40 AM
To: Scott Lawrence; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG


Intermediate Media Storage (IMS)=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
> Sent: 11. helmikuuta 2010 14:36
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>=20
> On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
> > In IESG review, it got pointed out this name conflicts with SRP=20
> > (Securer Remote Password) in the security context. Could=20
> folks propose=20
> > some other name for the WG be early next week.
>=20
> Unified Session Archival In Datastore (USAID)
>=20
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch

From vkg@alcatel-lucent.com  Thu Feb 11 06:41:34 2010
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tobia.castaldi@unina.it wrote:
Cullen Jennings <fluffy@cisco.com> wrote:
>
> In IESG review, it got pointed out this name conflicts with SRP  
> (Securer Remote Password) in the security context. Could folks  
> propose some other name for the WG be early next week.

Someone suggested "memorex", which was good.  But if "memorex"
is infringing on a possible trademark, how about

   yaad - Yet Another Acoustic Diverter

Plus, "yaad" is a Hindi word for "memory".

Ciao,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From Leon.Portman@nice.com  Thu Feb 11 06:54:48 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Thu, 11 Feb 2010 16:55:59 +0200
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Another option:

RUM - Recording of Unified Media


Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Vijay K. Gurbani
Sent: Thursday, February 11, 2010 4:42 PM
To: Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

tobia.castaldi@unina.it wrote:
Cullen Jennings <fluffy@cisco.com> wrote:
>
> In IESG review, it got pointed out this name conflicts with SRP =20
> (Securer Remote Password) in the security context. Could folks =20
> propose some other name for the WG be early next week.

Someone suggested "memorex", which was good.  But if "memorex"
is infringing on a possible trademark, how about

   yaad - Yet Another Acoustic Diverter

Plus, "yaad" is a Hindi word for "memory".

Ciao,

- vijay
--=20
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/
_______________________________________________
dispatch mailing list
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From lorenzo@meetecho.com  Thu Feb 11 06:54:52 2010
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From: Lorenzo Miniero <lorenzo@meetecho.com>
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SINARP (Sinarp Is Not A Recording Protocol)

L.


On Thu, 11 Feb 2010 08:42:10 -0600
"Vijay K. Gurbani" <vkg@alcatel-lucent.com> wrote:

> tobia.castaldi@unina.it wrote:
> Cullen Jennings <fluffy@cisco.com> wrote:
> >
> > In IESG review, it got pointed out this name conflicts with SRP  
> > (Securer Remote Password) in the security context. Could folks  
> > propose some other name for the WG be early next week.
> 
> Someone suggested "memorex", which was good.  But if "memorex"
> is infringing on a possible trademark, how about
> 
>    yaad - Yet Another Acoustic Diverter
> 
> Plus, "yaad" is a Hindi word for "memory".
> 
> Ciao,
> 
> - vijay
> -- 
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 


-- 
Lorenzo Miniero
Meetecho s.r.l.
http://www.meetecho.com/

From steve.logalbo@cybertech-na.com  Thu Feb 11 07:01:25 2010
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From: "Steven LoGalbo" <steve.logalbo@cybertech-na.com>
To: "Leon Portman" <Leon.Portman@nice.com>, "Cullen Jennings" <fluffy@cisco.com>
Cc: DISPATCH <dispatch@ietf.org>
Subject: Re: [dispatch] Name for Session Recording WG
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I like where you are going.  How about UCR / UMR- Unified Communication
Recording / Unified Media Recording

Regards,

Steve LoGalbo
steve.logalbo@cybertech-na.com
914-269-4099



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Leon Portman
Sent: Thursday, February 11, 2010 9:56 AM
To: Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

Another option:

RUM - Recording of Unified Media


Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Vijay K. Gurbani
Sent: Thursday, February 11, 2010 4:42 PM
To: Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

tobia.castaldi@unina.it wrote:
Cullen Jennings <fluffy@cisco.com> wrote:
>
> In IESG review, it got pointed out this name conflicts with SRP =20
> (Securer Remote Password) in the security context. Could folks =20
> propose some other name for the WG be early next week.

Someone suggested "memorex", which was good.  But if "memorex"
is infringing on a possible trademark, how about

   yaad - Yet Another Acoustic Diverter

Plus, "yaad" is a Hindi word for "memory".

Ciao,

- vijay
--=20
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/
_______________________________________________
dispatch mailing list
dispatch@ietf.org
https://www.ietf.org/mailman/listinfo/dispatch
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What happened to the goal of naming RAI WGs after alcohols? :-)

Margarita - MediA RecordinG And Replay In The internet Architecture

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Beh=
alf
> Of Vijay K. Gurbani
> Sent: Thursday, February 11, 2010 9:42 AM
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>
> tobia.castaldi@unina.it wrote:
> Cullen Jennings <fluffy@cisco.com> wrote:
> >
> > In IESG review, it got pointed out this name conflicts with SRP
> > (Securer Remote Password) in the security context. Could folks
> > propose some other name for the WG be early next week.
>
> Someone suggested "memorex", which was good.  But if "memorex"
> is infringing on a possible trademark, how about
>
>    yaad - Yet Another Acoustic Diverter
>
> Plus, "yaad" is a Hindi word for "memory".
>
> Ciao,
>
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From: "Steven LoGalbo" <steve.logalbo@cybertech-na.com>
To: "Jain, Rajnish" <Rajnish.Jain@ipc.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, "Cullen Jennings" <fluffy@cisco.com>
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DRUNK - Dynamic Recording of UNified Kommunications...

Regards,

Steve LoGalbo
steve.logalbo@cybertech-na.com
914-269-4099



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Jain, Rajnish
Sent: Thursday, February 11, 2010 10:06 AM
To: Vijay K. Gurbani; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

What happened to the goal of naming RAI WGs after alcohols? :-)

Margarita - MediA RecordinG And Replay In The internet Architecture

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
> Of Vijay K. Gurbani
> Sent: Thursday, February 11, 2010 9:42 AM
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>
> tobia.castaldi@unina.it wrote:
> Cullen Jennings <fluffy@cisco.com> wrote:
> >
> > In IESG review, it got pointed out this name conflicts with SRP
> > (Securer Remote Password) in the security context. Could folks
> > propose some other name for the WG be early next week.
>
> Someone suggested "memorex", which was good.  But if "memorex"
> is infringing on a possible trademark, how about
>
>    yaad - Yet Another Acoustic Diverter
>
> Plus, "yaad" is a Hindi word for "memory".
>
> Ciao,
>
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From Leon.Portman@nice.com  Thu Feb 11 07:31:09 2010
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From: Leon Portman <Leon.Portman@nice.com>
To: Steven LoGalbo <steve.logalbo@cybertech-na.com>, "Jain, Rajnish" <Rajnish.Jain@ipc.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, Cullen Jennings <fluffy@cisco.com>
Date: Thu, 11 Feb 2010 17:32:17 +0200
Thread-Topic: [dispatch] Name for Session Recording WG
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Good one

Leon

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On Behal=
f Of Steven LoGalbo
Sent: Thursday, February 11, 2010 5:22 PM
To: Jain, Rajnish; Vijay K. Gurbani; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

DRUNK - Dynamic Recording of UNified Kommunications...

Regards,

Steve LoGalbo
steve.logalbo@cybertech-na.com
914-269-4099



-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Jain, Rajnish
Sent: Thursday, February 11, 2010 10:06 AM
To: Vijay K. Gurbani; Cullen Jennings
Cc: DISPATCH
Subject: Re: [dispatch] Name for Session Recording WG

What happened to the goal of naming RAI WGs after alcohols? :-)

Margarita - MediA RecordinG And Replay In The internet Architecture

> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf
> Of Vijay K. Gurbani
> Sent: Thursday, February 11, 2010 9:42 AM
> To: Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>
> tobia.castaldi@unina.it wrote:
> Cullen Jennings <fluffy@cisco.com> wrote:
> >
> > In IESG review, it got pointed out this name conflicts with SRP
> > (Securer Remote Password) in the security context. Could folks
> > propose some other name for the WG be early next week.
>
> Someone suggested "memorex", which was good.  But if "memorex"
> is infringing on a possible trademark, how about
>
>    yaad - Yet Another Acoustic Diverter
>
> Plus, "yaad" is a Hindi word for "memory".
>
> Ciao,
>
> - vijay
> --
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch

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From spromano@unina.it  Thu Feb 11 09:03:26 2010
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Subject: Re: [dispatch] Name for Session Recording WG
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RESCUE -- REcording Sessions among inter-Communicating User Equipment

Simon

Il giorno 11/feb/2010, alle ore 15.07, "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com 
 > ha scritto:

>
> I quite like SRIP but I don't think we should call it a protocol.
>
> I suggest "Media Recording using the Session Initiation Protocol" or  
> "MRSIP".
>
> Regards
> Andy
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org]  
> On Behalf Of Steven LoGalbo
> Sent: 11 February 2010 13:57
> To: Christer Holmberg; Scott Lawrence; Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>
> SRIP - Session Recording Initiation Protocol
>
>
>
> -----Original Message-----
> From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Thursday, February 11, 2010 7:40 AM
> To: Scott Lawrence; Cullen Jennings
> Cc: DISPATCH
> Subject: Re: [dispatch] Name for Session Recording WG
>
>
> Intermediate Media Storage (IMS)
>
>> -----Original Message-----
>> From: dispatch-bounces@ietf.org
>> [mailto:dispatch-bounces@ietf.org] On Behalf Of Scott Lawrence
>> Sent: 11. helmikuuta 2010 14:36
>> To: Cullen Jennings
>> Cc: DISPATCH
>> Subject: Re: [dispatch] Name for Session Recording WG
>>
>> On Wed, 2010-02-10 at 14:29 -0700, Cullen Jennings wrote:
>>> In IESG review, it got pointed out this name conflicts with SRP
>>> (Securer Remote Password) in the security context. Could
>> folks propose
>>> some other name for the WG be early next week.
>>
>> Unified Session Archival In Datastore (USAID)
>>
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
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From dworley@avaya.com  Thu Feb 11 11:17:38 2010
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On Thu, 2010-02-11 at 13:43 +0100, Lorenzo Miniero wrote:
> I like this name, and the fact there's a metal band with the same name
> is totally irrelevant! ;)

Artist:  The Bobs
Song:    Naming The Band
------------------------------------------------------------------------------
(c) 1989 Richard Greene, Gunnar Madsen and Joe Finetti,
      Best of Breed Music (ASCAP)
    All Rights Reserved.  Used by Permission.
------------------------------------------------------------------------------

My parents were wrong, they named me "tree"
It's probably the cruelest thing they ever did to me
I thought a long time about changing my name
But I found a better way to deal with my shame
My buddies and me formed a band
We're gonna be famous, you understand

We're lookin' for a drummer
Or someone with a van
Our hair is getting longer
But the most important thing is namin' the band
Namin' the band

...



From jmpolk@cisco.com  Thu Feb 11 11:34:10 2010
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At 03:29 PM 2/10/2010, Cullen Jennings wrote:

>In IESG review, it got pointed out this name conflicts with SRP 
>(Securer Remote Password) in the security context. Could folks 
>propose some other name for the WG be early next week.

what about

          yomomma - Yet anOther Memory archive fOr MultiMediA

James


>Thanks, Cullen
>
>
>
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From vkg@alcatel-lucent.com  Thu Feb 11 11:53:44 2010
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Cullen: Now that the world has proof-positive of the
unbounded intellect of DISPATCHers in conjuring
up creative acronyms at short notice, here is a list of
entrants so far.  Your executive decision now is to
pick one of these or come up with a new one of your
own choosing.

The guilty party of each acronym is clearly identified
for recrimination (or renumeration) at some later time.

MARE      - Multimedia Assets REcording (Tobia Castaldi)
             Note: Italian word for "sea"
USAID     - Unified Session Archival In Datastore (Scott Lawrence)
IMS       - Intermediate Media Storage (Christer Holmberg)
NEVERMORE - Network Enabled diVERsion of Multimedia On
             Recording Equipment (Adam Roach)
RECON     - recording control (Leon Portman)
REX       - recording extensions (Leon Portman)
INTER     - Interactions Recording (Leon Portman)
MMREC     - multimedia recording (Leon Portman)
RUM       - Recording of Unified Media (Leon Portman)
             Note: Following the tradition of naming after alcohols.
SRIP      - Session Recording Initiation Protocol (Steve LoGalbo)
SRI       - Session Recording Interface? (Steve LoGalbo)
IPSR      - IP Session Recording? (Steve LoGalbo)
UCR / UMR - Unified Communication Recording / Unified Media Recording
             (Steve LoGalbo)
DRUNK     - Dynamic Recording of UNified Kommunications (Steve LoGalbo)
             Note: Following the tradition of naming after alcohols.
MRSIP     - Media Recording using the Session Initiation Protocol
            (Andrew Hutton)
YAAD      - Yet Another Acoustic Diverter (Vijay K. Gurbani)
             Note: Hindi word for "memory".
SINARP    - Sinarp Is Not A Recording Protocol (Lorenzo Miniero)
MARGARITA - MediA RecordinG And Replay In The internet Architecture
             (Rajnish Jain)
             Note: Following the tradition of naming after alcohols.
YOMOMMA   - Yet anOther Memory archive fOr MultiMediA (James Polk)

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From jmpolk@cisco.com  Thu Feb 11 15:52:02 2010
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At 01:54 PM 2/11/2010, Vijay K. Gurbani wrote:
>Cullen: Now that the world has proof-positive of the
>unbounded intellect of DISPATCHers in conjuring
>up creative acronyms at short notice, here is a list of
>entrants so far.  Your executive decision now is to
>pick one of these or come up with a new one of your
>own choosing.
>
>The guilty party of each acronym is clearly identified
>for recrimination (or renumeration) at some later time.
>
>MARE      - Multimedia Assets REcording (Tobia Castaldi)
>             Note: Italian word for "sea"
>USAID     - Unified Session Archival In Datastore (Scott Lawrence)
>IMS       - Intermediate Media Storage (Christer Holmberg)
>NEVERMORE - Network Enabled diVERsion of Multimedia On
>             Recording Equipment (Adam Roach)
>RECON     - recording control (Leon Portman)
>REX       - recording extensions (Leon Portman)
>INTER     - Interactions Recording (Leon Portman)
>MMREC     - multimedia recording (Leon Portman)
>RUM       - Recording of Unified Media (Leon Portman)
>             Note: Following the tradition of naming after alcohols.
>SRIP      - Session Recording Initiation Protocol (Steve LoGalbo)
>SRI       - Session Recording Interface? (Steve LoGalbo)
>IPSR      - IP Session Recording? (Steve LoGalbo)
>UCR / UMR - Unified Communication Recording / Unified Media Recording
>             (Steve LoGalbo)
>DRUNK     - Dynamic Recording of UNified Kommunications (Steve LoGalbo)
>             Note: Following the tradition of naming after alcohols.
>MRSIP     - Media Recording using the Session Initiation Protocol
>            (Andrew Hutton)
>YAAD      - Yet Another Acoustic Diverter (Vijay K. Gurbani)
>             Note: Hindi word for "memory".
>SINARP    - Sinarp Is Not A Recording Protocol (Lorenzo Miniero)
>MARGARITA - MediA RecordinG And Replay In The internet Architecture
>             (Rajnish Jain)
>             Note: Following the tradition of naming after alcohols.
>YOMOMMA   - Yet anOther Memory archive fOr MultiMediA (James Polk)

this spelling also works...

YOMAMMA  - Yet anOther Memory Archive for MultiMediA (James Polk)

I'm trying to come up with an explosion for HAMMERED, which would be 
really cool IMO


>- vijay
>--
>Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
>Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
>Web:   http://ect.bell-labs.com/who/vkg/
>_______________________________________________
>dispatch mailing list
>dispatch@ietf.org
>https://www.ietf.org/mailman/listinfo/dispatch


From dean.willis@softarmor.com  Thu Feb 11 16:20:24 2010
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On Feb 10, 2010, at 3:29 PM, Cullen Jennings wrote:

>
> In IESG review, it got pointed out this name conflicts with SRP  
> (Securer Remote Password) in the security context. Could folks  
> propose some other name for the WG be early next week.
>

I like "SIPREC". It's kindof like "shipwreck", which is a good  
semantic match.

--
Dean


From fluffy@cisco.com  Thu Feb 11 18:17:20 2010
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Wow - I laughed when I read this. We really need a "best of the IETF" =
mailing list.  I have no clue what to do. Perhaps I can fob it off to =
whoever NomCom chooses to replace me. I'll figure something out from =
these. Thanks,=20

Cullen

PS.  I'll have to add to the list

RAVEN - Recording Audio and Video Everywhere in the Network (Cullen)


On Feb 11, 2010, at 12:54 PM, Vijay K. Gurbani wrote:

> Cullen: Now that the world has proof-positive of the
> unbounded intellect of DISPATCHers in conjuring
> up creative acronyms at short notice, here is a list of
> entrants so far.  Your executive decision now is to
> pick one of these or come up with a new one of your
> own choosing.
>=20
> The guilty party of each acronym is clearly identified
> for recrimination (or renumeration) at some later time.
>=20
> MARE      - Multimedia Assets REcording (Tobia Castaldi)
>            Note: Italian word for "sea"
> USAID     - Unified Session Archival In Datastore (Scott Lawrence)
> IMS       - Intermediate Media Storage (Christer Holmberg)
> NEVERMORE - Network Enabled diVERsion of Multimedia On
>            Recording Equipment (Adam Roach)
> RECON     - recording control (Leon Portman)
> REX       - recording extensions (Leon Portman)
> INTER     - Interactions Recording (Leon Portman)
> MMREC     - multimedia recording (Leon Portman)
> RUM       - Recording of Unified Media (Leon Portman)
>            Note: Following the tradition of naming after alcohols.
> SRIP      - Session Recording Initiation Protocol (Steve LoGalbo)
> SRI       - Session Recording Interface? (Steve LoGalbo)
> IPSR      - IP Session Recording? (Steve LoGalbo)
> UCR / UMR - Unified Communication Recording / Unified Media Recording
>            (Steve LoGalbo)
> DRUNK     - Dynamic Recording of UNified Kommunications (Steve =
LoGalbo)
>            Note: Following the tradition of naming after alcohols.
> MRSIP     - Media Recording using the Session Initiation Protocol
>           (Andrew Hutton)
> YAAD      - Yet Another Acoustic Diverter (Vijay K. Gurbani)
>            Note: Hindi word for "memory".
> SINARP    - Sinarp Is Not A Recording Protocol (Lorenzo Miniero)
> MARGARITA - MediA RecordinG And Replay In The internet Architecture
>            (Rajnish Jain)
>            Note: Following the tradition of naming after alcohols.
> YOMOMMA   - Yet anOther Memory archive fOr MultiMediA (James Polk)
>=20
> - vijay
> --=20
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
> Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
> Web:   http://ect.bell-labs.com/who/vkg/


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From dean.willis@softarmor.com  Thu Feb 11 20:20:45 2010
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On Feb 11, 2010, at 8:18 PM, Cullen Jennings wrote:

>
> Wow - I laughed when I read this. We really need a "best of the  
> IETF" mailing list.  I have no clue what to do. Perhaps I can fob it  
> off to whoever NomCom chooses to replace me. I'll figure something  
> out from these. Thanks,
>
> Cullen
>
> PS.  I'll have to add to the list
>
> RAVEN - Recording Audio and Video Everywhere in the Network (Cullen)
>


On further reflection, I have to support NEVERMORE, as a logical  
extension from RAVEN.

 From Poe:

http://en.wikisource.org/wiki/The_Raven_(Poe)


--
Dean

From gonzalo.camarillo@ericsson.com  Sat Feb 13 23:26:39 2010
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Subject: [dispatch] [Fwd:  SIP OPTIONS rework proposal]
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Hi,

speaking as SIPCORE chair this time, the SIPCORE WG has received an
agenda request to discuss this topic in Anaheim. Given that OPTIONS
requests are specified in RFC 3261, it sounds reasonable for SIPCORE to
have an initial look at what to do with this. So, if you have comments
about this topic, you may want to discuss them on the SIPCORE list.

Thanks,

Gonzalo


-------- Original Message --------
Subject: 	[dispatch] SIP OPTIONS rework proposal
Date: 	Thu, 21 Jan 2010 15:31:36 +0100
From: 	Salvatore Loreto <salvatore.loreto@ericsson.com>
To: 	dispatch@ietf.org <dispatch@ietf.org>



Hi there,

there as been a short discussion in SIP Core ml
about the fact that the OPTIONS method to discovery capabilities is
pretty broken.

As discovery capabilities is a meaningful mechanism to provide enhanced
services,
it would be important, at least in my opinion, to do some serious rework
on it!

Below an initial description of the problem as derived from the exchange
of mails with Paul.

Comments are very welcome!

Regards
Sal

----------



OPTIONS as discovery capabilities mechanism
=============================

The SIP method OPTIONS allows you to query the capabilities of another
UA or a proxy server.
The method provides a way to discover information about the supported
methods,
content types, extensions, codecs, etc. without "ringing" the other party.

The target of the OPTIONS request is identified by the Request-URI,
which could identify another UA
or a SIP server. However SIP OPTIONS also inherits the "traceroute"
behaviour from HTTP/1.1, where
a server receiving an OPTIONS request with a Max-Forwards header field
value of 0 MAY respond
to the request regardless of the Request-URI.
So the original intent and design of OPTIONS seems to be that whoever
responds to an OPTIONS method
does it for itself.

However in the case the request is addressed to an AoR, and there is a
proxy responsible for that AoR
(that will typically route requests via contact routing), it is not
clear the right behaviour of the proxy.

    e.g. Alice sends a SIP OPTIONS request to discover the Bob
    capabilities; if Bob has registered two or more
           UAs, then the proxy responsible for the Bob AoR forks the
    OPTIONS request letting it arrive to all the
           registered UAs. A possible behavior for the proxy responsible
    for the Bob AoR may be to send back
           to Alice only the first response it receives. However this is
    just a possible behaviour, others are also
           possible or at least not prohibited!


A nice thing to be able to learn, through a discovery capabilities
mechanism, would be the full set of capabilities
associated to the different UAs an user has registered and not just the
capabilities of the UA that answers more quickly.
The question is whether OPTIONS is the way to achieve it, or it would
make more sense to leave OPTIONS alone
and create something new.



Moreover the assumption that the capabilities of a UA are stable over
time and are context free is also pretty broken.
UA capabilities can change for a lot of different reasons: a user switch
off/on one of his registered UAs; or context changed
than when the query has been received.

    For instance, if Bob's device is handling a call when it receives
    the query, does the response reflect what
    it can do concurrently with the call, or what it will be able to do
    once the call had ended?


So the description of the context and of the reasons that can let
capabilities to chance over time could provide useful
pieces of information.





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_______________________________________________
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>, "Cullen Jennings" <fluffy@cisco.com>
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=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Vijay K. Gurbani
> YAAD      - Yet Another Acoustic Diverter (Vijay K. Gurbani)
>              Note: Hindi word for "memory".

.... and Hebrew for target / destination :-)

From pkyzivat@cisco.com  Sun Feb 14 12:58:12 2010
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Cc: Mary Barnes <mbarnes@nortelnetworks.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Subject: [dispatch] Preliminary version of Expert review of draft-avasarala-dispatch-comm-div-notification-03
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I was requested to do an expert review of
draft-avasarala-dispatch-comm-div-notification-03.txt

Below I have reproduced the requirements from
http://tools.ietf.org/html/draft-peterson-rai-rfc3427bis-04#section-4.1
and the referenced requirements from RFC 3265, and interleaved my 
comments to them.

I did identify a number of issues - more than I expected to.
Most arose from carefully checking compliance to 3265.

I've tagged things that should specifically be fixed with ISSUE and NIT.

	Thanks,
	Paul

>    1.  A Designated Expert (as defined in RFC5226) must review the
>        proposal for applicability to SIP and conformance with these
>        guidelines.  The Designated Expert will send email to the IESG on
>        this determination.  The expert reviewer can cite one or more of
>        the guidelines that have not been followed in his/her opinion.

I'm doing that below.

>    2.  The proposed extension MUST NOT define an event template-package.

It does not.

>    3.  The function of the proposed package MUST NOT overlap with
>        current or planned chartered packages.

I am aware of no chartered work on this subject.
The draft itself has been debated at length on the dispatch wg mailing
list, but with a relatively small number of people actively involved.
Some, including me, have stated that a service resembling the one
proposed might be of general utility. But there have been no volunteers
to do the work of generalizing this. So I conclude that there are
currently no plans for chartered work on such a package.

>    4.  The event package MUST NOT redefine or contradict the normative
>        behavior of SIP events [RFC3265], SIP [RFC3261], or related
>        standards track extensions.  (See Section 4)

I did not identify any issues in this regard.

>    5.  The proposed package MUST NOT undermine SIP security in any
>        sense.  The Internet Draft proposing the new package MUST address
>        security issues in detail as if it were a Standards Track
>        document.  Security issues need to be discussed for arbitrary
>        usage scenarios (including the general Internet case).

The document identifies the specific security issues presented by this
package. It specifies that authentication and authorization must be
done, but leaves the means for authorization out of scope. While a bit
thin, this may be sufficient for the area of applicability.

>    6.  The proposed package MUST be clearly documented in an
>        (Individual) Informational RFC, and registered with IANA. 

The document under review will satisfy this requirement.

>        The package MUST document all the package considerations required
>        in Section 5 of SIP events [RFC3265].

(I believe this guideline has an incorrect reference, and should
reference section *4* of 3265. I'm treating it that way.)

> 4.1. Appropriateness of Usage

In my opinion this is an entirely appropriate use of sip and event packages.

> 4.2. Event Template-packages

N/A - template package not defined.

> 4.3. Amount of State to be Conveyed

> 4.3.1. Complete State Information
> 4.3.2. State Deltas

ISSUE: The document details the notification body, and some inferences
can be drawn about the state from the syntax of the notification body.
But this is tenuous. There should be an explicit specification of the
state maintained, and how it is mapped onto the notification body.

> 4.4. Event Package Responsibilities
> 4.4.1. Event Package Name

OK.

> 4.4.2. Event Package Parameters

ISSUE: There is a section for this, but it repeats the description of
the event package name rather than specifying the (lack of) event
parameters.

This seems to be a simple oversight/typo, but needs to be corrected.

> 4.4.3. SUBSCRIBE Bodies

The document specifies that a body MAY be present.
It then documents the semantics when such body is of type
"application/comm-div-info+xml".


ISSUE: The subscribe body is intended to be used as a filter on which
events/state are reported. The subscribe body contains data to be used
in the filtering process, but the decision process for filtering is not
specified. For instance, both a User-name and a User-URI may be
provided. It is implied, rather than stated, that the User-name field is
matched against the user portion of some URI, whereas the User-URI field
is matched against a complete URI. Also, there are a variety of field
types, and no indication how matches of multiple fields are combined.
(I.e. AND or OR.)

Its possible that all of that is defined by the CDIV service in some IMS
document. If so, rather than a generic reference, there should be
explicit references to that to define where this sort of semantic is
defined.

ISSUE: The document is a bit confusing because the same MIME-type and
XML syntax are used for the subscribe body and the notify body. Yet
there are specific portions of this syntax specific to subscriber bodies
and notify bodies. For instance it isn't clear if the filter information
given in a subscribe body is to be replayed in the notification body.
Nor is there any specification of what should happen if notifier
information is present in the subscribe body.

ISSUE?: Semantics not specified if a body of some other type is present.
This provides an unspecified extensibility hook that could be employed
without any revision to the documented behavior of the package. I'm
uncertain if this is appropriate or not.

> 4.4.4. Subscription Duration
> 
>    It is RECOMMENDED that event packages give a suggested range of times
>    considered reasonable for the duration of a subscription.  Such
>    packages MUST also define a default "Expires" value to be used if
>    none is specified.

No suggested or default value is given, nor is any reason given for not
suggesting a value. Text puts this to the discretion of the subscriber.

ISSUE: The document needs to be revised to provide a default value.

ISSUE: It also needs to either:
- provide a suggested value, or
- give a rationale for why a suggested value is not given

> 4.4.5. NOTIFY Bodies

Syntax is well specified.

Semantics is barely specified.

NIT: diversion-time-info does not specify a time zone, nor does it state
that it is in some specific zone such as UTC. This makes it difficult
for the subscriber to interpret. Perhaps not important if the
notification is delivered when the diversion occurs. But if an initial
subscribe can report a diversion that occurred in the past it could be a
problem. (Perhaps this is a typo - earlier specifications of time
include the zone. But the example shows usage without a zone.)

DISCLAIMER: I'm not qualified to verify the correctness of the XML
schema definition or the XML example.

> 4.4.6. Notifier processing of SUBSCRIBE requests

Nominal but perhaps sufficient.

> 4.4.7. Notifier generation of NOTIFY requests
>
>    This section of an event package describes the process by which the
>    notifier generates and sends a NOTIFY request.  This includes
>    detailed information about what events cause a NOTIFY to be sent,

This is covered.

>    how to compute the state information in the NOTIFY,

ISSUE: The sending of events is not tied directly to the state of the
resource.

ISSUE: Questionable. Document says that the NOTIFY *MAY* contain a body.
But doesn't specify when it does and when it doesn't.

>    how to generate
>    neutral or fake state information to hide authorization delays and
>    decisions from users, and whether state information is complete or
>    deltas for notifications; see section 4.3.  Such a section is
>    required.

ISSUE: The distinction between complete and delta state information is
not addressed, and is ambiguous. Its unclear whether a history of prior
diversions is retained and delivered, or if the state consists solely of
a diversion as it is happening, that then immediately disappears from
state after the notification.

> 4.4.8. Subscriber processing of NOTIFY requests

ISSUE: This section is very thin - it specifies nothing.
Perhaps there are no requirements in this regard.

If the state were better specified, then this section would probably
specify how the state is reconstructed based on the notifications.

> 4.4.9. Handling of forked requests

OK, not permitted.

> 4.4.10.  Rate of notifications

OK - specified.

> 4.4.11.  State Agents

OK - not required or expected.

> 4.4.12.  Examples

ISSUE: The only provided example is a sample of a notification body.
Flow diagrams and message detail, as called out in 3265, would be very
useful.

NIT: Section 8.2 is entitled "Sample Notification Body", but it contains
subsections entitled "Instance of communication diversion subscription
document" and "Instance of communication diversion notification
document". While the latter would be a notification body, the former
would presumably be a subscription body.

I'll observe that the examples suggest that comm-div-subs-info from the
subscribe body is *not* replayed in the notify body. But there is no
normative information about that.

> 4.4.13.  Use of URIs to Retrieve State

OK - not used.

>    7.  If determined by the Designated Expert or the chairs or ADs of
>        the DISPATCH WG, an applicability statement in the Informational
>        RFC MUST clearly document the useful scope of the proposal, and
>        explain its limitations and why it is not suitable for the
>        general use of SIP in the Internet.

The Applicability Statement in the document adequately documents the scope.

ISSUE?: It doesn't explain the limitations, or why the package hasn't
been designed for more general use.



From fluffy@cisco.com  Thu Feb 18 12:55:35 2010
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This charter got approved to go out for IETF LC so that should happen =
some time next week.

Cullen





Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From keith.drage@alcatel-lucent.com  Fri Feb 19 04:12:22 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: DISPATCH <dispatch@ietf.org>
Date: Fri, 19 Feb 2010 13:13:58 +0100
Thread-Topic: Comment on draft-liess-dispatch-alert-info-urns-00
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Subject: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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I was looking for the information on how this changed the Alert-Info header=
 field usage from RFC 3261 and did not find it.

RFC 3261 defines the usage of this header field only for INVITE requests.

While this draft gives examples, and therefore implies it is valid for some=
 responses, it is not clear whether that extension now extends beyond INVIT=
E, is all INVITE responses, or is limited to 180 responses. Could we have s=
ome more normative type statements in this regard.

Oh, and I'd actually like to know what the answer is!

regards

Keith


From L.Liess@telekom.de  Fri Feb 19 05:55:00 2010
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Subject: Re: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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Hi Keith,=20

We are just working on a new version of the draft. There are some
inconsistencies in the 00 version.=20
The draft just should define additional identifiers which can be sent in
the Alert-Info header. It does not change the header usage.=20

I hope things will be more clear in the new version. E.g. all "busy"-
stuff was deleted. We also distinguish between ring tones and ringback
tones and state that ring tones can be sent only in INVITE , while
ringback tones can be sent only in 180-ringing.=20

Kind regards
Laura =20




> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of DRAGE, Keith (Keith)
> Sent: Friday, February 19, 2010 1:14 PM
> To: DISPATCH
> Subject: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
>=20
> I was looking for the information on how this changed the=20
> Alert-Info header field usage from RFC 3261 and did not find it.
>=20
> RFC 3261 defines the usage of this header field only for=20
> INVITE requests.
>=20
> While this draft gives examples, and therefore implies it is=20
> valid for some responses, it is not clear whether that=20
> extension now extends beyond INVITE, is all INVITE responses,=20
> or is limited to 180 responses. Could we have some more=20
> normative type statements in this regard.
>=20
> Oh, and I'd actually like to know what the answer is!
>=20
> regards
>=20
> Keith
>=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>=20

From keith.drage@alcatel-lucent.com  Fri Feb 19 06:05:13 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "L.Liess@telekom.de" <L.Liess@telekom.de>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 19 Feb 2010 15:06:55 +0100
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So the header usage in RFC 3261 is INVITE requests and 180 responses to INV=
ITE requests only. That remains the case? Perhaps a single informative stat=
ement regarding the header field usage is unchanged from RFC 3261 might be =
appropriate, unless that is already covered elsewhere (I have not done a li=
ne by line scan).

regards

Keith



> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of L.Liess@telekom.de
> Sent: Friday, February 19, 2010 1:57 PM
> To: DRAGE, Keith (Keith); dispatch@ietf.org
> Subject: Re: [dispatch] Comment on=20
> draft-liess-dispatch-alert-info-urns-00
>=20
> Hi Keith,=20
>=20
> We are just working on a new version of the draft. There are=20
> some inconsistencies in the 00 version.=20
> The draft just should define additional identifiers which can=20
> be sent in the Alert-Info header. It does not change the=20
> header usage.=20
>=20
> I hope things will be more clear in the new version. E.g. all=20
> "busy"- stuff was deleted. We also distinguish between ring=20
> tones and ringback tones and state that ring tones can be=20
> sent only in INVITE , while ringback tones can be sent only=20
> in 180-ringing.=20
>=20
> Kind regards
> Laura =20
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of DRAGE, Keith (Keith)
> > Sent: Friday, February 19, 2010 1:14 PM
> > To: DISPATCH
> > Subject: [dispatch] Comment on=20
> draft-liess-dispatch-alert-info-urns-00
> >=20
> > I was looking for the information on how this changed the=20
> Alert-Info=20
> > header field usage from RFC 3261 and did not find it.
> >=20
> > RFC 3261 defines the usage of this header field only for INVITE=20
> > requests.
> >=20
> > While this draft gives examples, and therefore implies it=20
> is valid for=20
> > some responses, it is not clear whether that extension now extends=20
> > beyond INVITE, is all INVITE responses, or is limited to 180=20
> > responses. Could we have some more normative type=20
> statements in this=20
> > regard.
> >=20
> > Oh, and I'd actually like to know what the answer is!
> >=20
> > regards
> >=20
> > Keith
> >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From L.Liess@telekom.de  Fri Feb 19 06:29:44 2010
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From: <L.Liess@telekom.de>
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Subject: Re: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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Keith,=20

I think an informative statement regarding the header field usage is
unchanged from RFC 3261 is a good idea. I will look for a place for it.=20

I am a little confused by the text " to INVITE requests only." in your
first sentence. What did you mean with it?=20

Kind regards
Laura

=20

=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]=20
> Sent: Friday, February 19, 2010 3:07 PM
> To: Liess, Laura; dispatch@ietf.org
> Subject: RE: [dispatch] Comment on=20
> draft-liess-dispatch-alert-info-urns-00
>=20
> So the header usage in RFC 3261 is INVITE requests and 180=20
> responses to INVITE requests only. That remains the case?=20
> Perhaps a single informative statement regarding the header=20
> field usage is unchanged from RFC 3261 might be appropriate,=20
> unless that is already covered elsewhere (I have not done a=20
> line by line scan).
>=20
> regards
>=20
> Keith
>=20
>=20
>=20
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org=20
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of L.Liess@telekom.de
> > Sent: Friday, February 19, 2010 1:57 PM
> > To: DRAGE, Keith (Keith); dispatch@ietf.org
> > Subject: Re: [dispatch] Comment on=20
> > draft-liess-dispatch-alert-info-urns-00
> >=20
> > Hi Keith,=20
> >=20
> > We are just working on a new version of the draft. There are=20
> > some inconsistencies in the 00 version.=20
> > The draft just should define additional identifiers which can=20
> > be sent in the Alert-Info header. It does not change the=20
> > header usage.=20
> >=20
> > I hope things will be more clear in the new version. E.g. all=20
> > "busy"- stuff was deleted. We also distinguish between ring=20
> > tones and ringback tones and state that ring tones can be=20
> > sent only in INVITE , while ringback tones can be sent only=20
> > in 180-ringing.=20
> >=20
> > Kind regards
> > Laura =20
> >=20
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org
> > > [mailto:dispatch-bounces@ietf.org] On Behalf Of DRAGE,=20
> Keith (Keith)
> > > Sent: Friday, February 19, 2010 1:14 PM
> > > To: DISPATCH
> > > Subject: [dispatch] Comment on=20
> > draft-liess-dispatch-alert-info-urns-00
> > >=20
> > > I was looking for the information on how this changed the=20
> > Alert-Info=20
> > > header field usage from RFC 3261 and did not find it.
> > >=20
> > > RFC 3261 defines the usage of this header field only for INVITE=20
> > > requests.
> > >=20
> > > While this draft gives examples, and therefore implies it=20
> > is valid for=20
> > > some responses, it is not clear whether that extension=20
> now extends=20
> > > beyond INVITE, is all INVITE responses, or is limited to 180=20
> > > responses. Could we have some more normative type=20
> > statements in this=20
> > > regard.
> > >=20
> > > Oh, and I'd actually like to know what the answer is!
> > >=20
> > > regards
> > >=20
> > > Keith
> > >=20
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> > >=20
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >=20
>=20

From keith.drage@alcatel-lucent.com  Fri Feb 19 07:03:40 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "L.Liess@telekom.de" <L.Liess@telekom.de>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Fri, 19 Feb 2010 16:05:17 +0100
Thread-Topic: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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Subject: Re: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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Maybe I was being too precise but the words "180 responses to INVITE reques=
ts" needs to be read as a whole, i.e. it is a 160 response generated as a r=
esult of receiving an INVITE request.

Keith
=20

> -----Original Message-----
> From: L.Liess@telekom.de [mailto:L.Liess@telekom.de]=20
> Sent: Friday, February 19, 2010 2:30 PM
> To: DRAGE, Keith (Keith); dispatch@ietf.org
> Subject: RE: [dispatch] Comment on=20
> draft-liess-dispatch-alert-info-urns-00
>=20
>=20
> Keith,=20
>=20
> I think an informative statement regarding the header field=20
> usage is unchanged from RFC 3261 is a good idea. I will look=20
> for a place for it.=20
>=20
> I am a little confused by the text " to INVITE requests=20
> only." in your first sentence. What did you mean with it?=20
>=20
> Kind regards
> Laura
>=20
> =20
>=20
> =20
>=20
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> > Sent: Friday, February 19, 2010 3:07 PM
> > To: Liess, Laura; dispatch@ietf.org
> > Subject: RE: [dispatch] Comment on
> > draft-liess-dispatch-alert-info-urns-00
> >=20
> > So the header usage in RFC 3261 is INVITE requests and 180=20
> responses=20
> > to INVITE requests only. That remains the case?
> > Perhaps a single informative statement regarding the header field=20
> > usage is unchanged from RFC 3261 might be appropriate,=20
> unless that is=20
> > already covered elsewhere (I have not done a line by line scan).
> >=20
> > regards
> >=20
> > Keith
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: dispatch-bounces@ietf.org
> > > [mailto:dispatch-bounces@ietf.org] On Behalf Of L.Liess@telekom.de
> > > Sent: Friday, February 19, 2010 1:57 PM
> > > To: DRAGE, Keith (Keith); dispatch@ietf.org
> > > Subject: Re: [dispatch] Comment on
> > > draft-liess-dispatch-alert-info-urns-00
> > >=20
> > > Hi Keith,
> > >=20
> > > We are just working on a new version of the draft. There are some=20
> > > inconsistencies in the 00 version.
> > > The draft just should define additional identifiers which can be=20
> > > sent in the Alert-Info header. It does not change the=20
> header usage.
> > >=20
> > > I hope things will be more clear in the new version. E.g. all
> > > "busy"- stuff was deleted. We also distinguish between ring tones=20
> > > and ringback tones and state that ring tones can be sent only in=20
> > > INVITE , while ringback tones can be sent only in 180-ringing.
> > >=20
> > > Kind regards
> > > Laura
> > >=20
> > >=20
> > >=20
> > >=20
> > > > -----Original Message-----
> > > > From: dispatch-bounces@ietf.org
> > > > [mailto:dispatch-bounces@ietf.org] On Behalf Of DRAGE,
> > Keith (Keith)
> > > > Sent: Friday, February 19, 2010 1:14 PM
> > > > To: DISPATCH
> > > > Subject: [dispatch] Comment on
> > > draft-liess-dispatch-alert-info-urns-00
> > > >=20
> > > > I was looking for the information on how this changed the
> > > Alert-Info
> > > > header field usage from RFC 3261 and did not find it.
> > > >=20
> > > > RFC 3261 defines the usage of this header field only for INVITE=20
> > > > requests.
> > > >=20
> > > > While this draft gives examples, and therefore implies it
> > > is valid for
> > > > some responses, it is not clear whether that extension
> > now extends
> > > > beyond INVITE, is all INVITE responses, or is limited to 180=20
> > > > responses. Could we have some more normative type
> > > statements in this
> > > > regard.
> > > >=20
> > > > Oh, and I'd actually like to know what the answer is!
> > > >=20
> > > > regards
> > > >=20
> > > > Keith
> > > >=20
> > > > _______________________________________________
> > > > dispatch mailing list
> > > > dispatch@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/dispatch
> > > >=20
> > > _______________________________________________
> > > dispatch mailing list
> > > dispatch@ietf.org
> > > https://www.ietf.org/mailman/listinfo/dispatch
> > >=20
> >=20
> =

From L.Liess@telekom.de  Fri Feb 19 07:15:11 2010
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thread-topic: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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From: <L.Liess@telekom.de>
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X-OriginalArrivalTime: 19 Feb 2010 15:16:35.0439 (UTC) FILETIME=[8681C7F0:01CAB176]
Subject: Re: [dispatch] Comment on draft-liess-dispatch-alert-info-urns-00
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Fine, thanks.

Laura


> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]=20
> Sent: Friday, February 19, 2010 4:05 PM
> To: Liess, Laura; dispatch@ietf.org
> Subject: RE: [dispatch] Comment on=20
> draft-liess-dispatch-alert-info-urns-00
>=20
> Maybe I was being too precise but the words "180 responses to=20
> INVITE requests" needs to be read as a whole, i.e. it is a=20
> 160 response generated as a result of receiving an INVITE request.
>=20
> Keith
> =20
>=20
> > -----Original Message-----
> > From: L.Liess@telekom.de [mailto:L.Liess@telekom.de]=20
> > Sent: Friday, February 19, 2010 2:30 PM
> > To: DRAGE, Keith (Keith); dispatch@ietf.org
> > Subject: RE: [dispatch] Comment on=20
> > draft-liess-dispatch-alert-info-urns-00
> >=20
> >=20
> > Keith,=20
> >=20
> > I think an informative statement regarding the header field=20
> > usage is unchanged from RFC 3261 is a good idea. I will look=20
> > for a place for it.=20
> >=20
> > I am a little confused by the text " to INVITE requests=20
> > only." in your first sentence. What did you mean with it?=20
> >=20
> > Kind regards
> > Laura
> >=20
> > =20
> >=20
> > =20
> >=20
> > > -----Original Message-----
> > > From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> > > Sent: Friday, February 19, 2010 3:07 PM
> > > To: Liess, Laura; dispatch@ietf.org
> > > Subject: RE: [dispatch] Comment on
> > > draft-liess-dispatch-alert-info-urns-00
> > >=20
> > > So the header usage in RFC 3261 is INVITE requests and 180=20
> > responses=20
> > > to INVITE requests only. That remains the case?
> > > Perhaps a single informative statement regarding the header field=20
> > > usage is unchanged from RFC 3261 might be appropriate,=20
> > unless that is=20
> > > already covered elsewhere (I have not done a line by line scan).
> > >=20
> > > regards
> > >=20
> > > Keith
> > >=20
> > >=20
> > >=20
> > > > -----Original Message-----
> > > > From: dispatch-bounces@ietf.org
> > > > [mailto:dispatch-bounces@ietf.org] On Behalf Of=20
> L.Liess@telekom.de
> > > > Sent: Friday, February 19, 2010 1:57 PM
> > > > To: DRAGE, Keith (Keith); dispatch@ietf.org
> > > > Subject: Re: [dispatch] Comment on
> > > > draft-liess-dispatch-alert-info-urns-00
> > > >=20
> > > > Hi Keith,
> > > >=20
> > > > We are just working on a new version of the draft.=20
> There are some=20
> > > > inconsistencies in the 00 version.
> > > > The draft just should define additional identifiers=20
> which can be=20
> > > > sent in the Alert-Info header. It does not change the=20
> > header usage.
> > > >=20
> > > > I hope things will be more clear in the new version. E.g. all
> > > > "busy"- stuff was deleted. We also distinguish between=20
> ring tones=20
> > > > and ringback tones and state that ring tones can be=20
> sent only in=20
> > > > INVITE , while ringback tones can be sent only in 180-ringing.
> > > >=20
> > > > Kind regards
> > > > Laura
> > > >=20
> > > >=20
> > > >=20
> > > >=20
> > > > > -----Original Message-----
> > > > > From: dispatch-bounces@ietf.org
> > > > > [mailto:dispatch-bounces@ietf.org] On Behalf Of DRAGE,
> > > Keith (Keith)
> > > > > Sent: Friday, February 19, 2010 1:14 PM
> > > > > To: DISPATCH
> > > > > Subject: [dispatch] Comment on
> > > > draft-liess-dispatch-alert-info-urns-00
> > > > >=20
> > > > > I was looking for the information on how this changed the
> > > > Alert-Info
> > > > > header field usage from RFC 3261 and did not find it.
> > > > >=20
> > > > > RFC 3261 defines the usage of this header field only=20
> for INVITE=20
> > > > > requests.
> > > > >=20
> > > > > While this draft gives examples, and therefore implies it
> > > > is valid for
> > > > > some responses, it is not clear whether that extension
> > > now extends
> > > > > beyond INVITE, is all INVITE responses, or is limited to 180=20
> > > > > responses. Could we have some more normative type
> > > > statements in this
> > > > > regard.
> > > > >=20
> > > > > Oh, and I'd actually like to know what the answer is!
> > > > >=20
> > > > > regards
> > > > >=20
> > > > > Keith
> > > > >=20
> > > > > _______________________________________________
> > > > > dispatch mailing list
> > > > > dispatch@ietf.org
> > > > > https://www.ietf.org/mailman/listinfo/dispatch
> > > > >=20
> > > > _______________________________________________
> > > > dispatch mailing list
> > > > dispatch@ietf.org
> > > > https://www.ietf.org/mailman/listinfo/dispatch
> > > >=20
> > >=20
> >=20
>=20

From gmoczar@gmail.com  Fri Feb 19 09:55:08 2010
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From: Gabor Moczar <gmoczar@gmail.com>
To: dispatch@ietf.org
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Subject: [dispatch] Comments on Session Recording Protocol
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--001636458ca4a0e18f047ff7cead
Content-Type: text/plain; charset=ISO-8859-1

Dear Working Group Members,

I am new to this list and Leon Portman from Nice suggested to send my
comments on the Session Recording Protocol to here directly.
I have 8+ years experience with IP based call recording solutions as a
solution architect and product manager.
Let me know your opinion about my comments.

1. Meta data
SRP should provide a standard mechanism, which allows sending meta data in
addition to the media to the RS. An extendible set of fields could describe
the recorded call in a detailed way including calling party number, name,
agent id, etc.
Currently most implementations require implementing an additional interface
using standard or proprietary CTI to obtain such kind of information. These
implementations quite unique across the various platforms and considered to
add additional complexity, OAM overhead and failure point to the overall
solution.

2. Handling multiple media streams
SRP should allow RS to select, which media stream should be recorded or not
for a given session.
As unified communication emerge, more and more customers using voice, video,
instant messaging, etc. in their communication with their partners and
clients. Customers may would like to decide, which media or which direction
of the media should be recorded during the session. For compliance, probably
only the voice part of a video conversation is sufficient too.

3. Security and encryption
The draft very well describes the sensitivity of the topic from security
point of view, however the term "encryption" is missing from the
requirements part. I see that "eavesdropping protection" is used, but what
is the purpose of not using the most obvious solution for "eavesdropping
protection".
I would also add: SRP should also allow the recording of encrypted calls.

4. Redundancy
SRP should provide a mechanism to allow RCs to send the media to multiple
RSs at one time.
Probably endpoints with limited resources or networks with restricted
bandwidth will not able to provide this feature, but in mission critical
environments this could be a very nice way to provide redundancy. The
failover mechanism described in the draft (and used by some of the vendors)
is not protected from single point of failure.

5. Negotiating media handling capabilities
SRP should allow RC and RS to negotiate audio/video codec capabilities
(similar to SIP/SDP) and the RC should send media in the agreed codec.
In most cases the codec will be exactly the same as the one used in the
recorded session. However, high definition audio and video codecs are
getting used in more and more sessions, for recording, the requirement in
most cases allow to record the sessions in a lower resolution format. Either
the RC, either the RS should provide transcoding capabilities. I know
several video endpoints, which are able to send the video in 2 formats at
one time (HD and CIF) and I can also imagine devices with on-board
transcoding capabilities in the future (e.g. transcoding AAC-LD to G.729 for
the RS) and some of the current implementations also allow the transcoding
of the recording streams using network resources.

6. Others
There are some points, which are mentioned in the use cases section, but no
requirement is provided to support the given feature:
- Silent monitoring and real-time applications: the entire mechanism should
allow the users to implement a real-time monitoring functionality, which
means that the media has to be transmitted with considerably low delay, in a
real-time fashion to the RS.
- Handling complex calls: SRP should provide a mechanism to allow the RS to
link together certain call segments in a complex call scenario (e.g.
providing a global call identifier as a part of the meta data).

Best regards,
Gabor

--001636458ca4a0e18f047ff7cead
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Dear Working Group Members,</div><div><br></div><div>I am new to this =
list and Leon Portman from Nice suggested to send my comments on the Sessio=
n Recording Protocol to here directly.</div><div>I have 8+ years experience=
 with IP based call recording solutions as a solution architect and product=
 manager.</div>
<div>Let me know your opinion about my comments.</div><div><br></div><div>1=
. Meta data=A0</div><div>SRP should provide a standard mechanism, which all=
ows sending meta data in addition to the media to the RS. An extendible set=
 of fields could describe the recorded call in a detailed way including cal=
ling party number, name, agent id, etc.=A0</div>
<div>Currently most implementations require implementing an additional inte=
rface using standard or proprietary CTI to obtain such kind of information.=
 These implementations quite unique across the various platforms and consid=
ered to add additional complexity, OAM overhead and failure point to the ov=
erall solution.=A0</div>
<div><br></div><div>2. Handling multiple media streams=A0</div><div>SRP sho=
uld allow RS to select, which media stream should be recorded or not for a =
given session.=A0</div><div>As unified communication emerge, more and more =
customers using voice, video, instant messaging, etc. in their communicatio=
n with their partners and clients. Customers may would like to decide, whic=
h media or which direction of the media should be recorded during the sessi=
on. For compliance, probably only the voice part of a video conversation is=
 sufficient too.=A0</div>
<div><br></div><div>3. Security and encryption=A0</div><div>The draft very =
well describes the sensitivity of the topic from security point of view, ho=
wever the term &quot;encryption&quot; is missing from the requirements part=
. I see that &quot;eavesdropping protection&quot; is used, but what is the =
purpose of not using the most obvious solution for &quot;eavesdropping prot=
ection&quot;.=A0</div>
<div>I would also add: SRP should also allow the recording of encrypted cal=
ls.=A0</div><div><br></div><div>4. Redundancy=A0</div><div>SRP should provi=
de a mechanism to allow RCs to send the media to multiple RSs at one time.=
=A0</div>
<div>Probably endpoints with limited resources or networks with restricted =
bandwidth will not able to provide this feature, but in mission critical en=
vironments this could be a very nice way to provide redundancy. The failove=
r mechanism described in the draft (and used by some of the vendors) is not=
 protected from single point of failure.=A0</div>
<div><br></div><div>5. Negotiating media handling capabilities=A0</div><div=
>SRP should allow RC and RS to negotiate audio/video codec capabilities (si=
milar to SIP/SDP) and the RC should send media in the agreed codec.=A0</div=
>
<div>In most cases the codec will be exactly the same as the one used in th=
e recorded session. However, high definition audio and video codecs are get=
ting used in more and more sessions, for recording, the requirement in most=
 cases allow to record the sessions in a lower resolution format. Either th=
e RC, either the RS should provide transcoding capabilities. I know several=
 video endpoints, which are able to send the video in 2 formats at one time=
 (HD and CIF) and I can also imagine devices with on-board transcoding capa=
bilities in the future (e.g. transcoding AAC-LD to G.729 for the RS) and so=
me of the current implementations also allow the transcoding of the recordi=
ng streams using network resources.=A0</div>
<div><br></div><div>6. Others=A0</div><div>There are some points, which are=
 mentioned in the use cases section, but no requirement is provided to supp=
ort the given feature:=A0</div><div>- Silent monitoring and real-time appli=
cations: the entire mechanism should allow the users to implement a real-ti=
me monitoring functionality, which means that the media has to be transmitt=
ed with considerably low delay, in a real-time fashion to the RS.=A0</div>
<div>- Handling complex calls: SRP should provide a mechanism to allow the =
RS to link together certain call segments in a complex call scenario (e.g. =
providing a global call identifier as a part of the meta data).=A0</div><di=
v>
<br></div><div>Best regards,</div><div>Gabor</div>

--001636458ca4a0e18f047ff7cead--

From pkyzivat@cisco.com  Fri Feb 19 12:13:31 2010
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Subject: Re: [dispatch] Preliminary version of Expert review of	draft-avasarala-dispatch-comm-div-notification-03
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I've been having an offline discussion of this with Ranjit and Adam.
But it deserves public discussion. So after getting permission from the 
others I am going to forward those messages to the list. (There will be 
several.)

The main issue is about the state model for this event package, or lack 
thereof.

If you have thoughts on this, please speak up.

	Thanks,
	Paul

From pkyzivat@cisco.com  Fri Feb 19 12:15:16 2010
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-------- Original Message --------
Subject: Re: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Wed, 17 Feb 2010 21:34:48 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
CC: Adam Roach <adam@nostrum.com>, John-Luc Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>

Ranjit,

[ taking IESG off the dist list for now, and adding Adam as the sub/not
author. Perhaps we should take this back to the dispatch list. ]

I haven't had time to read through the revised document yet, but I
skimmed through your comments. There is one that I think we need to talk
about because it potentially has significant impact. (I've picked the
first place it came up, but its also mentioned in other parts of my
issues and your responses.)

Adam: Is this enough for you to understand the issue?

Avasarala Ranjit-A20990 wrote:

>> ISSUE: The document details the notification body, and some inferences
> 
>> can be drawn about the state from the syntax of the notification body.
>> But this is tenuous. There should be an explicit specification of the 
>> state maintained, and how it is mapped onto the notification body.
> 
> <Ranjit-Feb16> The notifications for communication diversions are sent
> as and when they occur depending on the filter criteria set by the
> subscribers. So there is no need of maintaining any state information
> for the package. 

That doesn't fit my understanding of how event packages are defined.

IIUC, the semantics of event packages are described relative to some
sort of "state", not against another incoming event. (Whether this is
literally true in the implementation or not.)

So minimally, when a diversion happens, some particular information is
culled from it, and becomes the new state. Then that causes
notifications to the existing subscriptions.

After that it either sticks around, or else the state transitions back
to "none". But if it transitions back to "none" then that would trigger
another round of notifications - which you clearly don't want.

So it makes sense that it sticks around, in which case, when a new
subscription comes in it will get a notification about that last
diversion. (Unless there can be some sort of "implicit" filter that
suppresses notification when the state becomes "none".)

Adam - is that valid???

	Thanks,
	Paul


From pkyzivat@cisco.com  Fri Feb 19 12:15:30 2010
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-------- Original Message --------
Subject: RE: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Fri, 19 Feb 2010 01:08:59 +0800
From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
CC: Adam Roach <adam@nostrum.com>, John-Luc Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com>

Hi Paul, Adam

Since communication diversion notification information is not related to
any state of the actual communication, I feel there is no need to
maintain state for this. Also the Communication diversion notification
message gets triggered after the communication diversion has occurred.

The main intent of CDIVN is to notify the subscriber of a communication
diversion that occurred on their behalf in the network. This
notification is not part of any communication dialog or part of the
communication state machine and is completely independent of the actual
communication (or call).

Comments?

Thanks

Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Thursday, February 18, 2010 8:05 AM
To: Avasarala Ranjit-A20990
Cc: Adam Roach; John-Luc Bakker
Subject: Re: [dispatch] Preliminary version of Expert review
ofdraft-avasarala-dispatch-comm-div-notification-03

Ranjit,

[ taking IESG off the dist list for now, and adding Adam as the sub/not
author. Perhaps we should take this back to the dispatch list. ]

I haven't had time to read through the revised document yet, but I
skimmed through your comments. There is one that I think we need to talk
about because it potentially has significant impact. (I've picked the
first place it came up, but its also mentioned in other parts of my
issues and your responses.)

Adam: Is this enough for you to understand the issue?

Avasarala Ranjit-A20990 wrote:

>> ISSUE: The document details the notification body, and some 
>> inferences
> 
>> can be drawn about the state from the syntax of the notification
body.
>> But this is tenuous. There should be an explicit specification of the

>> state maintained, and how it is mapped onto the notification body.
> 
> <Ranjit-Feb16> The notifications for communication diversions are sent

> as and when they occur depending on the filter criteria set by the 
> subscribers. So there is no need of maintaining any state information 
> for the package.

That doesn't fit my understanding of how event packages are defined.

IIUC, the semantics of event packages are described relative to some
sort of "state", not against another incoming event. (Whether this is
literally true in the implementation or not.)

So minimally, when a diversion happens, some particular information is
culled from it, and becomes the new state. Then that causes
notifications to the existing subscriptions.

After that it either sticks around, or else the state transitions back
to "none". But if it transitions back to "none" then that would trigger
another round of notifications - which you clearly don't want.

So it makes sense that it sticks around, in which case, when a new
subscription comes in it will get a notification about that last
diversion. (Unless there can be some sort of "implicit" filter that
suppresses notification when the state becomes "none".)

Adam - is that valid???

	Thanks,
	Paul


From pkyzivat@cisco.com  Fri Feb 19 12:15:35 2010
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-------- Original Message --------
Subject: 	Re: [dispatch] Preliminary version of Expert review
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: 	Thu, 18 Feb 2010 14:00:29 -0600
From: 	Adam Roach <adam@nostrum.com>
To: 	Avasarala Ranjit-A20990 <ranjit@motorola.com>
CC: 	Paul Kyzivat <pkyzivat@cisco.com>, John-Luc Bakker <jbakker@rim.com>
References: 	<4B72716A.90006@ericsson.com>
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com>
<4B786433.1060000@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com>
<4B795009.7050309@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>
<4B7CA748.5000709@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com>



Paul is correct. The real thrust of RFC 3265 is monitoring the state of
something. State becomes interesting at transitions (hence the focus on
events), but the events are inherently talking about the state of something.

In your case, I agree that it won't be call state. It appears to be
related to some combination of call diversion preferences and proxy (?)
processing state? I agree with Paul that the overall model is ill
defined (even after reading 24.604), so I can't provide any concrete
suggestions here. The best I can do is recommend you examine the
existing corpus of event packages and come to an understanding about how
they work:

conference                  package           IETF SIPPING Working Group 
   [RFC4575]
consent-pending-additions   package           Gonzalo Camarillo 
    [RFC5362]
dialog                      package           IETF SIPPING Working Group 
   [RFC4235]
kpml                        package           Eric Burger 
    [RFC4730]
message-summary             package           Rohan Mahy 
    [RFC3842]
poc-settings                package           Miguel A. Garcia-Martin 
    [RFC4354]
presence                    package           Jonathan Rosenberg 
    [RFC3856]
reg                         package           Jonathan Rosenberg 
    [RFC3680]
refer                       package           Robert Sparks 
    [RFC3515]
spirits-INDPs               package           Vijay K. Gurbani 
    [RFC3910]
spirits-user-prof           package           Vijay K. Gurbani 
    [RFC3910]
winfo                       template-package  Jonathan Rosenberg 
    [RFC3857]
xcap-diff                   package           IETF Real-time 
Applications  [RFC-ietf-sip-xcapevent-08.txt]
                                               and Infrastructure Area

You'll note that every one of these event packages actually talks about
the *state* of something, and describes sending notifications when that
*state* changes. And, with the exception of the dialog event package
(and arguably the spirits-INDPs event package),  you'll note that none
of these states are actual call state -- they're the state of the
resource as it pertains to the aspect that the event package monitors.

For your proposed div-notification event package, you need to define
what this state is and how it changes. If you can't do that, then an
RFC3265 event package probably isn't the right tool for the job you have
in mind.

/a

On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
> Hi Paul, Adam
>
> Since communication diversion notification information is not related to
> any state of the actual communication, I feel there is no need to
> maintain state for this. Also the Communication diversion notification
> message gets triggered after the communication diversion has occurred.
>
> The main intent of CDIVN is to notify the subscriber of a communication
> diversion that occurred on their behalf in the network. This
> notification is not part of any communication dialog or part of the
> communication state machine and is completely independent of the actual
> communication (or call).
>
> Comments?
>
> Thanks
>
> Regards
> Ranjit
>
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Thursday, February 18, 2010 8:05 AM
> To: Avasarala Ranjit-A20990
> Cc: Adam Roach; John-Luc Bakker
> Subject: Re: [dispatch] Preliminary version of Expert review
> ofdraft-avasarala-dispatch-comm-div-notification-03
>
> Ranjit,
>
> [ taking IESG off the dist list for now, and adding Adam as the sub/not
> author. Perhaps we should take this back to the dispatch list. ]
>
> I haven't had time to read through the revised document yet, but I
> skimmed through your comments. There is one that I think we need to talk
> about because it potentially has significant impact. (I've picked the
> first place it came up, but its also mentioned in other parts of my
> issues and your responses.)
>
> Adam: Is this enough for you to understand the issue?
>
> Avasarala Ranjit-A20990 wrote:
>
>   
>>> ISSUE: The document details the notification body, and some 
>>> inferences
>>>       
>>     
>>> can be drawn about the state from the syntax of the notification
>>>       
> body.
>   
>>> But this is tenuous. There should be an explicit specification of the
>>>       
>   
>>> state maintained, and how it is mapped onto the notification body.
>>>       
>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>     
>   
>> as and when they occur depending on the filter criteria set by the 
>> subscribers. So there is no need of maintaining any state information 
>> for the package.
>>     
> That doesn't fit my understanding of how event packages are defined.
>
> IIUC, the semantics of event packages are described relative to some
> sort of "state", not against another incoming event. (Whether this is
> literally true in the implementation or not.)
>
> So minimally, when a diversion happens, some particular information is
> culled from it, and becomes the new state. Then that causes
> notifications to the existing subscriptions.
>
> After that it either sticks around, or else the state transitions back
> to "none". But if it transitions back to "none" then that would trigger
> another round of notifications - which you clearly don't want.
>
> So it makes sense that it sticks around, in which case, when a new
> subscription comes in it will get a notification about that last
> diversion. (Unless there can be some sort of "implicit" filter that
> suppresses notification when the state becomes "none".)
>
> Adam - is that valid???
>
> 	Thanks,
> 	Paul
>   


From pkyzivat@cisco.com  Fri Feb 19 12:16:00 2010
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Date: Fri, 19 Feb 2010 15:17:43 -0500
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-------- Original Message --------
Subject: Re: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Thu, 18 Feb 2010 16:30:20 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
To: Adam Roach <adam@nostrum.com>
CC: Avasarala Ranjit-A20990 <ranjit@motorola.com>,        John-Luc 
Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com> 
<4B7D9C5D.6090308@nostrum.com>

Thanks Adam.

Ranjit - I was thinking that minimally your state could be described as:

1) the incoming sip message to the diverting system
2) the resultant outgoing sip message from the diverting system

Before the first diversion the state would be nil.

After the first diversion, and the resulting notifications, there is a
question whether you retain that state. Its conceptually simpler if you
do. But if you want your server to be "stateless" in this regard, then
maybe you have to think about it immediately transitioning back to nil
state, before another diversion can occur.

As I noted in my prior message, that amounts to another state change,
and hence calls for another set of notifications, unless they are
suppressed by a filter of some sort.

Another issue is what happens with "simultaneous" diversions? Are you
expecting them to be serialized, resulting in distinct back-to-back
notifications? Or in that case do you want a single notification with
both diversions? If you want that, then the "state" needs to
simultaneously include both.

	Thanks,
	Paul

Adam Roach wrote:
> Paul is correct. The real thrust of RFC 3265 is monitoring the state of 
> something. State becomes interesting at transitions (hence the focus on 
> events), but the events are inherently talking about the state of something.
> 
> In your case, I agree that it won't be call state. It appears to be 
> related to some combination of call diversion preferences and proxy (?) 
> processing state? I agree with Paul that the overall model is ill 
> defined (even after reading 24.604), so I can't provide any concrete 
> suggestions here. The best I can do is recommend you examine the 
> existing corpus of event packages and come to an understanding about how 
> they work:
> 
> conference                  package           IETF SIPPING Working Group   [RFC4575] 
> consent-pending-additions   package           Gonzalo Camarillo            [RFC5362]
> dialog                      package           IETF SIPPING Working Group   [RFC4235] 
> kpml                        package           Eric Burger                  [RFC4730]
> message-summary             package           Rohan Mahy                   [RFC3842]
> poc-settings                package           Miguel A. Garcia-Martin      [RFC4354]
> presence                    package           Jonathan Rosenberg           [RFC3856]
> reg                         package           Jonathan Rosenberg           [RFC3680]
> refer                       package           Robert Sparks                [RFC3515]
> spirits-INDPs               package           Vijay K. Gurbani             [RFC3910]
> spirits-user-prof           package           Vijay K. Gurbani             [RFC3910]
> winfo                       template-package  Jonathan Rosenberg           [RFC3857]
> xcap-diff                   package           IETF Real-time Applications  [RFC-ietf-sip-xcapevent-08.txt]
>                                               and Infrastructure Area
> 
> You'll note that every one of these event packages actually talks about 
> the *state* of something, and describes sending notifications when that 
> *state* changes. And, with the exception of the dialog event package 
> (and arguably the spirits-INDPs event package),  you'll note that none 
> of these states are actual call state -- they're the state of the 
> resource as it pertains to the aspect that the event package monitors.
> 
> For your proposed div-notification event package, you need to define 
> what this state is and how it changes. If you can't do that, then an 
> RFC3265 event package probably isn't the right tool for the job you have 
> in mind.
> 
> /a
> 
> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>> Hi Paul, Adam
>>
>> Since communication diversion notification information is not related to
>> any state of the actual communication, I feel there is no need to
>> maintain state for this. Also the Communication diversion notification
>> message gets triggered after the communication diversion has occurred.
>>
>> The main intent of CDIVN is to notify the subscriber of a communication
>> diversion that occurred on their behalf in the network. This
>> notification is not part of any communication dialog or part of the
>> communication state machine and is completely independent of the actual
>> communication (or call).
>>
>> Comments?
>>
>> Thanks
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Thursday, February 18, 2010 8:05 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Adam Roach; John-Luc Bakker
>> Subject: Re: [dispatch] Preliminary version of Expert review
>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>
>> Ranjit,
>>
>> [ taking IESG off the dist list for now, and adding Adam as the sub/not
>> author. Perhaps we should take this back to the dispatch list. ]
>>
>> I haven't had time to read through the revised document yet, but I
>> skimmed through your comments. There is one that I think we need to talk
>> about because it potentially has significant impact. (I've picked the
>> first place it came up, but its also mentioned in other parts of my
>> issues and your responses.)
>>
>> Adam: Is this enough for you to understand the issue?
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>   
>>>> ISSUE: The document details the notification body, and some 
>>>> inferences
>>>>       
>>>     
>>>> can be drawn about the state from the syntax of the notification
>>>>       
>> body.
>>   
>>>> But this is tenuous. There should be an explicit specification of the
>>>>       
>>   
>>>> state maintained, and how it is mapped onto the notification body.
>>>>       
>>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>>     
>>   
>>> as and when they occur depending on the filter criteria set by the 
>>> subscribers. So there is no need of maintaining any state information 
>>> for the package.
>>>     
>> That doesn't fit my understanding of how event packages are defined.
>>
>> IIUC, the semantics of event packages are described relative to some
>> sort of "state", not against another incoming event. (Whether this is
>> literally true in the implementation or not.)
>>
>> So minimally, when a diversion happens, some particular information is
>> culled from it, and becomes the new state. Then that causes
>> notifications to the existing subscriptions.
>>
>> After that it either sticks around, or else the state transitions back
>> to "none". But if it transitions back to "none" then that would trigger
>> another round of notifications - which you clearly don't want.
>>
>> So it makes sense that it sticks around, in which case, when a new
>> subscription comes in it will get a notification about that last
>> diversion. (Unless there can be some sort of "implicit" filter that
>> suppresses notification when the state becomes "none".)
>>
>> Adam - is that valid???
>>
>> 	Thanks,
>> 	Paul
>>   
> 


From pkyzivat@cisco.com  Fri Feb 19 12:16:38 2010
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Date: Fri, 19 Feb 2010 15:18:34 -0500
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-------- Original Message --------
Subject: Re: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Thu, 18 Feb 2010 15:37:44 -0600
From: Adam Roach <adam@nostrum.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
CC: Avasarala Ranjit-A20990 <ranjit@motorola.com>,        John-Luc 
Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com> 
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com>

Another way to examine this problem might be to consider the state as
being composed of the list of diverted callers. You can age entries out,
and reset the list when diversion is shut off. Doing this makes it much
clearer what should happen for the near-simultaneous case.

Note that this is conceptually similar to what the winfo event package
template does with its "waiting" state.

/a

On 2/18/10 15:30, Feb 18, Paul Kyzivat wrote:
> Thanks Adam.
>
> Ranjit - I was thinking that minimally your state could be described as:
>
> 1) the incoming sip message to the diverting system
> 2) the resultant outgoing sip message from the diverting system
>
> Before the first diversion the state would be nil.
>
> After the first diversion, and the resulting notifications, there is a 
> question whether you retain that state. Its conceptually simpler if 
> you do. But if you want your server to be "stateless" in this regard, 
> then maybe you have to think about it immediately transitioning back 
> to nil state, before another diversion can occur.
>
> As I noted in my prior message, that amounts to another state change, 
> and hence calls for another set of notifications, unless they are 
> suppressed by a filter of some sort.
>
> Another issue is what happens with "simultaneous" diversions? Are you 
> expecting them to be serialized, resulting in distinct back-to-back 
> notifications? Or in that case do you want a single notification with 
> both diversions? If you want that, then the "state" needs to 
> simultaneously include both.
>
>     Thanks,
>     Paul
>
> Adam Roach wrote:
>> Paul is correct. The real thrust of RFC 3265 is monitoring the state 
>> of something. State becomes interesting at transitions (hence the 
>> focus on events), but the events are inherently talking about the 
>> state of something.
>>
>> In your case, I agree that it won't be call state. It appears to be 
>> related to some combination of call diversion preferences and proxy 
>> (?) processing state? I agree with Paul that the overall model is ill 
>> defined (even after reading 24.604), so I can't provide any concrete 
>> suggestions here. The best I can do is recommend you examine the 
>> existing corpus of event packages and come to an understanding about 
>> how they work:
>>
>> conference                  package           IETF SIPPING Working 
>> Group   [RFC4575] consent-pending-additions   package           
>> Gonzalo Camarillo            [RFC5362]
>> dialog                      package           IETF SIPPING Working 
>> Group   [RFC4235] kpml                        package           Eric 
>> Burger                  [RFC4730]
>> message-summary             package           Rohan 
>> Mahy                   [RFC3842]
>> poc-settings                package           Miguel A. 
>> Garcia-Martin      [RFC4354]
>> presence                    package           Jonathan 
>> Rosenberg           [RFC3856]
>> reg                         package           Jonathan 
>> Rosenberg           [RFC3680]
>> refer                       package           Robert 
>> Sparks                [RFC3515]
>> spirits-INDPs               package           Vijay K. 
>> Gurbani             [RFC3910]
>> spirits-user-prof           package           Vijay K. 
>> Gurbani             [RFC3910]
>> winfo                       template-package  Jonathan 
>> Rosenberg           [RFC3857]
>> xcap-diff                   package           IETF Real-time 
>> Applications  [RFC-ietf-sip-xcapevent-08.txt]
>>                                               and Infrastructure Area
>>
>> You'll note that every one of these event packages actually talks 
>> about the *state* of something, and describes sending notifications 
>> when that *state* changes. And, with the exception of the dialog 
>> event package (and arguably the spirits-INDPs event package),  you'll 
>> note that none of these states are actual call state -- they're the 
>> state of the resource as it pertains to the aspect that the event 
>> package monitors.
>>
>> For your proposed div-notification event package, you need to define 
>> what this state is and how it changes. If you can't do that, then an 
>> RFC3265 event package probably isn't the right tool for the job you 
>> have in mind.
>>
>> /a
>>
>> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>>> Hi Paul, Adam
>>>
>>> Since communication diversion notification information is not 
>>> related to
>>> any state of the actual communication, I feel there is no need to
>>> maintain state for this. Also the Communication diversion notification
>>> message gets triggered after the communication diversion has occurred.
>>>
>>> The main intent of CDIVN is to notify the subscriber of a communication
>>> diversion that occurred on their behalf in the network. This
>>> notification is not part of any communication dialog or part of the
>>> communication state machine and is completely independent of the actual
>>> communication (or call).
>>>
>>> Comments?
>>>
>>> Thanks
>>>
>>> Regards
>>> Ranjit
>>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] Sent: Thursday, 
>>> February 18, 2010 8:05 AM
>>> To: Avasarala Ranjit-A20990
>>> Cc: Adam Roach; John-Luc Bakker
>>> Subject: Re: [dispatch] Preliminary version of Expert review
>>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>>
>>> Ranjit,
>>>
>>> [ taking IESG off the dist list for now, and adding Adam as the sub/not
>>> author. Perhaps we should take this back to the dispatch list. ]
>>>
>>> I haven't had time to read through the revised document yet, but I
>>> skimmed through your comments. There is one that I think we need to 
>>> talk
>>> about because it potentially has significant impact. (I've picked the
>>> first place it came up, but its also mentioned in other parts of my
>>> issues and your responses.)
>>>
>>> Adam: Is this enough for you to understand the issue?
>>>
>>> Avasarala Ranjit-A20990 wrote:
>>>
>>>>> ISSUE: The document details the notification body, and some 
>>>>> inferences
>>>>> can be drawn about the state from the syntax of the notification
>>> body.
>>>>> But this is tenuous. There should be an explicit specification of the
>>>>> state maintained, and how it is mapped onto the notification body.
>>>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>>> as and when they occur depending on the filter criteria set by the 
>>>> subscribers. So there is no need of maintaining any state 
>>>> information for the package.
>>> That doesn't fit my understanding of how event packages are defined.
>>>
>>> IIUC, the semantics of event packages are described relative to some
>>> sort of "state", not against another incoming event. (Whether this is
>>> literally true in the implementation or not.)
>>>
>>> So minimally, when a diversion happens, some particular information is
>>> culled from it, and becomes the new state. Then that causes
>>> notifications to the existing subscriptions.
>>>
>>> After that it either sticks around, or else the state transitions back
>>> to "none". But if it transitions back to "none" then that would trigger
>>> another round of notifications - which you clearly don't want.
>>>
>>> So it makes sense that it sticks around, in which case, when a new
>>> subscription comes in it will get a notification about that last
>>> diversion. (Unless there can be some sort of "implicit" filter that
>>> suppresses notification when the state becomes "none".)
>>>
>>> Adam - is that valid???
>>>
>>>     Thanks,
>>>     Paul
>>



From pkyzivat@cisco.com  Fri Feb 19 12:16:57 2010
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-------- Original Message --------
Subject: Re: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Thu, 18 Feb 2010 16:58:51 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
To: Adam Roach <adam@nostrum.com>
CC: Avasarala Ranjit-A20990 <ranjit@motorola.com>,        John-Luc 
Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com> 
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com> 
<4B7DB328.1020306@nostrum.com>



Adam Roach wrote:
> Another way to examine this problem might be to consider the state as 
> being composed of the list of diverted callers. You can age entries out, 
> and reset the list when diversion is shut off. Doing this makes it much 
> clearer what should happen for the near-simultaneous case.
> 
> Note that this is conceptually similar to what the winfo event package 
> template does with its "waiting" state.

I'm just inferring from what Ranjit has said that they envision a server
that doesn't keep *any* state about diverted calls. I'm trying to see if
this can be so and still conform to the 3265 view of the world.

Do you think its a valid approach to have state instantaneously appear
and disappear, and for a filter to block the delivery of the transitions
back to nil state?

	Thanks,
	Paul

> /a
> 
> On 2/18/10 15:30, Feb 18, Paul Kyzivat wrote:
>> Thanks Adam.
>>
>> Ranjit - I was thinking that minimally your state could be described as:
>>
>> 1) the incoming sip message to the diverting system
>> 2) the resultant outgoing sip message from the diverting system
>>
>> Before the first diversion the state would be nil.
>>
>> After the first diversion, and the resulting notifications, there is a 
>> question whether you retain that state. Its conceptually simpler if 
>> you do. But if you want your server to be "stateless" in this regard, 
>> then maybe you have to think about it immediately transitioning back 
>> to nil state, before another diversion can occur.
>>
>> As I noted in my prior message, that amounts to another state change, 
>> and hence calls for another set of notifications, unless they are 
>> suppressed by a filter of some sort.
>>
>> Another issue is what happens with "simultaneous" diversions? Are you 
>> expecting them to be serialized, resulting in distinct back-to-back 
>> notifications? Or in that case do you want a single notification with 
>> both diversions? If you want that, then the "state" needs to 
>> simultaneously include both.
>>
>>     Thanks,
>>     Paul
>>
>> Adam Roach wrote:
>>> Paul is correct. The real thrust of RFC 3265 is monitoring the state 
>>> of something. State becomes interesting at transitions (hence the 
>>> focus on events), but the events are inherently talking about the 
>>> state of something.
>>>
>>> In your case, I agree that it won't be call state. It appears to be 
>>> related to some combination of call diversion preferences and proxy 
>>> (?) processing state? I agree with Paul that the overall model is ill 
>>> defined (even after reading 24.604), so I can't provide any concrete 
>>> suggestions here. The best I can do is recommend you examine the 
>>> existing corpus of event packages and come to an understanding about 
>>> how they work:
>>>
>>> conference                  package           IETF SIPPING Working 
>>> Group   [RFC4575] consent-pending-additions   package           
>>> Gonzalo Camarillo            [RFC5362]
>>> dialog                      package           IETF SIPPING Working 
>>> Group   [RFC4235] kpml                        package           Eric 
>>> Burger                  [RFC4730]
>>> message-summary             package           Rohan 
>>> Mahy                   [RFC3842]
>>> poc-settings                package           Miguel A. 
>>> Garcia-Martin      [RFC4354]
>>> presence                    package           Jonathan 
>>> Rosenberg           [RFC3856]
>>> reg                         package           Jonathan 
>>> Rosenberg           [RFC3680]
>>> refer                       package           Robert 
>>> Sparks                [RFC3515]
>>> spirits-INDPs               package           Vijay K. 
>>> Gurbani             [RFC3910]
>>> spirits-user-prof           package           Vijay K. 
>>> Gurbani             [RFC3910]
>>> winfo                       template-package  Jonathan 
>>> Rosenberg           [RFC3857]
>>> xcap-diff                   package           IETF Real-time 
>>> Applications  [RFC-ietf-sip-xcapevent-08.txt]
>>>                                               and Infrastructure Area
>>>
>>> You'll note that every one of these event packages actually talks 
>>> about the *state* of something, and describes sending notifications 
>>> when that *state* changes. And, with the exception of the dialog 
>>> event package (and arguably the spirits-INDPs event package),  you'll 
>>> note that none of these states are actual call state -- they're the 
>>> state of the resource as it pertains to the aspect that the event 
>>> package monitors.
>>>
>>> For your proposed div-notification event package, you need to define 
>>> what this state is and how it changes. If you can't do that, then an 
>>> RFC3265 event package probably isn't the right tool for the job you 
>>> have in mind.
>>>
>>> /a
>>>
>>> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>>>> Hi Paul, Adam
>>>>
>>>> Since communication diversion notification information is not 
>>>> related to
>>>> any state of the actual communication, I feel there is no need to
>>>> maintain state for this. Also the Communication diversion notification
>>>> message gets triggered after the communication diversion has occurred.
>>>>
>>>> The main intent of CDIVN is to notify the subscriber of a communication
>>>> diversion that occurred on their behalf in the network. This
>>>> notification is not part of any communication dialog or part of the
>>>> communication state machine and is completely independent of the actual
>>>> communication (or call).
>>>>
>>>> Comments?
>>>>
>>>> Thanks
>>>>
>>>> Regards
>>>> Ranjit
>>>>
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] Sent: Thursday, 
>>>> February 18, 2010 8:05 AM
>>>> To: Avasarala Ranjit-A20990
>>>> Cc: Adam Roach; John-Luc Bakker
>>>> Subject: Re: [dispatch] Preliminary version of Expert review
>>>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>>>
>>>> Ranjit,
>>>>
>>>> [ taking IESG off the dist list for now, and adding Adam as the sub/not
>>>> author. Perhaps we should take this back to the dispatch list. ]
>>>>
>>>> I haven't had time to read through the revised document yet, but I
>>>> skimmed through your comments. There is one that I think we need to 
>>>> talk
>>>> about because it potentially has significant impact. (I've picked the
>>>> first place it came up, but its also mentioned in other parts of my
>>>> issues and your responses.)
>>>>
>>>> Adam: Is this enough for you to understand the issue?
>>>>
>>>> Avasarala Ranjit-A20990 wrote:
>>>>
>>>>>> ISSUE: The document details the notification body, and some 
>>>>>> inferences
>>>>>> can be drawn about the state from the syntax of the notification
>>>> body.
>>>>>> But this is tenuous. There should be an explicit specification of the
>>>>>> state maintained, and how it is mapped onto the notification body.
>>>>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>>>> as and when they occur depending on the filter criteria set by the 
>>>>> subscribers. So there is no need of maintaining any state 
>>>>> information for the package.
>>>> That doesn't fit my understanding of how event packages are defined.
>>>>
>>>> IIUC, the semantics of event packages are described relative to some
>>>> sort of "state", not against another incoming event. (Whether this is
>>>> literally true in the implementation or not.)
>>>>
>>>> So minimally, when a diversion happens, some particular information is
>>>> culled from it, and becomes the new state. Then that causes
>>>> notifications to the existing subscriptions.
>>>>
>>>> After that it either sticks around, or else the state transitions back
>>>> to "none". But if it transitions back to "none" then that would trigger
>>>> another round of notifications - which you clearly don't want.
>>>>
>>>> So it makes sense that it sticks around, in which case, when a new
>>>> subscription comes in it will get a notification about that last
>>>> diversion. (Unless there can be some sort of "implicit" filter that
>>>> suppresses notification when the state becomes "none".)
>>>>
>>>> Adam - is that valid???
>>>>
>>>>     Thanks,
>>>>     Paul
>>>
> 
> 


From pkyzivat@cisco.com  Fri Feb 19 12:17:07 2010
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-------- Original Message --------
Subject: 	Re: [dispatch] Preliminary version of Expert review
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: 	Thu, 18 Feb 2010 16:10:07 -0600
From: 	Adam Roach <adam@nostrum.com>
To: 	Paul Kyzivat <pkyzivat@cisco.com>
CC: 	Avasarala Ranjit-A20990 <ranjit@motorola.com>, John-Luc Bakker
<jbakker@rim.com>
References: 	<4B72716A.90006@ericsson.com>
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com>
<4B786433.1060000@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com>
<4B795009.7050309@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>
<4B7CA748.5000709@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com>
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com>
<4B7DB328.1020306@nostrum.com> <4B7DB81B.4030201@cisco.com>



On 2/18/10 15:58, Feb 18, Paul Kyzivat wrote:
>
>
> Adam Roach wrote:
>> Another way to examine this problem might be to consider the state as 
>> being composed of the list of diverted callers. You can age entries 
>> out, and reset the list when diversion is shut off. Doing this makes 
>> it much clearer what should happen for the near-simultaneous case.
>>
>> Note that this is conceptually similar to what the winfo event 
>> package template does with its "waiting" state.
>
> I'm just inferring from what Ranjit has said that they envision a 
> server that doesn't keep *any* state about diverted calls. I'm trying 
> to see if this can be so and still conform to the 3265 view of the world.
>
> Do you think its a valid approach to have state instantaneously appear 
> and disappear, and for a filter to block the delivery of the 
> transitions back to nil state?

I think that's a bit tortured, as state models go. In particular, we've
defined (and are defining) mechanisms that allow various modifications
to the way SUBSCRIBE and NOTIFY work, and they assume more persistence
than what you've described. For example, consider how your proposed
state model would interact with subnot-etags or the ongoing throttling
work. It doesn't make any sense, does it? Because we're assuming real
state behind the event package, not instantaneous events that result in
no persistent state.

Really, none of this should be surprising to anyone; it's spelled out
pretty clearly in the definition of an event package as given in RFC 3265:

    Event Package: An event package is an additional specification which
       defines a _set of state information_ to be reported by a notifier to
       a subscriber.  Event packages also define further syntax and
       semantics based on the framework defined by this document required
       to convey such state information.

If you don't want to track _state_, then this is the wrong tool. If you
want to use this tool, you need to define a state model.

/a

From pkyzivat@cisco.com  Fri Feb 19 12:17:21 2010
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-------- Original Message --------
Subject: 	Re: [dispatch] Preliminary version of Expert review
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: 	Thu, 18 Feb 2010 16:10:07 -0600
From: 	Adam Roach <adam@nostrum.com>
To: 	Paul Kyzivat <pkyzivat@cisco.com>
CC: 	Avasarala Ranjit-A20990 <ranjit@motorola.com>, John-Luc Bakker
<jbakker@rim.com>
References: 	<4B72716A.90006@ericsson.com>
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com>
<4B786433.1060000@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com>
<4B795009.7050309@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>
<4B7CA748.5000709@cisco.com>
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com>
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com>
<4B7DB328.1020306@nostrum.com> <4B7DB81B.4030201@cisco.com>



On 2/18/10 15:58, Feb 18, Paul Kyzivat wrote:
>
>
> Adam Roach wrote:
>> Another way to examine this problem might be to consider the state as 
>> being composed of the list of diverted callers. You can age entries 
>> out, and reset the list when diversion is shut off. Doing this makes 
>> it much clearer what should happen for the near-simultaneous case.
>>
>> Note that this is conceptually similar to what the winfo event 
>> package template does with its "waiting" state.
>
> I'm just inferring from what Ranjit has said that they envision a 
> server that doesn't keep *any* state about diverted calls. I'm trying 
> to see if this can be so and still conform to the 3265 view of the world.
>
> Do you think its a valid approach to have state instantaneously appear 
> and disappear, and for a filter to block the delivery of the 
> transitions back to nil state?

I think that's a bit tortured, as state models go. In particular, we've
defined (and are defining) mechanisms that allow various modifications
to the way SUBSCRIBE and NOTIFY work, and they assume more persistence
than what you've described. For example, consider how your proposed
state model would interact with subnot-etags or the ongoing throttling
work. It doesn't make any sense, does it? Because we're assuming real
state behind the event package, not instantaneous events that result in
no persistent state.

Really, none of this should be surprising to anyone; it's spelled out
pretty clearly in the definition of an event package as given in RFC 3265:

    Event Package: An event package is an additional specification which
       defines a _set of state information_ to be reported by a notifier to
       a subscriber.  Event packages also define further syntax and
       semantics based on the framework defined by this document required
       to convey such state information.

If you don't want to track _state_, then this is the wrong tool. If you
want to use this tool, you need to define a state model.

/a

From pkyzivat@cisco.com  Fri Feb 19 12:17:54 2010
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Subject: [dispatch] [Fwd: RE: Preliminary version of Expert review ofdraft-avasarala-dispatch-comm-div-notification-03]
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-------- Original Message --------
Subject: RE: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Fri, 19 Feb 2010 17:59:05 +0800
From: Avasarala Ranjit-A20990 <ranjit@motorola.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Adam Roach <adam@nostrum.com>
CC: John-Luc Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com> 
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com>

Hi Paul

In case of IMS, the diverting system would be the Application Server
(AS) that is hosting the CDIVN service. Now if we need the concept of
state, then we need to monitor the state of the AS or the CDIVN service.

So true as you say, we prefer the notification server to be "Stateless"
and hence it would start with "idle" state before sending a notification
information and will transition to "send" state and immediately back to
"idle" state waiting for another diversion to occur. Now when another
diversion occurs, if the notification server needs to send a
notification (depending on filter criteria), then it will transition to
"send" state to send the notification, else it will remain in "idle"
state.

True simultaneous diversions will be serialized and the server needs to
switch back and forth between states to send multiple notifications i.e
idle->send->idle.


Is this ok?

Thanks

Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Friday, February 19, 2010 3:00 AM
To: Adam Roach
Cc: Avasarala Ranjit-A20990; John-Luc Bakker
Subject: Re: [dispatch] Preliminary version of Expert review
ofdraft-avasarala-dispatch-comm-div-notification-03

Thanks Adam.

Ranjit - I was thinking that minimally your state could be described as:

1) the incoming sip message to the diverting system
2) the resultant outgoing sip message from the diverting system

Before the first diversion the state would be nil.

After the first diversion, and the resulting notifications, there is a
question whether you retain that state. Its conceptually simpler if you
do. But if you want your server to be "stateless" in this regard, then
maybe you have to think about it immediately transitioning back to nil
state, before another diversion can occur.

As I noted in my prior message, that amounts to another state change,
and hence calls for another set of notifications, unless they are
suppressed by a filter of some sort.

Another issue is what happens with "simultaneous" diversions? Are you
expecting them to be serialized, resulting in distinct back-to-back
notifications? Or in that case do you want a single notification with
both diversions? If you want that, then the "state" needs to
simultaneously include both.

	Thanks,
	Paul

Adam Roach wrote:
> Paul is correct. The real thrust of RFC 3265 is monitoring the state 
> of something. State becomes interesting at transitions (hence the 
> focus on events), but the events are inherently talking about the
state of something.
> 
> In your case, I agree that it won't be call state. It appears to be 
> related to some combination of call diversion preferences and proxy 
> (?) processing state? I agree with Paul that the overall model is ill 
> defined (even after reading 24.604), so I can't provide any concrete 
> suggestions here. The best I can do is recommend you examine the 
> existing corpus of event packages and come to an understanding about 
> how they work:
> 
> conference                  package           IETF SIPPING Working
Group   [RFC4575]
> consent-pending-additions   package           Gonzalo Camarillo
[RFC5362]
> dialog                      package           IETF SIPPING Working
Group   [RFC4235]
> kpml                        package           Eric Burger
[RFC4730]
> message-summary             package           Rohan Mahy
[RFC3842]
> poc-settings                package           Miguel A. Garcia-Martin
[RFC4354]
> presence                    package           Jonathan Rosenberg
[RFC3856]
> reg                         package           Jonathan Rosenberg
[RFC3680]
> refer                       package           Robert Sparks
[RFC3515]
> spirits-INDPs               package           Vijay K. Gurbani
[RFC3910]
> spirits-user-prof           package           Vijay K. Gurbani
[RFC3910]
> winfo                       template-package  Jonathan Rosenberg
[RFC3857]
> xcap-diff                   package           IETF Real-time
Applications  [RFC-ietf-sip-xcapevent-08.txt]
>                                               and Infrastructure Area
> 
> You'll note that every one of these event packages actually talks 
> about the *state* of something, and describes sending notifications 
> when that
> *state* changes. And, with the exception of the dialog event package 
> (and arguably the spirits-INDPs event package),  you'll note that none

> of these states are actual call state -- they're the state of the 
> resource as it pertains to the aspect that the event package monitors.
> 
> For your proposed div-notification event package, you need to define 
> what this state is and how it changes. If you can't do that, then an
> RFC3265 event package probably isn't the right tool for the job you 
> have in mind.
> 
> /a
> 
> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>> Hi Paul, Adam
>>
>> Since communication diversion notification information is not related

>> to any state of the actual communication, I feel there is no need to 
>> maintain state for this. Also the Communication diversion 
>> notification message gets triggered after the communication diversion
has occurred.
>>
>> The main intent of CDIVN is to notify the subscriber of a 
>> communication diversion that occurred on their behalf in the network.

>> This notification is not part of any communication dialog or part of 
>> the communication state machine and is completely independent of the 
>> actual communication (or call).
>>
>> Comments?
>>
>> Thanks
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Thursday, February 18, 2010 8:05 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Adam Roach; John-Luc Bakker
>> Subject: Re: [dispatch] Preliminary version of Expert review
>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>
>> Ranjit,
>>
>> [ taking IESG off the dist list for now, and adding Adam as the 
>> sub/not author. Perhaps we should take this back to the dispatch 
>> list. ]
>>
>> I haven't had time to read through the revised document yet, but I 
>> skimmed through your comments. There is one that I think we need to 
>> talk about because it potentially has significant impact. (I've 
>> picked the first place it came up, but its also mentioned in other 
>> parts of my issues and your responses.)
>>
>> Adam: Is this enough for you to understand the issue?
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>   
>>>> ISSUE: The document details the notification body, and some 
>>>> inferences
>>>>       
>>>     
>>>> can be drawn about the state from the syntax of the notification
>>>>       
>> body.
>>   
>>>> But this is tenuous. There should be an explicit specification of 
>>>> the
>>>>       
>>   
>>>> state maintained, and how it is mapped onto the notification body.
>>>>       
>>> <Ranjit-Feb16> The notifications for communication diversions are 
>>> sent
>>>     
>>   
>>> as and when they occur depending on the filter criteria set by the 
>>> subscribers. So there is no need of maintaining any state 
>>> information for the package.
>>>     
>> That doesn't fit my understanding of how event packages are defined.
>>
>> IIUC, the semantics of event packages are described relative to some 
>> sort of "state", not against another incoming event. (Whether this is

>> literally true in the implementation or not.)
>>
>> So minimally, when a diversion happens, some particular information 
>> is culled from it, and becomes the new state. Then that causes 
>> notifications to the existing subscriptions.
>>
>> After that it either sticks around, or else the state transitions 
>> back to "none". But if it transitions back to "none" then that would 
>> trigger another round of notifications - which you clearly don't
want.
>>
>> So it makes sense that it sticks around, in which case, when a new 
>> subscription comes in it will get a notification about that last 
>> diversion. (Unless there can be some sort of "implicit" filter that 
>> suppresses notification when the state becomes "none".)
>>
>> Adam - is that valid???
>>
>> 	Thanks,
>> 	Paul
>>   
> 


From pkyzivat@cisco.com  Fri Feb 19 12:18:06 2010
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Subject: [dispatch] [Fwd: Re: Preliminary version of Expert review ofdraft-avasarala-dispatch-comm-div-notification-03]
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-------- Original Message --------
Subject: Re: [dispatch] Preliminary version of Expert review 
ofdraft-avasarala-dispatch-comm-div-notification-03
Date: Fri, 19 Feb 2010 08:20:28 -0600
From: Adam Roach <adam@nostrum.com>
To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
CC: Paul Kyzivat <pkyzivat@cisco.com>, John-Luc Bakker <jbakker@rim.com>
References: <4B72716A.90006@ericsson.com> 
<4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com> 
<4B786433.1060000@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com> 
<4B795009.7050309@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com> 
<4B7CA748.5000709@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com> 
<4B7D9C5D.6090308@nostrum.com> <4B7DB16C.2050601@cisco.com> 
<750BBC72E178114F9DC4872EBFF29A5B0A318004@ZMY16EXM66.ds.mot.com>

On 2/19/10 03:59, Feb 19, Avasarala Ranjit-A20990 wrote:
> Hi Paul
>
> In case of IMS, the diverting system would be the Application Server
> (AS) that is hosting the CDIVN service. Now if we need the concept of
> state, then we need to monitor the state of the AS or the CDIVN service.
>
> So true as you say, we prefer the notification server to be "Stateless"
> and hence it would start with "idle" state before sending a notification
> information and will transition to "send" state and immediately back to
> "idle" state waiting for another diversion to occur. Now when another
> diversion occurs, if the notification server needs to send a
> notification (depending on filter criteria), then it will transition to
> "send" state to send the notification, else it will remain in "idle"
> state.
>
> True simultaneous diversions will be serialized and the server needs to
> switch back and forth between states to send multiple notifications i.e
> idle->send->idle.
>
>
> Is this ok?
>    

I don't think so. See my other notes on the topic.

/a


From ankriste@cisco.com  Fri Feb 19 14:38:53 2010
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version of Expert review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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What is the state model for KPML? Seems to me that that's just as 
stateless as this CDIV package.

Thanks,
Anders

On 2/19/2010 12:17 PM, Paul Kyzivat wrote:
>
>
> -------- Original Message --------
> Subject: Re: [dispatch] Preliminary version of Expert review
> ofdraft-avasarala-dispatch-comm-div-notification-03
> Date: Thu, 18 Feb 2010 14:00:29 -0600
> From: Adam Roach <adam@nostrum.com>
> To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
> CC: Paul Kyzivat <pkyzivat@cisco.com>, John-Luc Bakker <jbakker@rim.com>
> References: <4B72716A.90006@ericsson.com>
> <4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com>
> <4B786433.1060000@cisco.com>
> <750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com>
> <4B795009.7050309@cisco.com>
> <750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>
> <4B7CA748.5000709@cisco.com>
> <750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com>
>
>
>
> Paul is correct. The real thrust of RFC 3265 is monitoring the state of
> something. State becomes interesting at transitions (hence the focus on
> events), but the events are inherently talking about the state of
> something.
>
> In your case, I agree that it won't be call state. It appears to be
> related to some combination of call diversion preferences and proxy (?)
> processing state? I agree with Paul that the overall model is ill
> defined (even after reading 24.604), so I can't provide any concrete
> suggestions here. The best I can do is recommend you examine the
> existing corpus of event packages and come to an understanding about how
> they work:
>
> conference package IETF SIPPING Working Group [RFC4575]
> consent-pending-additions package Gonzalo Camarillo [RFC5362]
> dialog package IETF SIPPING Working Group [RFC4235]
> kpml package Eric Burger [RFC4730]
> message-summary package Rohan Mahy [RFC3842]
> poc-settings package Miguel A. Garcia-Martin [RFC4354]
> presence package Jonathan Rosenberg [RFC3856]
> reg package Jonathan Rosenberg [RFC3680]
> refer package Robert Sparks [RFC3515]
> spirits-INDPs package Vijay K. Gurbani [RFC3910]
> spirits-user-prof package Vijay K. Gurbani [RFC3910]
> winfo template-package Jonathan Rosenberg [RFC3857]
> xcap-diff package IETF Real-time Applications
> [RFC-ietf-sip-xcapevent-08.txt]
> and Infrastructure Area
>
> You'll note that every one of these event packages actually talks about
> the *state* of something, and describes sending notifications when that
> *state* changes. And, with the exception of the dialog event package
> (and arguably the spirits-INDPs event package), you'll note that none
> of these states are actual call state -- they're the state of the
> resource as it pertains to the aspect that the event package monitors.
>
> For your proposed div-notification event package, you need to define
> what this state is and how it changes. If you can't do that, then an
> RFC3265 event package probably isn't the right tool for the job you have
> in mind.
>
> /a
>
> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>> Hi Paul, Adam
>>
>> Since communication diversion notification information is not related to
>> any state of the actual communication, I feel there is no need to
>> maintain state for this. Also the Communication diversion notification
>> message gets triggered after the communication diversion has occurred.
>>
>> The main intent of CDIVN is to notify the subscriber of a communication
>> diversion that occurred on their behalf in the network. This
>> notification is not part of any communication dialog or part of the
>> communication state machine and is completely independent of the actual
>> communication (or call).
>>
>> Comments?
>>
>> Thanks
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] Sent: Thursday,
>> February 18, 2010 8:05 AM
>> To: Avasarala Ranjit-A20990
>> Cc: Adam Roach; John-Luc Bakker
>> Subject: Re: [dispatch] Preliminary version of Expert review
>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>
>> Ranjit,
>>
>> [ taking IESG off the dist list for now, and adding Adam as the sub/not
>> author. Perhaps we should take this back to the dispatch list. ]
>>
>> I haven't had time to read through the revised document yet, but I
>> skimmed through your comments. There is one that I think we need to talk
>> about because it potentially has significant impact. (I've picked the
>> first place it came up, but its also mentioned in other parts of my
>> issues and your responses.)
>>
>> Adam: Is this enough for you to understand the issue?
>>
>> Avasarala Ranjit-A20990 wrote:
>>
>>>> ISSUE: The document details the notification body, and some inferences
>>>> can be drawn about the state from the syntax of the notification
>> body.
>>>> But this is tenuous. There should be an explicit specification of the
>>>> state maintained, and how it is mapped onto the notification body.
>>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>> as and when they occur depending on the filter criteria set by the
>>> subscribers. So there is no need of maintaining any state information
>>> for the package.
>> That doesn't fit my understanding of how event packages are defined.
>>
>> IIUC, the semantics of event packages are described relative to some
>> sort of "state", not against another incoming event. (Whether this is
>> literally true in the implementation or not.)
>>
>> So minimally, when a diversion happens, some particular information is
>> culled from it, and becomes the new state. Then that causes
>> notifications to the existing subscriptions.
>>
>> After that it either sticks around, or else the state transitions back
>> to "none". But if it transitions back to "none" then that would trigger
>> another round of notifications - which you clearly don't want.
>>
>> So it makes sense that it sticks around, in which case, when a new
>> subscription comes in it will get a notification about that last
>> diversion. (Unless there can be some sort of "implicit" filter that
>> suppresses notification when the state becomes "none".)
>>
>> Adam - is that valid???
>>
>> Thanks,
>> Paul
>
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
>

From pkyzivat@cisco.com  Fri Feb 19 15:42:11 2010
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version of Expert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Anders Kristensen wrote:
> What is the state model for KPML? Seems to me that that's just as 
> stateless as this CDIV package.

Sure it does. The state consists of zero or more buffered DTMF characters.

Depending on how you look at it, there may be separate state for each 
subscriber - which is populated with incoming dtmf characters - or else 
you can think of it as a single buffer with a separate starting point 
for each subscriber.

The state is essential for the handling of resubscribes that change the 
filter.

	Thanks,
	Paul

> Thanks,
> Anders
> 
> On 2/19/2010 12:17 PM, Paul Kyzivat wrote:
>>
>>
>> -------- Original Message --------
>> Subject: Re: [dispatch] Preliminary version of Expert review
>> ofdraft-avasarala-dispatch-comm-div-notification-03
>> Date: Thu, 18 Feb 2010 14:00:29 -0600
>> From: Adam Roach <adam@nostrum.com>
>> To: Avasarala Ranjit-A20990 <ranjit@motorola.com>
>> CC: Paul Kyzivat <pkyzivat@cisco.com>, John-Luc Bakker <jbakker@rim.com>
>> References: <4B72716A.90006@ericsson.com>
>> <4B72BC3D.20701@cisco.com><4B72BE4A.6030508@ericsson.com>
>> <4B786433.1060000@cisco.com>
>> <750BBC72E178114F9DC4872EBFF29A5B0A2CD7FB@ZMY16EXM66.ds.mot.com>
>> <4B795009.7050309@cisco.com>
>> <750BBC72E178114F9DC4872EBFF29A5B0A2CD9EA@ZMY16EXM66.ds.mot.com>
>> <4B7CA748.5000709@cisco.com>
>> <750BBC72E178114F9DC4872EBFF29A5B0A317F0A@ZMY16EXM66.ds.mot.com>
>>
>>
>>
>> Paul is correct. The real thrust of RFC 3265 is monitoring the state of
>> something. State becomes interesting at transitions (hence the focus on
>> events), but the events are inherently talking about the state of
>> something.
>>
>> In your case, I agree that it won't be call state. It appears to be
>> related to some combination of call diversion preferences and proxy (?)
>> processing state? I agree with Paul that the overall model is ill
>> defined (even after reading 24.604), so I can't provide any concrete
>> suggestions here. The best I can do is recommend you examine the
>> existing corpus of event packages and come to an understanding about how
>> they work:
>>
>> conference package IETF SIPPING Working Group [RFC4575]
>> consent-pending-additions package Gonzalo Camarillo [RFC5362]
>> dialog package IETF SIPPING Working Group [RFC4235]
>> kpml package Eric Burger [RFC4730]
>> message-summary package Rohan Mahy [RFC3842]
>> poc-settings package Miguel A. Garcia-Martin [RFC4354]
>> presence package Jonathan Rosenberg [RFC3856]
>> reg package Jonathan Rosenberg [RFC3680]
>> refer package Robert Sparks [RFC3515]
>> spirits-INDPs package Vijay K. Gurbani [RFC3910]
>> spirits-user-prof package Vijay K. Gurbani [RFC3910]
>> winfo template-package Jonathan Rosenberg [RFC3857]
>> xcap-diff package IETF Real-time Applications
>> [RFC-ietf-sip-xcapevent-08.txt]
>> and Infrastructure Area
>>
>> You'll note that every one of these event packages actually talks about
>> the *state* of something, and describes sending notifications when that
>> *state* changes. And, with the exception of the dialog event package
>> (and arguably the spirits-INDPs event package), you'll note that none
>> of these states are actual call state -- they're the state of the
>> resource as it pertains to the aspect that the event package monitors.
>>
>> For your proposed div-notification event package, you need to define
>> what this state is and how it changes. If you can't do that, then an
>> RFC3265 event package probably isn't the right tool for the job you have
>> in mind.
>>
>> /a
>>
>> On 2/18/10 11:08, Feb 18, Avasarala Ranjit-A20990 wrote:
>>> Hi Paul, Adam
>>>
>>> Since communication diversion notification information is not related to
>>> any state of the actual communication, I feel there is no need to
>>> maintain state for this. Also the Communication diversion notification
>>> message gets triggered after the communication diversion has occurred.
>>>
>>> The main intent of CDIVN is to notify the subscriber of a communication
>>> diversion that occurred on their behalf in the network. This
>>> notification is not part of any communication dialog or part of the
>>> communication state machine and is completely independent of the actual
>>> communication (or call).
>>>
>>> Comments?
>>>
>>> Thanks
>>>
>>> Regards
>>> Ranjit
>>>
>>> -----Original Message-----
>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] Sent: Thursday,
>>> February 18, 2010 8:05 AM
>>> To: Avasarala Ranjit-A20990
>>> Cc: Adam Roach; John-Luc Bakker
>>> Subject: Re: [dispatch] Preliminary version of Expert review
>>> ofdraft-avasarala-dispatch-comm-div-notification-03
>>>
>>> Ranjit,
>>>
>>> [ taking IESG off the dist list for now, and adding Adam as the sub/not
>>> author. Perhaps we should take this back to the dispatch list. ]
>>>
>>> I haven't had time to read through the revised document yet, but I
>>> skimmed through your comments. There is one that I think we need to talk
>>> about because it potentially has significant impact. (I've picked the
>>> first place it came up, but its also mentioned in other parts of my
>>> issues and your responses.)
>>>
>>> Adam: Is this enough for you to understand the issue?
>>>
>>> Avasarala Ranjit-A20990 wrote:
>>>
>>>>> ISSUE: The document details the notification body, and some inferences
>>>>> can be drawn about the state from the syntax of the notification
>>> body.
>>>>> But this is tenuous. There should be an explicit specification of the
>>>>> state maintained, and how it is mapped onto the notification body.
>>>> <Ranjit-Feb16> The notifications for communication diversions are sent
>>>> as and when they occur depending on the filter criteria set by the
>>>> subscribers. So there is no need of maintaining any state information
>>>> for the package.
>>> That doesn't fit my understanding of how event packages are defined.
>>>
>>> IIUC, the semantics of event packages are described relative to some
>>> sort of "state", not against another incoming event. (Whether this is
>>> literally true in the implementation or not.)
>>>
>>> So minimally, when a diversion happens, some particular information is
>>> culled from it, and becomes the new state. Then that causes
>>> notifications to the existing subscriptions.
>>>
>>> After that it either sticks around, or else the state transitions back
>>> to "none". But if it transitions back to "none" then that would trigger
>>> another round of notifications - which you clearly don't want.
>>>
>>> So it makes sense that it sticks around, in which case, when a new
>>> subscription comes in it will get a notification about that last
>>> diversion. (Unless there can be some sort of "implicit" filter that
>>> suppresses notification when the state becomes "none".)
>>>
>>> Adam - is that valid???
>>>
>>> Thanks,
>>> Paul
>>
>> _______________________________________________
>> dispatch mailing list
>> dispatch@ietf.org
>> https://www.ietf.org/mailman/listinfo/dispatch
>>
> _______________________________________________
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> dispatch@ietf.org
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> 

From ankriste@cisco.com  Fri Feb 19 16:28:01 2010
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On 2/19/2010 3:43 PM, Paul Kyzivat wrote:
>
>
> Anders Kristensen wrote:
>> What is the state model for KPML? Seems to me that that's just as
>> stateless as this CDIV package.
>
> Sure it does. The state consists of zero or more buffered DTMF characters.
>
> Depending on how you look at it, there may be separate state for each
> subscriber - which is populated with incoming dtmf characters - or else
> you can think of it as a single buffer with a separate starting point
> for each subscriber.

Well, to me all the talk of buffering feels just as a way to make it 
appear state oriented when really these are just events. ISTM that the 
KPML model of buffers could be applied to *any* non-state based event 
package to make it comply with the 3265 requirement that there must be 
some actual state underneath. (I haven't read the KPML spec in years so 
admittedly the details escape me.)

Would you say entity tags are a good match for KPML?

The notion of different state per observer or different views of the 
same state also doesn't make sense to me. There's just one resource 
being observed so it doesn't seem right that two subscriptions should 
have different views of the resource's state.

>
> The state is essential for the handling of resubscribes that change the
> filter.

Why can't a filter apply to events without there being any state (beyond 
event attributes)?

Thanks,
Anders

From zhouzp@huawei.com  Sat Feb 20 00:44:32 2010
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Date: Sat, 20 Feb 2010 16:46:05 +0800
From: zhipeng zhou <zhouzp@huawei.com>
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Hi Gabor and all,
I like to share my viewpoint for this excellent comment email.
Gabor has listed many valuable features related to the Recording device and
service.
 IMO, absolutely the major feature for recording is focused on the Media
interaction. For example the security can be an independent solution.
So from my point of view, it can focus on the media architecture and
related content flow.

 What is your opinion.
Thanks.
Zhipeng

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Subject: dispatch Digest, Vol 11, Issue 33

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Today's Topics:

   1. Comments on Session Recording Protocol (Gabor Moczar)


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

Message: 1
Date: Fri, 19 Feb 2010 18:56:47 +0100
From: Gabor Moczar <gmoczar@gmail.com>
Subject: [dispatch] Comments on Session Recording Protocol
To: dispatch@ietf.org
Message-ID:
	<6a2adf621002190956jae37f66i791acbf627095007@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Dear Working Group Members,

I am new to this list and Leon Portman from Nice suggested to send my
comments on the Session Recording Protocol to here directly.
I have 8+ years experience with IP based call recording solutions as a
solution architect and product manager.
Let me know your opinion about my comments.

1. Meta data
SRP should provide a standard mechanism, which allows sending meta data in
addition to the media to the RS. An extendible set of fields could describe
the recorded call in a detailed way including calling party number, name,
agent id, etc.
Currently most implementations require implementing an additional interface
using standard or proprietary CTI to obtain such kind of information. These
implementations quite unique across the various platforms and considered to
add additional complexity, OAM overhead and failure point to the overall
solution.

2. Handling multiple media streams
SRP should allow RS to select, which media stream should be recorded or not
for a given session.
As unified communication emerge, more and more customers using voice, video,
instant messaging, etc. in their communication with their partners and
clients. Customers may would like to decide, which media or which direction
of the media should be recorded during the session. For compliance, probably
only the voice part of a video conversation is sufficient too.

3. Security and encryption
The draft very well describes the sensitivity of the topic from security
point of view, however the term "encryption" is missing from the
requirements part. I see that "eavesdropping protection" is used, but what
is the purpose of not using the most obvious solution for "eavesdropping
protection".
I would also add: SRP should also allow the recording of encrypted calls.

4. Redundancy
SRP should provide a mechanism to allow RCs to send the media to multiple
RSs at one time.
Probably endpoints with limited resources or networks with restricted
bandwidth will not able to provide this feature, but in mission critical
environments this could be a very nice way to provide redundancy. The
failover mechanism described in the draft (and used by some of the vendors)
is not protected from single point of failure.

5. Negotiating media handling capabilities SRP should allow RC and RS to
negotiate audio/video codec capabilities (similar to SIP/SDP) and the RC
should send media in the agreed codec.
In most cases the codec will be exactly the same as the one used in the
recorded session. However, high definition audio and video codecs are
getting used in more and more sessions, for recording, the requirement in
most cases allow to record the sessions in a lower resolution format. Either
the RC, either the RS should provide transcoding capabilities. I know
several video endpoints, which are able to send the video in 2 formats at
one time (HD and CIF) and I can also imagine devices with on-board
transcoding capabilities in the future (e.g. transcoding AAC-LD to G.729 for
the RS) and some of the current implementations also allow the transcoding
of the recording streams using network resources.

6. Others
There are some points, which are mentioned in the use cases section, but no
requirement is provided to support the given feature:
- Silent monitoring and real-time applications: the entire mechanism should
allow the users to implement a real-time monitoring functionality, which
means that the media has to be transmitted with considerably low delay, in a
real-time fashion to the RS.
- Handling complex calls: SRP should provide a mechanism to allow the RS to
link together certain call segments in a complex call scenario (e.g.
providing a global call identifier as a part of the meta data).

Best regards,
Gabor
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version of Expert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Anders,

I think I will leave it to Adam, as the expert, to explain this.
I have found the state model the easiest way to conceptualize the 
behavior. I don't think there is any fundamental requirement that this 
force the unnecessary retention of state - its an abstract model rather 
than a concrete one.

In any case, for the draft at hand, I found that without such a 
description I can't fully understand the semantics of the package in a 
deterministic way.

Regarding kpml - it could have been described more clearly. I agree it 
suggests per-subscriber state, but as I noted earlier, it can also be 
understood in terms of common state. These views are functionally 
equivalent.

	Thanks,
	Paul

Anders Kristensen wrote:
> 
> On 2/19/2010 3:43 PM, Paul Kyzivat wrote:
>>
>>
>> Anders Kristensen wrote:
>>> What is the state model for KPML? Seems to me that that's just as
>>> stateless as this CDIV package.
>>
>> Sure it does. The state consists of zero or more buffered DTMF 
>> characters.
>>
>> Depending on how you look at it, there may be separate state for each
>> subscriber - which is populated with incoming dtmf characters - or else
>> you can think of it as a single buffer with a separate starting point
>> for each subscriber.
> 
> Well, to me all the talk of buffering feels just as a way to make it 
> appear state oriented when really these are just events. ISTM that the 
> KPML model of buffers could be applied to *any* non-state based event 
> package to make it comply with the 3265 requirement that there must be 
> some actual state underneath. (I haven't read the KPML spec in years so 
> admittedly the details escape me.)
> 
> Would you say entity tags are a good match for KPML?
> 
> The notion of different state per observer or different views of the 
> same state also doesn't make sense to me. There's just one resource 
> being observed so it doesn't seem right that two subscriptions should 
> have different views of the resource's state.
> 
>>
>> The state is essential for the handling of resubscribes that change the
>> filter.
> 
> Why can't a filter apply to events without there being any state (beyond 
> event attributes)?
> 
> Thanks,
> Anders
> 

From zhouzp@huawei.com  Sat Feb 20 18:42:42 2010
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Date: Sun, 21 Feb 2010 10:44:22 +0800
From: zhipeng zhou <zhouzp@huawei.com>
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Subject: [dispatch] A question on "draft-loreto-dispatch-disaggregated-media-01"
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Hi Salvatore and Gonzalo:
=20
After reading your newly uploaded
"draft-loreto-dispatch-disaggregated-media-01", esp. your section 4,=20
I have a question as following :
Since Bob's device will communicate with Alice's three devices, three
channels or sessions will be available to implement this scenario.
While the sentece in this document "Since this type of scenario is not
covered by existing mechanisms, we
   propose to initiate work on SIP extensions to support it." shows that =
you
like to implement this scenario through a new way.
So, may you pls give a rough diagram and  indicate the advantage of your
proposed new solution.
Thanks.
Zhipeng
=20

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E-Mail:  <mailto:zhouzp@huawei.com> zhouzp@huawei.com
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<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>Hi=20
</SPAN>Salvatore&nbsp;<SPAN class=3D168382702-21022010>and=20
Gonzalo:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN =
class=3D168382702-21022010></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>After reading =
your newly=20
uploaded "draft-loreto-dispatch-disaggregated-media-01", esp. your =
section 4,=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>I have a =
question&nbsp;as=20
following&nbsp;:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>Since Bob's =
device will=20
communicate with&nbsp;Alice's three devices, three channels or sessions =
will be=20
available to implement this scenario.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>While the =
sentece in this=20
document "Since this type of scenario is not covered by existing =
mechanisms,=20
we<BR>&nbsp;&nbsp; propose to initiate work on SIP extensions to support =
it."=20
shows that you like to implement this scenario through a new=20
way.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN class=3D168382702-21022010>So, may you pls =
give a=20
rough diagram and &nbsp;indicate the advantage of your proposed new=20
solution.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN =
class=3D168382702-21022010>Thanks.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial><SPAN =
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From ranjit@motorola.com  Mon Feb 22 00:55:44 2010
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, "Anders Kristensen" <ankriste@cisco.com>, "Adam Roach" <adam@nostrum.com>
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Hi all

Actually for the state information to be included, I can think of state
for a communication diversion notification information from either
subscriber point of view or from the notifier (server) point of view.

>From the subscriber point of view, the states could be whether the
subscriber is subscribed and if the notifications are being generated or
not. So here the state model would be

Subscribed->terminated

When in subscribed, the notifier can transition from idle to generate
notification and to send notification and finally to idle

Something like idle->generate->send->idle  OR idle->generate->idle
              =20
In IDLE state, the notifier waits for diversions to occur. Once a
diversion occurs, the notifier transitions to "GENERATE" state and looks
for any filter criteria. If it finds one and it matches, then it
generates the notification and transitions to "SEND" state. If there is
no filter criteria, then the notifier transitions back to "IDLE" state
and does not send any notification.
=20
Comments?

Regards
Ranjit

-----Original Message-----
From: dispatch-bounces@ietf.org [mailto:dispatch-bounces@ietf.org] On
Behalf Of Paul Kyzivat
Sent: Saturday, February 20, 2010 8:35 PM
To: Anders Kristensen
Cc: dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

Anders,

I think I will leave it to Adam, as the expert, to explain this.
I have found the state model the easiest way to conceptualize the
behavior. I don't think there is any fundamental requirement that this
force the unnecessary retention of state - its an abstract model rather
than a concrete one.

In any case, for the draft at hand, I found that without such a
description I can't fully understand the semantics of the package in a
deterministic way.

Regarding kpml - it could have been described more clearly. I agree it
suggests per-subscriber state, but as I noted earlier, it can also be
understood in terms of common state. These views are functionally
equivalent.

	Thanks,
	Paul

Anders Kristensen wrote:
>=20
> On 2/19/2010 3:43 PM, Paul Kyzivat wrote:
>>
>>
>> Anders Kristensen wrote:
>>> What is the state model for KPML? Seems to me that that's just as=20
>>> stateless as this CDIV package.
>>
>> Sure it does. The state consists of zero or more buffered DTMF=20
>> characters.
>>
>> Depending on how you look at it, there may be separate state for each

>> subscriber - which is populated with incoming dtmf characters - or=20
>> else you can think of it as a single buffer with a separate starting=20
>> point for each subscriber.
>=20
> Well, to me all the talk of buffering feels just as a way to make it=20
> appear state oriented when really these are just events. ISTM that the

> KPML model of buffers could be applied to *any* non-state based event=20
> package to make it comply with the 3265 requirement that there must be

> some actual state underneath. (I haven't read the KPML spec in years=20
> so admittedly the details escape me.)
>=20
> Would you say entity tags are a good match for KPML?
>=20
> The notion of different state per observer or different views of the=20
> same state also doesn't make sense to me. There's just one resource=20
> being observed so it doesn't seem right that two subscriptions should=20
> have different views of the resource's state.
>=20
>>
>> The state is essential for the handling of resubscribes that change=20
>> the filter.
>=20
> Why can't a filter apply to events without there being any state=20
> (beyond event attributes)?
>=20
> Thanks,
> Anders
>=20
_______________________________________________
dispatch mailing list
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From john.elwell@siemens-enterprise.com  Mon Feb 22 03:53:46 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Gabor Moczar <gmoczar@gmail.com>, "dispatch@ietf.org" <dispatch@ietf.org>
Date: Mon, 22 Feb 2010 12:55:41 +0100
Thread-Topic: [dispatch] Comments on Session Recording Protocol
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Gabor,

Just addressing a few of your points below:=20

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Gabor Moczar
> Sent: 19 February 2010 17:57
> To: dispatch@ietf.org
> Subject: [dispatch] Comments on Session Recording Protocol
>=20
> Dear Working Group Members,
>=20
> I am new to this list and Leon Portman from Nice suggested to=20
> send my comments on the Session Recording Protocol to here directly.
> I have 8+ years experience with IP based call recording=20
> solutions as a solution architect and product manager.
> Let me know your opinion about my comments.
>=20
> 1. Meta data=20
> SRP should provide a standard mechanism, which allows sending=20
> meta data in addition to the media to the RS. An extendible=20
> set of fields could describe the recorded call in a detailed=20
> way including calling party number, name, agent id, etc.=20
> Currently most implementations require implementing an=20
> additional interface using standard or proprietary CTI to=20
> obtain such kind of information. These implementations quite=20
> unique across the various platforms and considered to add=20
> additional complexity, OAM overhead and failure point to the=20
> overall solution.=20
>=20
> 2. Handling multiple media streams=20
> SRP should allow RS to select, which media stream should be=20
> recorded or not for a given session.=20
> As unified communication emerge, more and more customers=20
> using voice, video, instant messaging, etc. in their=20
> communication with their partners and clients. Customers may=20
> would like to decide, which media or which direction of the=20
> media should be recorded during the session. For compliance,=20
> probably only the voice part of a video conversation is=20
> sufficient too.=20
>=20
> 3. Security and encryption=20
> The draft very well describes the sensitivity of the topic=20
> from security point of view, however the term "encryption" is=20
> missing from the requirements part. I see that "eavesdropping=20
> protection" is used, but what is the purpose of not using the=20
> most obvious solution for "eavesdropping protection".=20
[JRE] Encryption does not necessarily give you protection against eavesdrop=
ping, e.g., if keys are too weak or are exchanged in inappropriate ways. I =
would say the requirement is for protection against eavesdropping.

> I would also add: SRP should also allow the recording of=20
> encrypted calls.=20
[JRE] Can you please clarify whether you mean:
- recording of media in the clear, said media having been transported to/fr=
om the communications partner in encrypted form; or
- recording of encrypted media, so that it can be played back only with kno=
wledge of the keys used for the original session?


>=20
> 4. Redundancy=20
> SRP should provide a mechanism to allow RCs to send the media=20
> to multiple RSs at one time.=20
> Probably endpoints with limited resources or networks with=20
> restricted bandwidth will not able to provide this feature,=20
> but in mission critical environments this could be a very=20
> nice way to provide redundancy. The failover mechanism=20
> described in the draft (and used by some of the vendors) is=20
> not protected from single point of failure.=20
>=20
> 5. Negotiating media handling capabilities=20
> SRP should allow RC and RS to negotiate audio/video codec=20
> capabilities (similar to SIP/SDP) and the RC should send=20
> media in the agreed codec.=20
> In most cases the codec will be exactly the same as the one=20
> used in the recorded session. However, high definition audio=20
> and video codecs are getting used in more and more sessions,=20
> for recording, the requirement in most cases allow to record=20
> the sessions in a lower resolution format. Either the RC,=20
> either the RS should provide transcoding capabilities. I know=20
> several video endpoints, which are able to send the video in=20
> 2 formats at one time (HD and CIF) and I can also imagine=20
> devices with on-board transcoding capabilities in the future=20
> (e.g. transcoding AAC-LD to G.729 for the RS) and some of the=20
> current implementations also allow the transcoding of the=20
> recording streams using network resources.=20
[JRE] Yes, but bear in mind that some devices might not have resources to d=
ecode and re-encode. So although we could have a requirement that allows th=
e possibility of different codecs, I think there also has to be a requireme=
nt to support devices that are unable to transcode.

>=20
> 6. Others=20
> There are some points, which are mentioned in the use cases=20
> section, but no requirement is provided to support the given feature:=20
> - Silent monitoring and real-time applications: the entire=20
> mechanism should allow the users to implement a real-time=20
> monitoring functionality, which means that the media has to=20
> be transmitted with considerably low delay, in a real-time=20
> fashion to the RS.=20
[JRE] I think we need to be clear whether this is within scope, and if so I=
 would agree that some sort of requirement on timing would be appropriate.

> - Handling complex calls: SRP should provide a mechanism to=20
> allow the RS to link together certain call segments in a=20
> complex call scenario (e.g. providing a global call=20
> identifier as a part of the meta data).=20
[JRE] This is a topic that has arisen in other contexts, and if a solution =
is to be provided I would imagine it would fall outside the scope of the SR=
P work.

John

From scottlawrenc@avaya.com  Mon Feb 22 06:04:38 2010
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From: "Scott Lawrence" <scottlawrenc@avaya.com>
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On Fri, 2010-02-19 at 18:56 +0100, Gabor Moczar wrote:
> 
> 3. Security and encryption  

> The draft very well describes the sensitivity of the topic from
> security point of view, however the term "encryption" is missing from
> the requirements part. I see that "eavesdropping protection" is used,
> but what is the purpose of not using the most obvious solution for
> "eavesdropping protection".

Part of what you're encountering here is just the difference in
preferred language.  This is a requirements document, not a solution
document - it is laying out what goals a solution much achieve, and it
is generally preferred that such a document _not_ specify any more than
needed about how the goals are met.

Encryption technologies can be used in many different ways to achieve
different ends, so saying "encryption" in the context of this document
would not actually specify which of those ends the encryption technology
was to be employed for.  By writing the requirement as "confidentiality
protection" that is made explicit, and there really isn't any other way
to meet that requirement (in the Internet) than encryption, but writing
it this way is actually more precise.

>  I would also add: SRP should also allow the recording of encrypted
> calls. 

There are a bunch of potential landmines in this area in that the IETF
is very careful not to specify anything that allows an intermediate
party to penetrate end-to-end confidentiality (there's that word again)
without the knowledge and cooperation of the parties.  That's not to say
that at least some interpretations of your requirement can't be met, but
it does mean that the requirement must be carefully constructed to avoid
creating problems.



From pkyzivat@cisco.com  Mon Feb 22 16:27:39 2010
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Avasarala Ranjit-A20990 wrote:
> Hi all
> 
> Actually for the state information to be included, I can think of state
> for a communication diversion notification information from either
> subscriber point of view or from the notifier (server) point of view.

I really think the state ought to be per subscribed resource (identified 
by the R-URI of the SUBSCRIBE when it reaches the notifier plus the 
event type.) The subscriber point of view really is a *view* of that. If 
it differs from one subscriber to another, then that really ought to be 
considered a filter, whether an explicit one supplied by the subscriber 
or an implicit one provided by the notifier based on policy.

>>From the subscriber point of view, the states could be whether the
> subscriber is subscribed and if the notifications are being generated or
> not. So here the state model would be
> 
> Subscribed->terminated
> 
> When in subscribed, the notifier can transition from idle to generate
> notification and to send notification and finally to idle
> 
> Something like idle->generate->send->idle  OR idle->generate->idle
>                
> In IDLE state, the notifier waits for diversions to occur. Once a
> diversion occurs, the notifier transitions to "GENERATE" state and looks
> for any filter criteria. If it finds one and it matches, then it
> generates the notification and transitions to "SEND" state. If there is
> no filter criteria, then the notifier transitions back to "IDLE" state
> and does not send any notification.

I think you are not understanding the kind of state model we are talking 
about.

Think of the state as a collection of data that may change over time, 
that is visible to the notifier. When something in the state changes in 
that state, the notifier sends a NOTIFY to subscribers, saying "the 
state of the resource changed, here is the new value."

In the simplistic form, you could view that state as exactly the data 
needed to populate the body of your NOTIFY.

It gets a more complicated if there can be multiple subscribers, and you 
don't allow them all to see all of the data. (I'm not certain if you 
have restrictions like that or not.) If you do, then there might be data 
in the state that come subscribers get and others do not.

In your case, it appears from your xml, that your state consists of zero 
or more diversions. (Specifically it can be more than one.) And for each 
there an originating URI, a diverting URI, a diverted-to URI, a 
diverting time, and a diverting reason.

But the diverting-URI is redundant - its equivalent to the resource that 
is the target of the subscription, and so will be constant throughout 
the subscription. I don't understand why it is present. My suspicion is 
that you have in mind that the server will serve more than one diverting 
user and so the *server* state consists of info for each diverting user 
it serves. But unless you allow a subscription to address the *server* 
rather than a particular diverting user, this is invisible to the state 
of a given resource subscription target.

If I have that right, then this state changes over time as calls are 
diverted. In particular, a new diversion is added. Then, simplistically, 
each notify would contain not only the most recent diversion, but all 
the prior diversions as well.

Or you could define it so that older diversions are removed, using any 
of a variety of algorithms. E.g. you might keep the most recent N of 
them, or all that have occurred in the last N minutes. (E.g. you might 
keep only the single most recent one.) But that does imply that somebody 
subscribing *after* a diversion has occurred will get a notification 
immediately with whatever that state is - say the most recent diversion, 
even if that occurred a long time ago. (This could be a feature, but the 
implementation needs to support it by retaining the state.)

It starts to get messy if you don't want to retain the state. Then you 
must pretend that the new diversion was added to the (previously empty) 
state, caused notification(s), and then was immediately removed from the 
state. If viewed that way, the removal would also cause a notification. 
I suggested how that might be handled, but Adam thought it improper.

	Thanks,
	Paul

From ranjit@motorola.com  Tue Feb 23 02:52:02 2010
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thread-topic: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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References: <4B7EF1DA.60002@cisco.com> <4B7F1368.6040603@cisco.com><4B7F223D.3080008@cisco.com> <4B7F2CFD.8070901@cisco.com> <4B7FFA2B.5080303@cisco.com> <750BBC72E178114F9DC4872EBFF29A5B0A31824D@ZMY16EXM66.ds.mot.com> <4B832169.8090100@cisco.com>
From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Hi Paul

My comments <Ranjit-Feb23>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, February 23, 2010 5:59 AM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]



Avasarala Ranjit-A20990 wrote:
> Hi all
>=20
> Actually for the state information to be included, I can think of=20
> state for a communication diversion notification information from=20
> either subscriber point of view or from the notifier (server) point of
view.

I really think the state ought to be per subscribed resource (identified
by the R-URI of the SUBSCRIBE when it reaches the notifier plus the
event type.) The subscriber point of view really is a *view* of that. If
it differs from one subscriber to another, then that really ought to be
considered a filter, whether an explicit one supplied by the subscriber
or an implicit one provided by the notifier based on policy.

>>From the subscriber point of view, the states could be whether the
> subscriber is subscribed and if the notifications are being generated=20
> or not. So here the state model would be
>=20
> Subscribed->terminated
>=20
> When in subscribed, the notifier can transition from idle to generate=20
> notification and to send notification and finally to idle
>=20
> Something like idle->generate->send->idle  OR idle->generate->idle
>               =20
> In IDLE state, the notifier waits for diversions to occur. Once a=20
> diversion occurs, the notifier transitions to "GENERATE" state and=20
> looks for any filter criteria. If it finds one and it matches, then it

> generates the notification and transitions to "SEND" state. If there=20
> is no filter criteria, then the notifier transitions back to "IDLE"=20
> state and does not send any notification.

I think you are not understanding the kind of state model we are talking
about.

Think of the state as a collection of data that may change over time,
that is visible to the notifier. When something in the state changes in
that state, the notifier sends a NOTIFY to subscribers, saying "the
state of the resource changed, here is the new value."

<Ranjit-Feb23> In case of CDIV, the notifier triggers a NOTIFY when ever
a diversion occurs and the filter criteria matches. So there is no
resource as such whose value changes - except that notifier can maintain
the count of number of diversions that have occurred and how many times
the filter criteria matched and how many notifications were sent. So
every time a notification is sent, these values get incremented.

In the simplistic form, you could view that state as exactly the data
needed to populate the body of your NOTIFY.
<Ranjit-Feb23> Then the state needs to contain the caller details,
diversion time, reason for diversion and if filter criteria matched or
not.

It gets a more complicated if there can be multiple subscribers, and you
don't allow them all to see all of the data. (I'm not certain if you
have restrictions like that or not.) If you do, then there might be data
in the state that come subscribers get and others do not.
<Ranjit-Feb23> No need of such a thing because a particular subscriber
can only subscribe to his or her diversions and not others.

In your case, it appears from your xml, that your state consists of zero
or more diversions. (Specifically it can be more than one.) And for each
there an originating URI, a diverting URI, a diverted-to URI, a
diverting time, and a diverting reason.
<Ranjit-Feb23> could be.=20

But the diverting-URI is redundant - its equivalent to the resource that
is the target of the subscription, and so will be constant throughout
the subscription. I don't understand why it is present. My suspicion is
that you have in mind that the server will serve more than one diverting
user and so the *server* state consists of info for each diverting user
it serves. But unless you allow a subscription to address the *server*
rather than a particular diverting user, this is invisible to the state
of a given resource subscription target.
<Ranjit-Feb23> Yes there is no need of diverting-URI.=20

If I have that right, then this state changes over time as calls are
diverted. In particular, a new diversion is added. Then, simplistically,
each notify would contain not only the most recent diversion, but all
the prior diversions as well.
<Ranjit-Feb23> the set of previous diversions can be taken care of by
use of History-Info header. Each notification will contain information
about only one diversion which gets triggered when the diversion occurs.

Or you could define it so that older diversions are removed, using any
of a variety of algorithms. E.g. you might keep the most recent N of
them, or all that have occurred in the last N minutes. (E.g. you might
keep only the single most recent one.) But that does imply that somebody
subscribing *after* a diversion has occurred will get a notification
immediately with whatever that state is - say the most recent diversion,
even if that occurred a long time ago. (This could be a feature, but the
implementation needs to support it by retaining the state.)

It starts to get messy if you don't want to retain the state. Then you
must pretend that the new diversion was added to the (previously empty)
state, caused notification(s), and then was immediately removed from the
state. If viewed that way, the removal would also cause a notification.=20
I suggested how that might be handled, but Adam thought it improper.
<Ranjit-Feb23> ok.

	Thanks,
	Paul

From pkyzivat@cisco.com  Tue Feb 23 06:08:47 2010
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Inline. I've trimmed out irrelevant stuff

Avasarala Ranjit-A20990 wrote:
> Hi Paul
> 
> My comments <Ranjit-Feb23> 
> 
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 

> I think you are not understanding the kind of state model we are talking
> about.
> 
> Think of the state as a collection of data that may change over time,
> that is visible to the notifier. When something in the state changes in
> that state, the notifier sends a NOTIFY to subscribers, saying "the
> state of the resource changed, here is the new value."
> 
> <Ranjit-Feb23> In case of CDIV, the notifier triggers a NOTIFY when ever
> a diversion occurs and the filter criteria matches. So there is no
> resource as such whose value changes - except that notifier can maintain
> the count of number of diversions that have occurred and how many times
> the filter criteria matched and how many notifications were sent. So
> every time a notification is sent, these values get incremented.

I'm trying to find another way to conceptualize your situation - one 
that works better with the 3265 way of describing events.

> In the simplistic form, you could view that state as exactly the data
> needed to populate the body of your NOTIFY.
> <Ranjit-Feb23> Then the state needs to contain the caller details,
> diversion time, reason for diversion and if filter criteria matched or
> not.
> 
> It gets a more complicated if there can be multiple subscribers, and you
> don't allow them all to see all of the data. (I'm not certain if you
> have restrictions like that or not.) If you do, then there might be data
> in the state that come subscribers get and others do not.
> <Ranjit-Feb23> No need of such a thing because a particular subscriber
> can only subscribe to his or her diversions and not others.

OK. Then we won't worry about that.

> In your case, it appears from your xml, that your state consists of zero
> or more diversions. (Specifically it can be more than one.) And for each
> there an originating URI, a diverting URI, a diverted-to URI, a
> diverting time, and a diverting reason.
> <Ranjit-Feb23> could be. 
> 
> But the diverting-URI is redundant - its equivalent to the resource that
> is the target of the subscription, and so will be constant throughout
> the subscription. I don't understand why it is present. My suspicion is
> that you have in mind that the server will serve more than one diverting
> user and so the *server* state consists of info for each diverting user
> it serves. But unless you allow a subscription to address the *server*
> rather than a particular diverting user, this is invisible to the state
> of a given resource subscription target.
> <Ranjit-Feb23> Yes there is no need of diverting-URI. 

OK. Then it could be removed from the notification body, and the filter 
too, because it is redundant. Right?

Its being there confuses people like me, who assume that if its there it 
must have a purpose.

> If I have that right, then this state changes over time as calls are
> diverted. In particular, a new diversion is added. Then, simplistically,
> each notify would contain not only the most recent diversion, but all
> the prior diversions as well.
> <Ranjit-Feb23> the set of previous diversions can be taken care of by
> use of History-Info header.

You are thinking of something different than I am.
You are talking about previous diversions of the *same call*.
I'm talking about diversions of previous calls by the same diverting user.

IOW,
- at 2:00 a call from alice was diverted to bob.
- at 3:00 a call from carol was diverted to ted.
...

Its certainly possible to *imagine* such a state, gradually increasing 
with time as other calls are diverted.

> Each notification will contain information
> about only one diversion which gets triggered when the diversion occurs.

So, if my phone has been *off*, when I turn it on I won't get a 
notification about a diversion that occurred while it was off? Is that a 
feature, or a limitation of the design?

If notifications of diversions are only delivered in realtime as they 
occur, then why do you bother to include the time of the diversion in 
the notification?

> Or you could define it so that older diversions are removed, using any
> of a variety of algorithms. E.g. you might keep the most recent N of
> them, or all that have occurred in the last N minutes. (E.g. you might
> keep only the single most recent one.) But that does imply that somebody
> subscribing *after* a diversion has occurred will get a notification
> immediately with whatever that state is - say the most recent diversion,
> even if that occurred a long time ago. (This could be a feature, but the
> implementation needs to support it by retaining the state.)
> 
> It starts to get messy if you don't want to retain the state. Then you
> must pretend that the new diversion was added to the (previously empty)
> state, caused notification(s), and then was immediately removed from the
> state. If viewed that way, the removal would also cause a notification. 
> I suggested how that might be handled, but Adam thought it improper.
> <Ranjit-Feb23> ok.

The trick here is to find a way to model the state, and the implicit and 
explicit filters, to achieve the sort of results you want and permit the 
sort of implementation you want. There are no doubt several ways that 
may work.

I think what you need is something like:

- the state consists (abstractly) of x, y, z
- the state is adjusted in the following way when
   another call is diverted
- the manner of matching a particular component in a
   filter against the state is ...
- the manner of dealing with multiple components in
   a filter is ...
- the manner of populating the notification body from
   the state is ...

	Thanks,
	Paul

From salvatore.loreto@ericsson.com  Tue Feb 23 06:16:19 2010
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Date: Tue, 23 Feb 2010 16:18:14 +0200
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [dispatch] A question on	"draft-loreto-dispatch-disaggregated-media-01"
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Hi Zhipeng,

thanks for the question.

The advantage of the proposed solution is that the user does not need a 
device that is able to act as a controller:

indeed the proposed solution give the possibility for a user to create a 
multimedia session combining different media streams,
coming from different devices under his/her control (where none of those 
devices can act as a controller
for the other devices in the call), so that they are treated by the far 
end of the session as a single media session.

cheers
/Sal




On 02/21/2010 04:44 AM, zhipeng zhou wrote:
> Hi Salvatore and Gonzalo:
> After reading your newly uploaded 
> "draft-loreto-dispatch-disaggregated-media-01", esp. your section 4,
> I have a question as following :
> Since Bob's device will communicate with Alice's three devices, three 
> channels or sessions will be available to implement this scenario.
> While the sentece in this document "Since this type of scenario is not 
> covered by existing mechanisms, we
>    propose to initiate work on SIP extensions to support it." shows 
> that you like to implement this scenario through a new way.
> So, may you pls give a rough diagram and  indicate the advantage of 
> your proposed new solution.
> Thanks.
> Zhipeng
>
> -----------------------------------------------------
>
> Huawei Software Technologies Co.,Ltd.
>
> Floor 2, Building A, NO.48,Ning Nan AV., Nanjing , P.R.of China
> Zipcode:210012
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> Fax:(+86) 25-82276760
>
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>
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--------------070900010004000203010004
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Hi Zhipeng,<br>
<br>
thanks for the question.<br>
<br>
The advantage of the proposed solution is that the user does not need a
device that is able to act as a controller:<br>
<br>
indeed the proposed solution give the possibility for a user to create
a multimedia session combining different media streams,<br>
coming from different devices under his/her control (where none of
those devices can act as a controller<br>
for the other devices in the call), so that they are treated by the far
end of the session as a single media session.<br>
<br>
cheers<br>
/Sal<br>
<br>
<br>
<br>
<br>
On 02/21/2010 04:44 AM, zhipeng zhou wrote:
<blockquote cite="mid:000301cab29f$c6651d20$1856a80a@china.huawei.com"
 type="cite">
  <meta http-equiv="Context-Type" content="text/html; charset=gb2312">
  <div> <span>Hi </span>Salvatore&nbsp;<span>and Gonzalo:</span> </div>
  <div> <span></span> &nbsp;</div>
  <div> <span>After reading your newly uploaded
"draft-loreto-dispatch-disaggregated-media-01", esp. your section 4, </span>
  </div>
  <div> <span>I have a question&nbsp;as following&nbsp;:</span> </div>
  <div> <span>Since Bob's device will communicate with&nbsp;Alice's three
devices, three channels or sessions will be available to implement this
scenario.</span> </div>
  <div> <span>While the sentece in this document "Since this type of
scenario is not covered by existing mechanisms, we<br>
&nbsp;&nbsp; propose to initiate work on SIP extensions to support it." shows
that you like to implement this scenario through a new way.</span> </div>
  <div> <span>So, may you pls give a rough diagram and &nbsp;indicate the
advantage of your proposed new solution.</span> </div>
  <div> <span>Thanks.</span> </div>
  <div> <span>Zhipeng</span> </div>
  <div> &nbsp;</div>
  <object></object>
  <div>
  <p><span lang="EN-US"></span><span lang="EN-US">-----------------------------------------------------</span><span
 lang="EN-US"> </span></p>
  <p><span lang="EN-US">Huawei Software Technologies Co.<span>,Ltd</span>.
  </span></p>
  <p><span lang="EN-US">Floor 2, Building A, NO.48</span>&#65292;<span><span
 lang="EN-US">Ning</span></span><span lang="EN-US"> Nan <span>AV.<span>,
Nanjing , </span>P.R.of</span> China <br>
Zipcode:210012<br>
E-Mail: <a moz-do-not-send="true"
 title="mailto:zhangrenzhou@huawei.com" href="mailto:zhouzp@huawei.com"><span>zhouzp@huawei.com</span></a><br>
Phone:(+86) 25-82276771<br>
Fax:(+86) 25-82276760</span><span lang="EN-US"> </span></p>
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13404162849<br>
-----------------------------------------------------</span><span
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  <p><span lang="EN-US"> &nbsp; </span></p>
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 lang="EN-US"> </span></p>
  <p>&#65288;&#21253;&#25324;&#20294;&#19981;&#38480;&#20110;&#20840;&#37096;&#25110;&#37096;&#20998;&#22320;&#27844;&#38706;&#12289;&#22797;&#21046;&#12289;&#25110;&#25955;&#21457;&#65289;&#26412;&#37038;&#20214;&#20013;&#30340;&#20449;&#24687;&#12290;<span lang="EN-US"> </span></p>
  <p><span lang="EN-US"><span>&nbsp;&nbsp;&nbsp; </span></span>&#22914;&#26524;&#24744;&#38169;&#25910;&#20102;&#26412;&#37038;&#20214;&#65292;&#35831;&#24744;&#31435;&#21363;&#30005;&#35805;&#25110;&#37038;&#20214;&#36890;&#30693;&#21457;&#20214;
&#20154;&#24182;&#21024;&#38500;&#26412;&#37038;&#20214;&#65281;<span lang="EN-US"><br>
***************************************************************************************************************</span></p>
  <p><span lang="EN-US"><br>
***************************************************************************************************************<br>
  <span>&nbsp;&nbsp;&nbsp; </span>This e-mail and its attachments contain
confidential information from HUAWEI, which is intended only for the</span><span
 lang="EN-US"> </span></p>
  <p><span lang="EN-US">person or entity whose address is listed above.
Any use of the information contained herein in any way(including,</span><span
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thread-topic: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi Paul.

Thanks for the comments. <Ranjit-Feb23-2>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Tuesday, February 23, 2010 7:40 PM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]

Inline. I've trimmed out irrelevant stuff

Avasarala Ranjit-A20990 wrote:
> Hi Paul
>=20
> My comments <Ranjit-Feb23>
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]

> I think you are not understanding the kind of state model we are=20
> talking about.
>=20
> Think of the state as a collection of data that may change over time,=20
> that is visible to the notifier. When something in the state changes=20
> in that state, the notifier sends a NOTIFY to subscribers, saying "the

> state of the resource changed, here is the new value."
>=20
> <Ranjit-Feb23> In case of CDIV, the notifier triggers a NOTIFY when=20
> ever a diversion occurs and the filter criteria matches. So there is=20
> no resource as such whose value changes - except that notifier can=20
> maintain the count of number of diversions that have occurred and how=20
> many times the filter criteria matched and how many notifications were

> sent. So every time a notification is sent, these values get
incremented.

I'm trying to find another way to conceptualize your situation - one
that works better with the 3265 way of describing events.

> In the simplistic form, you could view that state as exactly the data=20
> needed to populate the body of your NOTIFY.
> <Ranjit-Feb23> Then the state needs to contain the caller details,=20
> diversion time, reason for diversion and if filter criteria matched or

> not.
>=20
> It gets a more complicated if there can be multiple subscribers, and=20
> you don't allow them all to see all of the data. (I'm not certain if=20
> you have restrictions like that or not.) If you do, then there might=20
> be data in the state that come subscribers get and others do not.
> <Ranjit-Feb23> No need of such a thing because a particular subscriber

> can only subscribe to his or her diversions and not others.

OK. Then we won't worry about that.

> In your case, it appears from your xml, that your state consists of=20
> zero or more diversions. (Specifically it can be more than one.) And=20
> for each there an originating URI, a diverting URI, a diverted-to URI,

> a diverting time, and a diverting reason.
> <Ranjit-Feb23> could be.=20
>=20
> But the diverting-URI is redundant - its equivalent to the resource=20
> that is the target of the subscription, and so will be constant=20
> throughout the subscription. I don't understand why it is present. My=20
> suspicion is that you have in mind that the server will serve more=20
> than one diverting user and so the *server* state consists of info for

> each diverting user it serves. But unless you allow a subscription to=20
> address the *server* rather than a particular diverting user, this is=20
> invisible to the state of a given resource subscription target.
> <Ranjit-Feb23> Yes there is no need of diverting-URI.=20

OK. Then it could be removed from the notification body, and the filter
too, because it is redundant. Right?

Its being there confuses people like me, who assume that if its there it
must have a purpose.

> If I have that right, then this state changes over time as calls are=20
> diverted. In particular, a new diversion is added. Then,=20
> simplistically, each notify would contain not only the most recent=20
> diversion, but all the prior diversions as well.
> <Ranjit-Feb23> the set of previous diversions can be taken care of by=20
> use of History-Info header.

You are thinking of something different than I am.
You are talking about previous diversions of the *same call*.
I'm talking about diversions of previous calls by the same diverting
user.

IOW,
- at 2:00 a call from alice was diverted to bob.
- at 3:00 a call from carol was diverted to ted.
...
<Ranjit-Feb23-2> Ok here as you said, the notification always contains
the most recent and latest diversion that occurred and the one that
matched the subscriber's filter criteria.=20

Its certainly possible to *imagine* such a state, gradually increasing
with time as other calls are diverted.

> Each notification will contain information about only one diversion=20
> which gets triggered when the diversion occurs.

So, if my phone has been *off*, when I turn it on I won't get a
notification about a diversion that occurred while it was off? Is that a
feature, or a limitation of the design?
<Ranjit-Feb23-2> The notifications are temporarily stored in the server
when the phone is *off* and they would be sent one by one when the phone
is *on*. So yes it's a feature in the server to monitor phone's status
and store the notifications if required - the amount of storage could
depend on server policy and some may or may not. No hard rule that they
should. If they do not store, then only the latest notification would be
sent.

If notifications of diversions are only delivered in realtime as they
occur, then why do you bother to include the time of the diversion in
the notification?
<Ranjit-Feb23-2> true. This makes sense only when notifications are
stored and sent later (phone off scenario)

> Or you could define it so that older diversions are removed, using any

> of a variety of algorithms. E.g. you might keep the most recent N of=20
> them, or all that have occurred in the last N minutes. (E.g. you might

> keep only the single most recent one.) But that does imply that=20
> somebody subscribing *after* a diversion has occurred will get a=20
> notification immediately with whatever that state is - say the most=20
> recent diversion, even if that occurred a long time ago. (This could=20
> be a feature, but the implementation needs to support it by retaining=20
> the state.)
>=20
> It starts to get messy if you don't want to retain the state. Then you

> must pretend that the new diversion was added to the (previously=20
> empty) state, caused notification(s), and then was immediately removed

> from the state. If viewed that way, the removal would also cause a
notification.
> I suggested how that might be handled, but Adam thought it improper.
> <Ranjit-Feb23> ok.

The trick here is to find a way to model the state, and the implicit and
explicit filters, to achieve the sort of results you want and permit the
sort of implementation you want. There are no doubt several ways that
may work.

I think what you need is something like:

- the state consists (abstractly) of x, y, z
- the state is adjusted in the following way when
   another call is diverted
- the manner of matching a particular component in a
   filter against the state is ...
- the manner of dealing with multiple components in
   a filter is ...
- the manner of populating the notification body from
   the state is ...

<Ranjit-Feb23-2> need to think further on this.

	Thanks,
	Paul

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Date: Tue, 23 Feb 2010 17:47:57 +0100
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From: Gabor Moczar <gmoczar@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] Comments on Session Recording Protocol
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THX John
See my answers below.

On Mon, Feb 22, 2010 at 12:55 PM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> Gabor,
>
> Just addressing a few of your points below:
>
> > -----Original Message-----
> > From: dispatch-bounces@ietf.org
> > [mailto:dispatch-bounces@ietf.org] On Behalf Of Gabor Moczar
> > Sent: 19 February 2010 17:57
> > To: dispatch@ietf.org
> > Subject: [dispatch] Comments on Session Recording Protocol
> >
> > Dear Working Group Members,
> >
> > I am new to this list and Leon Portman from Nice suggested to
> > send my comments on the Session Recording Protocol to here directly.
> > I have 8+ years experience with IP based call recording
> > solutions as a solution architect and product manager.
> > Let me know your opinion about my comments.
> >
> > 1. Meta data
> > SRP should provide a standard mechanism, which allows sending
> > meta data in addition to the media to the RS. An extendible
> > set of fields could describe the recorded call in a detailed
> > way including calling party number, name, agent id, etc.
> > Currently most implementations require implementing an
> > additional interface using standard or proprietary CTI to
> > obtain such kind of information. These implementations quite
> > unique across the various platforms and considered to add
> > additional complexity, OAM overhead and failure point to the
> > overall solution.
> >
> > 2. Handling multiple media streams
> > SRP should allow RS to select, which media stream should be
> > recorded or not for a given session.
> > As unified communication emerge, more and more customers
> > using voice, video, instant messaging, etc. in their
> > communication with their partners and clients. Customers may
> > would like to decide, which media or which direction of the
> > media should be recorded during the session. For compliance,
> > probably only the voice part of a video conversation is
> > sufficient too.
> >
> > 3. Security and encryption
> > The draft very well describes the sensitivity of the topic
> > from security point of view, however the term "encryption" is
> > missing from the requirements part. I see that "eavesdropping
> > protection" is used, but what is the purpose of not using the
> > most obvious solution for "eavesdropping protection".
> [JRE] Encryption does not necessarily give you protection against
> eavesdropping, e.g., if keys are too weak or are exchanged in inappropriate
> ways. I would say the requirement is for protection against eavesdropping.
>
Ok. I was thinking on similar mechanism like the one used in SIP with TLS
and SRTP.

>
> > I would also add: SRP should also allow the recording of
> > encrypted calls.
> [JRE] Can you please clarify whether you mean:
> - recording of media in the clear, said media having been transported
> to/from the communications partner in encrypted form; or
> - recording of encrypted media, so that it can be played back only with
> knowledge of the keys used for the original session?
>
> The second one. After reading Scott's comment, probably it cannot be easily
incorporated into the standard, but the requirements is a real one. We had
several customers asking for this feature...

>
> >
> > 4. Redundancy
> > SRP should provide a mechanism to allow RCs to send the media
> > to multiple RSs at one time.
> > Probably endpoints with limited resources or networks with
> > restricted bandwidth will not able to provide this feature,
> > but in mission critical environments this could be a very
> > nice way to provide redundancy. The failover mechanism
> > described in the draft (and used by some of the vendors) is
> > not protected from single point of failure.
> >
> > 5. Negotiating media handling capabilities
> > SRP should allow RC and RS to negotiate audio/video codec
> > capabilities (similar to SIP/SDP) and the RC should send
> > media in the agreed codec.
> > In most cases the codec will be exactly the same as the one
> > used in the recorded session. However, high definition audio
> > and video codecs are getting used in more and more sessions,
> > for recording, the requirement in most cases allow to record
> > the sessions in a lower resolution format. Either the RC,
> > either the RS should provide transcoding capabilities. I know
> > several video endpoints, which are able to send the video in
> > 2 formats at one time (HD and CIF) and I can also imagine
> > devices with on-board transcoding capabilities in the future
> > (e.g. transcoding AAC-LD to G.729 for the RS) and some of the
> > current implementations also allow the transcoding of the
> > recording streams using network resources.
> [JRE] Yes, but bear in mind that some devices might not have resources to
> decode and re-encode. So although we could have a requirement that allows
> the possibility of different codecs, I think there also has to be a
> requirement to support devices that are unable to transcode.
>
> Agree.

> >
> > 6. Others
> > There are some points, which are mentioned in the use cases
> > section, but no requirement is provided to support the given feature:
> > - Silent monitoring and real-time applications: the entire
> > mechanism should allow the users to implement a real-time
> > monitoring functionality, which means that the media has to
> > be transmitted with considerably low delay, in a real-time
> > fashion to the RS.
> [JRE] I think we need to be clear whether this is within scope, and if so I
> would agree that some sort of requirement on timing would be appropriate.
>
> > - Handling complex calls: SRP should provide a mechanism to
> > allow the RS to link together certain call segments in a
> > complex call scenario (e.g. providing a global call
> > identifier as a part of the meta data).
> [JRE] This is a topic that has arisen in other contexts, and if a solution
> is to be provided I would imagine it would fall outside the scope of the SRP
> work.
>
I think, it is a very important feature, but I am aware of
the differences among the various platforms/vendors, so probably not all of
them can meet this requirement. If this feature is not included in the
standard, recorder vendors will be forced to implement or use existing
TAPI/JTAPI/DMCC/etc. interfaces, which should be avoided in my opinion...

>
> John
>

--001636459320d2ef390480474fcc
Content-Type: text/html; charset=ISO-8859-1
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<div>THX John</div><div>See my answers below.</div><br><div class=3D"gmail_=
quote">On Mon, Feb 22, 2010 at 12:55 PM, Elwell, John <span dir=3D"ltr">&lt=
;<a href=3D"mailto:john.elwell@siemens-enterprise.com" target=3D"_blank">jo=
hn.elwell@siemens-enterprise.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Gabor,<br>
<br>
Just addressing a few of your points below:<br>
<div><div></div><div><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank">d=
ispatch-bounces@ietf.org</a><br>
&gt; [mailto:<a href=3D"mailto:dispatch-bounces@ietf.org" target=3D"_blank"=
>dispatch-bounces@ietf.org</a>] On Behalf Of Gabor Moczar<br>
&gt; Sent: 19 February 2010 17:57<br>
&gt; To: <a href=3D"mailto:dispatch@ietf.org" target=3D"_blank">dispatch@ie=
tf.org</a><br>
&gt; Subject: [dispatch] Comments on Session Recording Protocol<br>
&gt;<br>
&gt; Dear Working Group Members,<br>
&gt;<br>
&gt; I am new to this list and Leon Portman from Nice suggested to<br>
&gt; send my comments on the Session Recording Protocol to here directly.<b=
r>
&gt; I have 8+ years experience with IP based call recording<br>
&gt; solutions as a solution architect and product manager.<br>
&gt; Let me know your opinion about my comments.<br>
&gt;<br>
&gt; 1. Meta data<br>
&gt; SRP should provide a standard mechanism, which allows sending<br>
&gt; meta data in addition to the media to the RS. An extendible<br>
&gt; set of fields could describe the recorded call in a detailed<br>
&gt; way including calling party number, name, agent id, etc.<br>
&gt; Currently most implementations require implementing an<br>
&gt; additional interface using standard or proprietary CTI to<br>
&gt; obtain such kind of information. These implementations quite<br>
&gt; unique across the various platforms and considered to add<br>
&gt; additional complexity, OAM overhead and failure point to the<br>
&gt; overall solution.<br>
&gt;<br>
&gt; 2. Handling multiple media streams<br>
&gt; SRP should allow RS to select, which media stream should be<br>
&gt; recorded or not for a given session.<br>
&gt; As unified communication emerge, more and more customers<br>
&gt; using voice, video, instant messaging, etc. in their<br>
&gt; communication with their partners and clients. Customers may<br>
&gt; would like to decide, which media or which direction of the<br>
&gt; media should be recorded during the session. For compliance,<br>
&gt; probably only the voice part of a video conversation is<br>
&gt; sufficient too.<br>
&gt;<br>
&gt; 3. Security and encryption<br>
&gt; The draft very well describes the sensitivity of the topic<br>
&gt; from security point of view, however the term &quot;encryption&quot; i=
s<br>
&gt; missing from the requirements part. I see that &quot;eavesdropping<br>
&gt; protection&quot; is used, but what is the purpose of not using the<br>
&gt; most obvious solution for &quot;eavesdropping protection&quot;.<br>
</div></div>[JRE] Encryption does not necessarily give you protection again=
st eavesdropping, e.g., if keys are too weak or are exchanged in inappropri=
ate ways. I would say the requirement is for protection against eavesdroppi=
ng.<br>
</blockquote><div>Ok. I was thinking on similar mechanism like the one used=
 in SIP with TLS and SRTP.</div><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div><br>
&gt; I would also add: SRP should also allow the recording of<br>
&gt; encrypted calls.<br>
</div>[JRE] Can you please clarify whether you mean:<br>
- recording of media in the clear, said media having been transported to/fr=
om the communications partner in encrypted form; or<br>
- recording of encrypted media, so that it can be played back only with kno=
wledge of the keys used for the original session?<br>
<div><div></div><div><br></div></div></blockquote><div>The second one. Afte=
r reading Scott&#39;s comment, probably it cannot be easily incorporated in=
to the standard, but the requirements is a real one. We had several custome=
rs asking for this feature...</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div>
<br>
&gt;<br>
&gt; 4. Redundancy<br>
&gt; SRP should provide a mechanism to allow RCs to send the media<br>
&gt; to multiple RSs at one time.<br>
&gt; Probably endpoints with limited resources or networks with<br>
&gt; restricted bandwidth will not able to provide this feature,<br>
&gt; but in mission critical environments this could be a very<br>
&gt; nice way to provide redundancy. The failover mechanism<br>
&gt; described in the draft (and used by some of the vendors) is<br>
&gt; not protected from single point of failure.<br>
&gt;<br>
&gt; 5. Negotiating media handling capabilities<br>
&gt; SRP should allow RC and RS to negotiate audio/video codec<br>
&gt; capabilities (similar to SIP/SDP) and the RC should send<br>
&gt; media in the agreed codec.<br>
&gt; In most cases the codec will be exactly the same as the one<br>
&gt; used in the recorded session. However, high definition audio<br>
&gt; and video codecs are getting used in more and more sessions,<br>
&gt; for recording, the requirement in most cases allow to record<br>
&gt; the sessions in a lower resolution format. Either the RC,<br>
&gt; either the RS should provide transcoding capabilities. I know<br>
&gt; several video endpoints, which are able to send the video in<br>
&gt; 2 formats at one time (HD and CIF) and I can also imagine<br>
&gt; devices with on-board transcoding capabilities in the future<br>
&gt; (e.g. transcoding AAC-LD to G.729 for the RS) and some of the<br>
&gt; current implementations also allow the transcoding of the<br>
&gt; recording streams using network resources.<br>
</div></div>[JRE] Yes, but bear in mind that some devices might not have re=
sources to decode and re-encode. So although we could have a requirement th=
at allows the possibility of different codecs, I think there also has to be=
 a requirement to support devices that are unable to transcode.<br>


<div><br></div></blockquote><div>Agree.=A0</div><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex"><div>
&gt;<br>
&gt; 6. Others<br>
&gt; There are some points, which are mentioned in the use cases<br>
&gt; section, but no requirement is provided to support the given feature:<=
br>
&gt; - Silent monitoring and real-time applications: the entire<br>
&gt; mechanism should allow the users to implement a real-time<br>
&gt; monitoring functionality, which means that the media has to<br>
&gt; be transmitted with considerably low delay, in a real-time<br>
&gt; fashion to the RS.<br>
</div>[JRE] I think we need to be clear whether this is within scope, and i=
f so I would agree that some sort of requirement on timing would be appropr=
iate.<br>
<div><br>
&gt; - Handling complex calls: SRP should provide a mechanism to<br>
&gt; allow the RS to link together certain call segments in a<br>
&gt; complex call scenario (e.g. providing a global call<br>
&gt; identifier as a part of the meta data).<br>
</div>[JRE] This is a topic that has arisen in other contexts, and if a sol=
ution is to be provided I would imagine it would fall outside the scope of =
the SRP work.<br></blockquote><div>I think, it is a very important feature,=
 but I am aware of the=A0differences=A0among the various platforms/vendors,=
 so probably not all of them can meet this requirement. If this feature is =
not included in the standard, recorder vendors will be forced to implement =
or use existing TAPI/JTAPI/DMCC/etc. interfaces, which should be avoided in=
 my opinion...</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<font color=3D"#888888"><br>
John<br>
</font></blockquote></div><br>

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Date: Tue, 23 Feb 2010 17:54:02 +0100
Message-ID: <6a2adf621002230854h70e00126lc81e54c807d10a82@mail.gmail.com>
From: Gabor Moczar <gmoczar@gmail.com>
To: Scott Lawrence <scottlawrenc@avaya.com>
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THX Scott for the detailed clarification and I am now fully aware of
the appropriate terminology required at this stage.
I agree that the "recording of encrypted calls" topic is a very sensitive
one, but as I previously wrote to John, the requirement is from real
customers...

On Mon, Feb 22, 2010 at 3:06 PM, Scott Lawrence <scottlawrenc@avaya.com>wrote:

> On Fri, 2010-02-19 at 18:56 +0100, Gabor Moczar wrote:
> >
> > 3. Security and encryption
>
> > The draft very well describes the sensitivity of the topic from
> > security point of view, however the term "encryption" is missing from
> > the requirements part. I see that "eavesdropping protection" is used,
> > but what is the purpose of not using the most obvious solution for
> > "eavesdropping protection".
>
> Part of what you're encountering here is just the difference in
> preferred language.  This is a requirements document, not a solution
> document - it is laying out what goals a solution much achieve, and it
> is generally preferred that such a document _not_ specify any more than
> needed about how the goals are met.
>
> Encryption technologies can be used in many different ways to achieve
> different ends, so saying "encryption" in the context of this document
> would not actually specify which of those ends the encryption technology
> was to be employed for.  By writing the requirement as "confidentiality
> protection" that is made explicit, and there really isn't any other way
> to meet that requirement (in the Internet) than encryption, but writing
> it this way is actually more precise.
>
> >  I would also add: SRP should also allow the recording of encrypted
> > calls.
>
> There are a bunch of potential landmines in this area in that the IETF
> is very careful not to specify anything that allows an intermediate
> party to penetrate end-to-end confidentiality (there's that word again)
> without the knowledge and cooperation of the parties.  That's not to say
> that at least some interpretations of your requirement can't be met, but
> it does mean that the requirement must be carefully constructed to avoid
> creating problems.
>
>
>

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THX Scott for the detailed clarification and I am now fully aware of the=A0=
appropriate terminology required=A0at this stage.<div>I agree that the &quo=
t;recording of encrypted calls&quot; topic is a very sensitive one, but as =
I previously wrote to John, the requirement is from real customers...<br>
<br><div class=3D"gmail_quote">On Mon, Feb 22, 2010 at 3:06 PM, Scott Lawre=
nce <span dir=3D"ltr">&lt;<a href=3D"mailto:scottlawrenc@avaya.com">scottla=
wrenc@avaya.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" =
style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class=3D"im">On Fri, 2010-02-19 at 18:56 +0100, Gabor Moczar wrote:<br=
>
&gt;<br>
&gt; 3. Security and encryption<br>
<br>
&gt; The draft very well describes the sensitivity of the topic from<br>
&gt; security point of view, however the term &quot;encryption&quot; is mis=
sing from<br>
&gt; the requirements part. I see that &quot;eavesdropping protection&quot;=
 is used,<br>
&gt; but what is the purpose of not using the most obvious solution for<br>
&gt; &quot;eavesdropping protection&quot;.<br>
<br>
</div>Part of what you&#39;re encountering here is just the difference in<b=
r>
preferred language. =A0This is a requirements document, not a solution<br>
document - it is laying out what goals a solution much achieve, and it<br>
is generally preferred that such a document _not_ specify any more than<br>
needed about how the goals are met.<br>
<br>
Encryption technologies can be used in many different ways to achieve<br>
different ends, so saying &quot;encryption&quot; in the context of this doc=
ument<br>
would not actually specify which of those ends the encryption technology<br=
>
was to be employed for. =A0By writing the requirement as &quot;confidential=
ity<br>
protection&quot; that is made explicit, and there really isn&#39;t any othe=
r way<br>
to meet that requirement (in the Internet) than encryption, but writing<br>
it this way is actually more precise.<br>
<div class=3D"im"><br>
&gt; =A0I would also add: SRP should also allow the recording of encrypted<=
br>
&gt; calls.<br>
<br>
</div>There are a bunch of potential landmines in this area in that the IET=
F<br>
is very careful not to specify anything that allows an intermediate<br>
party to penetrate end-to-end confidentiality (there&#39;s that word again)=
<br>
without the knowledge and cooperation of the parties. =A0That&#39;s not to =
say<br>
that at least some interpretations of your requirement can&#39;t be met, bu=
t<br>
it does mean that the requirement must be carefully constructed to avoid<br=
>
creating problems.<br>
<br>
<br>
</blockquote></div><br></div>

--001485f33beca0385a04804765ce--

From john.elwell@siemens-enterprise.com  Tue Feb 23 09:07:39 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Gabor Moczar <gmoczar@gmail.com>
Date: Tue, 23 Feb 2010 18:09:39 +0100
Thread-Topic: [dispatch] Comments on Session Recording Protocol
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Gabor,

More on a couple of topics:

> -----Original Message-----
> From: Gabor Moczar [mailto:gmoczar@gmail.com]=20
> Sent: 23 February 2010 16:48
> To: Elwell, John
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] Comments on Session Recording Protocol
>=20
> 	>
> 	> 3. Security and encryption
> 	> The draft very well describes the sensitivity of the topic
> 	> from security point of view, however the term "encryption" is
> 	> missing from the requirements part. I see that "eavesdropping
> 	> protection" is used, but what is the purpose of not using the
> 	> most obvious solution for "eavesdropping protection".
> =09
> 	[JRE] Encryption does not necessarily give you=20
> protection against eavesdropping, e.g., if keys are too weak=20
> or are exchanged in inappropriate ways. I would say the=20
> requirement is for protection against eavesdropping.
> =09
>=20
> Ok. I was thinking on similar mechanism like the one used in=20
> SIP with TLS and SRTP.
[JRE] Sure, that might be a solution, but the requirement is prevention of =
eavesdropping.


>=20
>=20
> 	> I would also add: SRP should also allow the recording of
> 	> encrypted calls.
> =09
> 	[JRE] Can you please clarify whether you mean:
> 	- recording of media in the clear, said media having=20
> been transported to/from the communications partner in=20
> encrypted form; or
> 	- recording of encrypted media, so that it can be=20
> played back only with knowledge of the keys used for the=20
> original session?
> =09
>=20
>=20
> The second one. After reading Scott's comment, probably it=20
> cannot be easily incorporated into the standard, but the=20
> requirements is a real one. We had several customers asking=20
> for this feature...
[JRE] There are actually 3 points at which media can be encrypted:
- during transmission as part of the original communication session;
- during transmission as part of the media replication session from recordi=
ng client to recording server;
- during storage on the recording server (and subsequent retrieval for play=
back purposes, although this is getting somewhat beyond the scope of the pr=
oposed charter).

I think you are asking that it be encrypted once and remain encrypted for a=
ll three stages.

If we are talking real-time media transported with SRTP, there is a problem=
, because SRTP is tightly coupled to the RTP transport for the session conc=
erned. This makes it unsuitable for store and forward use, as discussed in =
draft-mattsson-srtp-store-and-forward. Similarly, it makes it unsuitable fo=
r use across two separate RTP sessions. It seems to me to be necessary to d=
ecrypt SRTP from the communication session at the recording client, and the=
n re-encrypt for transmission  to the recording server, and then decrypt an=
d re-encrypt again for storage, and so on. Possibly the 2nd and 3rd stages =
could be dealt with in one go using a technique such as in draft-naslund-sr=
tp-saf, but it is unlikely that the first stage could be combined, since th=
is would requiring cooperation of the remote party's UA.

John

From gmoczar@gmail.com  Tue Feb 23 09:25:15 2010
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Date: Tue, 23 Feb 2010 18:27:08 +0100
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From: Gabor Moczar <gmoczar@gmail.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [dispatch] Comments on Session Recording Protocol
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--001636c5b467f703b9048047dba2
Content-Type: text/plain; charset=ISO-8859-1

On Tue, Feb 23, 2010 at 6:09 PM, Elwell, John <
john.elwell@siemens-enterprise.com> wrote:

> Gabor,
>
> More on a couple of topics:
>
> > -----Original Message-----
> > From: Gabor Moczar [mailto:gmoczar@gmail.com]
> > Sent: 23 February 2010 16:48
> > To: Elwell, John
> > Cc: dispatch@ietf.org
> > Subject: Re: [dispatch] Comments on Session Recording Protocol
> >
> >       >
> >       > 3. Security and encryption
> >       > The draft very well describes the sensitivity of the topic
> >       > from security point of view, however the term "encryption" is
> >       > missing from the requirements part. I see that "eavesdropping
> >       > protection" is used, but what is the purpose of not using the
> >       > most obvious solution for "eavesdropping protection".
> >
> >       [JRE] Encryption does not necessarily give you
> > protection against eavesdropping, e.g., if keys are too weak
> > or are exchanged in inappropriate ways. I would say the
> > requirement is for protection against eavesdropping.
> >
> >
> > Ok. I was thinking on similar mechanism like the one used in
> > SIP with TLS and SRTP.
> [JRE] Sure, that might be a solution, but the requirement is prevention of
> eavesdropping.
>
>
> >
> >
> >       > I would also add: SRP should also allow the recording of
> >       > encrypted calls.
> >
> >       [JRE] Can you please clarify whether you mean:
> >       - recording of media in the clear, said media having
> > been transported to/from the communications partner in
> > encrypted form; or
> >       - recording of encrypted media, so that it can be
> > played back only with knowledge of the keys used for the
> > original session?
> >
> >
> >
> > The second one. After reading Scott's comment, probably it
> > cannot be easily incorporated into the standard, but the
> > requirements is a real one. We had several customers asking
> > for this feature...
> [JRE] There are actually 3 points at which media can be encrypted:
> - during transmission as part of the original communication session;
> - during transmission as part of the media replication session from
> recording client to recording server;
> - during storage on the recording server (and subsequent retrieval for
> playback purposes, although this is getting somewhat beyond the scope of the
> proposed charter).
>
> I think you are asking that it be encrypted once and remain encrypted for
> all three stages.
>
> If we are talking real-time media transported with SRTP, there is a
> problem, because SRTP is tightly coupled to the RTP transport for the
> session concerned. This makes it unsuitable for store and forward use, as
> discussed in draft-mattsson-srtp-store-and-forward. Similarly, it makes it
> unsuitable for use across two separate RTP sessions. It seems to me to be
> necessary to decrypt SRTP from the communication session at the recording
> client, and then re-encrypt for transmission  to the recording server, and
> then decrypt and re-encrypt again for storage, and so on. Possibly the 2nd
> and 3rd stages could be dealt with in one go using a technique such as in
> draft-naslund-srtp-saf, but it is unlikely that the first stage could be
> combined, since this would requiring cooperation of the remote party's UA.
>

You are right. The implementations, I am aware, do exactly the same thing
you described: re-encrypt for transmission to the RS with new keys.
Storing recordings in an encrypted format is a kind of must have feature on
the market, but I agree, it is out of scope of this work.

>
> John
>

--001636c5b467f703b9048047dba2
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Tue, Feb 23, 2010 at 6:09 PM, Elwell,=
 John <span dir=3D"ltr">&lt;<a href=3D"mailto:john.elwell@siemens-enterpris=
e.com">john.elwell@siemens-enterprise.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex;">
Gabor,<br>
<br>
More on a couple of topics:<br>
<div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: Gabor Moczar [mailto:<a href=3D"mailto:gmoczar@gmail.com">gmocza=
r@gmail.com</a>]<br>
&gt; Sent: 23 February 2010 16:48<br>
&gt; To: Elwell, John<br>
&gt; Cc: <a href=3D"mailto:dispatch@ietf.org">dispatch@ietf.org</a><br>
&gt; Subject: Re: [dispatch] Comments on Session Recording Protocol<br>
&gt;<br>
&gt; =A0 =A0 =A0 &gt;<br>
&gt; =A0 =A0 =A0 &gt; 3. Security and encryption<br>
&gt; =A0 =A0 =A0 &gt; The draft very well describes the sensitivity of the =
topic<br>
&gt; =A0 =A0 =A0 &gt; from security point of view, however the term &quot;e=
ncryption&quot; is<br>
&gt; =A0 =A0 =A0 &gt; missing from the requirements part. I see that &quot;=
eavesdropping<br>
&gt; =A0 =A0 =A0 &gt; protection&quot; is used, but what is the purpose of =
not using the<br>
&gt; =A0 =A0 =A0 &gt; most obvious solution for &quot;eavesdropping protect=
ion&quot;.<br>
&gt;<br>
&gt; =A0 =A0 =A0 [JRE] Encryption does not necessarily give you<br>
&gt; protection against eavesdropping, e.g., if keys are too weak<br>
&gt; or are exchanged in inappropriate ways. I would say the<br>
&gt; requirement is for protection against eavesdropping.<br>
&gt;<br>
&gt;<br>
&gt; Ok. I was thinking on similar mechanism like the one used in<br>
&gt; SIP with TLS and SRTP.<br>
</div>[JRE] Sure, that might be a solution, but the requirement is preventi=
on of eavesdropping.<br>
<div class=3D"im"><br>
<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 &gt; I would also add: SRP should also allow the recording=
 of<br>
&gt; =A0 =A0 =A0 &gt; encrypted calls.<br>
&gt;<br>
&gt; =A0 =A0 =A0 [JRE] Can you please clarify whether you mean:<br>
&gt; =A0 =A0 =A0 - recording of media in the clear, said media having<br>
&gt; been transported to/from the communications partner in<br>
&gt; encrypted form; or<br>
&gt; =A0 =A0 =A0 - recording of encrypted media, so that it can be<br>
&gt; played back only with knowledge of the keys used for the<br>
&gt; original session?<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The second one. After reading Scott&#39;s comment, probably it<br>
&gt; cannot be easily incorporated into the standard, but the<br>
&gt; requirements is a real one. We had several customers asking<br>
&gt; for this feature...<br>
</div>[JRE] There are actually 3 points at which media can be encrypted:<br=
>
- during transmission as part of the original communication session;<br>
- during transmission as part of the media replication session from recordi=
ng client to recording server;<br>
- during storage on the recording server (and subsequent retrieval for play=
back purposes, although this is getting somewhat beyond the scope of the pr=
oposed charter).<br>
<br>
I think you are asking that it be encrypted once and remain encrypted for a=
ll three stages.<br>
<br>
If we are talking real-time media transported with SRTP, there is a problem=
, because SRTP is tightly coupled to the RTP transport for the session conc=
erned. This makes it unsuitable for store and forward use, as discussed in =
draft-mattsson-srtp-store-and-forward. Similarly, it makes it unsuitable fo=
r use across two separate RTP sessions. It seems to me to be necessary to d=
ecrypt SRTP from the communication session at the recording client, and the=
n re-encrypt for transmission =A0to the recording server, and then decrypt =
and re-encrypt again for storage, and so on. Possibly the 2nd and 3rd stage=
s could be dealt with in one go using a technique such as in draft-naslund-=
srtp-saf, but it is unlikely that the first stage could be combined, since =
this would requiring cooperation of the remote party&#39;s UA.<br>
</blockquote><div><br></div><div>You are right. The implementations, I am a=
ware, do exactly the same thing you described: re-encrypt for transmission =
to the RS with new keys.</div><div>Storing recordings in an encrypted forma=
t is a kind of must have feature on the market, but I agree, it is out of s=
cope of this work.=A0</div>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<font color=3D"#888888"><br>
John<br>
</font></blockquote></div><br>

--001636c5b467f703b9048047dba2--

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Avasarala Ranjit-A20990 wrote:
> Hi Paul.
> 
> Thanks for the comments. <Ranjit-Feb23-2> 
> 
> 
> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Tuesday, February 23, 2010 7:40 PM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> Inline. I've trimmed out irrelevant stuff
> 
> Avasarala Ranjit-A20990 wrote:
>> Hi Paul
>>
>> My comments <Ranjit-Feb23>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> 
>> I think you are not understanding the kind of state model we are 
>> talking about.
>>
>> Think of the state as a collection of data that may change over time, 
>> that is visible to the notifier. When something in the state changes 
>> in that state, the notifier sends a NOTIFY to subscribers, saying "the
> 
>> state of the resource changed, here is the new value."
>>
>> <Ranjit-Feb23> In case of CDIV, the notifier triggers a NOTIFY when 
>> ever a diversion occurs and the filter criteria matches. So there is 
>> no resource as such whose value changes - except that notifier can 
>> maintain the count of number of diversions that have occurred and how 
>> many times the filter criteria matched and how many notifications were
> 
>> sent. So every time a notification is sent, these values get
> incremented.
> 
> I'm trying to find another way to conceptualize your situation - one
> that works better with the 3265 way of describing events.
> 
>> In the simplistic form, you could view that state as exactly the data 
>> needed to populate the body of your NOTIFY.
>> <Ranjit-Feb23> Then the state needs to contain the caller details, 
>> diversion time, reason for diversion and if filter criteria matched or
> 
>> not.
>>
>> It gets a more complicated if there can be multiple subscribers, and 
>> you don't allow them all to see all of the data. (I'm not certain if 
>> you have restrictions like that or not.) If you do, then there might 
>> be data in the state that come subscribers get and others do not.
>> <Ranjit-Feb23> No need of such a thing because a particular subscriber
> 
>> can only subscribe to his or her diversions and not others.
> 
> OK. Then we won't worry about that.
> 
>> In your case, it appears from your xml, that your state consists of 
>> zero or more diversions. (Specifically it can be more than one.) And 
>> for each there an originating URI, a diverting URI, a diverted-to URI,
> 
>> a diverting time, and a diverting reason.
>> <Ranjit-Feb23> could be. 
>>
>> But the diverting-URI is redundant - its equivalent to the resource 
>> that is the target of the subscription, and so will be constant 
>> throughout the subscription. I don't understand why it is present. My 
>> suspicion is that you have in mind that the server will serve more 
>> than one diverting user and so the *server* state consists of info for
> 
>> each diverting user it serves. But unless you allow a subscription to 
>> address the *server* rather than a particular diverting user, this is 
>> invisible to the state of a given resource subscription target.
>> <Ranjit-Feb23> Yes there is no need of diverting-URI. 
> 
> OK. Then it could be removed from the notification body, and the filter
> too, because it is redundant. Right?
> 
> Its being there confuses people like me, who assume that if its there it
> must have a purpose.
> 
>> If I have that right, then this state changes over time as calls are 
>> diverted. In particular, a new diversion is added. Then, 
>> simplistically, each notify would contain not only the most recent 
>> diversion, but all the prior diversions as well.
>> <Ranjit-Feb23> the set of previous diversions can be taken care of by 
>> use of History-Info header.
> 
> You are thinking of something different than I am.
> You are talking about previous diversions of the *same call*.
> I'm talking about diversions of previous calls by the same diverting
> user.
> 
> IOW,
> - at 2:00 a call from alice was diverted to bob.
> - at 3:00 a call from carol was diverted to ted.
> ...
> <Ranjit-Feb23-2> Ok here as you said, the notification always contains
> the most recent and latest diversion that occurred and the one that
> matched the subscriber's filter criteria. 

Except for what you say below

> Its certainly possible to *imagine* such a state, gradually increasing
> with time as other calls are diverted.
> 
>> Each notification will contain information about only one diversion 
>> which gets triggered when the diversion occurs.
> 
> So, if my phone has been *off*, when I turn it on I won't get a
> notification about a diversion that occurred while it was off? Is that a
> feature, or a limitation of the design?
> <Ranjit-Feb23-2> The notifications are temporarily stored in the server
> when the phone is *off* and they would be sent one by one when the phone
> is *on*. So yes it's a feature in the server to monitor phone's status
> and store the notifications if required - the amount of storage could
> depend on server policy and some may or may not. No hard rule that they
> should. If they do not store, then only the latest notification would be
> sent.

See, you really *do* have state!!!

IIUC, you are saying that the server might store history of multiple 
diversions, or just the most recent one, or none.

> If notifications of diversions are only delivered in realtime as they
> occur, then why do you bother to include the time of the diversion in
> the notification?
> <Ranjit-Feb23-2> true. This makes sense only when notifications are
> stored and sent later (phone off scenario)

When you say "phone off", I presume you mean "phone not subscribed to 
this event package". Is that right?

What do you do about multiple phones (extensions) on the same AOR? E.g. 
one at the office and another at home. Consider that case, and that I 
have been out of the office for some time, and arrive home without 
visiting the office. My home phone has been off, but the office phone 
has been on the whole time. Will I see the diversions that have occurred 
while I was out?

If you clear the state as soon as you send one notification with it, 
then when I get home I will see no diversions.

It would all be much cleaner if you kept the most recent N diversions in 
the server, where N>=1. When a new one comes in, if that makes N+1, then 
discard the oldest, and *then* send the notification to whoever is 
subscribed. If somebody else subscribes later, send what you have.

	Thanks,
	Paul

>> Or you could define it so that older diversions are removed, using any
> 
>> of a variety of algorithms. E.g. you might keep the most recent N of 
>> them, or all that have occurred in the last N minutes. (E.g. you might
> 
>> keep only the single most recent one.) But that does imply that 
>> somebody subscribing *after* a diversion has occurred will get a 
>> notification immediately with whatever that state is - say the most 
>> recent diversion, even if that occurred a long time ago. (This could 
>> be a feature, but the implementation needs to support it by retaining 
>> the state.)
>>
>> It starts to get messy if you don't want to retain the state. Then you
> 
>> must pretend that the new diversion was added to the (previously 
>> empty) state, caused notification(s), and then was immediately removed
> 
>> from the state. If viewed that way, the removal would also cause a
> notification.
>> I suggested how that might be handled, but Adam thought it improper.
>> <Ranjit-Feb23> ok.
> 
> The trick here is to find a way to model the state, and the implicit and
> explicit filters, to achieve the sort of results you want and permit the
> sort of implementation you want. There are no doubt several ways that
> may work.
> 
> I think what you need is something like:
> 
> - the state consists (abstractly) of x, y, z
> - the state is adjusted in the following way when
>    another call is diverted
> - the manner of matching a particular component in a
>    filter against the state is ...
> - the manner of dealing with multiple components in
>    a filter is ...
> - the manner of populating the notification body from
>    the state is ...
> 
> <Ranjit-Feb23-2> need to think further on this.
> 
> 	Thanks,
> 	Paul
> 

From zhouzp@huawei.com  Tue Feb 23 18:07:26 2010
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Date: Wed, 24 Feb 2010 10:09:18 +0800
From: zhipeng zhou <zhouzp@huawei.com>
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Hi Sal,
Thanks for your response.
While my question is what the difference for the far end device on whether
it hold single or muliple sessions with the other devices.
I mean, if the far end device hold several seperate sessions with the
household devices, for the user it is also available to get the integrated
experience by the support of the client.
For the communication itself, what profit can be got your proposal. The far
end device can maitain several sessions simultaneously.
Thanks.
Zhipeng

////////////////////////////////////////////////////////////////////////////
//////////////////////////////
Message: 4
Date: Tue, 23 Feb 2010 16:18:14 +0200
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
Subject: Re: [dispatch] A question	on
	"draft-loreto-dispatch-disaggregated-media-01"
To: dispatch@ietf.org
Message-ID: <4B83E3A6.6080701@ericsson.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"

Hi Zhipeng,

thanks for the question.

The advantage of the proposed solution is that the user does not need a 
device that is able to act as a controller:

indeed the proposed solution give the possibility for a user to create a 
multimedia session combining different media streams,
coming from different devices under his/her control (where none of those 
devices can act as a controller
for the other devices in the call), so that they are treated by the far 
end of the session as a single media session.

cheers
/Sal




On 02/21/2010 04:44 AM, zhipeng zhou wrote:
> Hi Salvatore and Gonzalo:
> After reading your newly uploaded 
> "draft-loreto-dispatch-disaggregated-media-01", esp. your section 4,
> I have a question as following :
> Since Bob's device will communicate with Alice's three devices, three 
> channels or sessions will be available to implement this scenario.
> While the sentece in this document "Since this type of scenario is not 
> covered by existing mechanisms, we
>    propose to initiate work on SIP extensions to support it." shows 
> that you like to implement this scenario through a new way.
> So, may you pls give a rough diagram and  indicate the advantage of 
> your proposed new solution.
> Thanks.
> Zhipeng
>
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Hi Salvatore and Gonzalo,

I also think there are requirements to express a particular relationship 
between two or more of streams(m= lines) in different sessions/dialogs, 
while people do not want to combine there m= lines in one SDP or doing so 
is too hard for them. 
And further, I think we should allow;
1. combine only part of session1's streams and part of session2's streams 
as one group;
2. one session's differet streams can be in differet groups.

And as we have the mechanism to express a particular relationship between 
two or more of streams(m= lines) in one session(RFC3388), we may do 
extension on this base. 

If we can do the extension along this way, I have some suggestion:
1. To group media streams in different session(SDP), then we need the 
placeholder of session1's stream in session2;
2. While grouping streams in different sessions, session ID may be needed 
as part of the index. 

Thanks,

Gao

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<br><font size=2 face="sans-serif">Hi Salvatore and Gonzalo,</font>
<br>
<br><font size=2 face="sans-serif">I also think there are requirements
to express a particular relationship between two or more of streams(m=
lines) in different sessions/dialogs, while people do not want to combine
there m= lines in one SDP or doing so is too hard for them. </font>
<br><font size=2 face="sans-serif">And further, I think we should allow;</font>
<br><font size=2 face="sans-serif">1. combine only part of session1's streams
and part of session2's streams as one group;</font>
<br><font size=2 face="sans-serif">2. one session's differet streams can
be in differet groups.</font>
<br>
<br><font size=2 face="sans-serif">And as we have the mechanism to express
a particular relationship between two or more of streams(m= lines) in one
session(RFC3388), we may do extension on this base. </font>
<br>
<br><font size=2 face="sans-serif">If we can do the extension along this
way, I have some suggestion:</font>
<br><font size=2 face="sans-serif">1. To group media streams in different
session(SDP), then we need the placeholder of session1's stream in session2;</font>
<br><font size=2 face="sans-serif">2. While grouping streams in different
sessions, session ID may be needed as part of the index. </font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Avasarala Ranjit-A20990 <ranjit@motorola.com>
Date: Wed, 24 Feb 2010 09:40:13 +0100
Thread-Topic: [dispatch] [Fwd: Re: Preliminary version	ofExpert	review ofdraft-avasarala-dispatch-comm-div-notification-03]
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version	ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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I appreciate Paul's efforts to try to sort this out, but it seems to me tha=
t we are having to go to great lengths to make this fit the RFC 3265 model.=
 Perhaps HTTP would be a better way of getting call log information, includ=
ing not only diverted calls but also, say, missed calls that were not diver=
ted (e.g., when no UA registered and no diversion conditions apply). Parame=
ters in the GET request could filter for particular types (e.g., diverted c=
alls only) and indicate how far to go back in history. For the UA to become=
 aware that a state change has occurred, the event package in draft-roach-h=
ttp-subscribe could be used.

John

> -----Original Message-----
> From: dispatch-bounces@ietf.org=20
> [mailto:dispatch-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 23 February 2010 18:48
> To: Avasarala Ranjit-A20990
> Cc: dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version=20
> ofExpert review ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
>=20
>=20
> Avasarala Ranjit-A20990 wrote:
> > Hi Paul.
> >=20
> > Thanks for the comments. <Ranjit-Feb23-2>=20
> >=20
> >=20
> > Regards
> > Ranjit
> >=20
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> > Sent: Tuesday, February 23, 2010 7:40 PM
> > To: Avasarala Ranjit-A20990
> > Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> > Subject: Re: [dispatch] [Fwd: Re: Preliminary version=20
> ofExpert review
> > ofdraft-avasarala-dispatch-comm-div-notification-03]
> >=20
> > Inline. I've trimmed out irrelevant stuff
> >=20
> > Avasarala Ranjit-A20990 wrote:
> >> Hi Paul
> >>
> >> My comments <Ranjit-Feb23>
> >>
> >>
> >> Regards
> >> Ranjit
> >>
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >=20
> >> I think you are not understanding the kind of state model we are=20
> >> talking about.
> >>
> >> Think of the state as a collection of data that may change=20
> over time,=20
> >> that is visible to the notifier. When something in the=20
> state changes=20
> >> in that state, the notifier sends a NOTIFY to subscribers,=20
> saying "the
> >=20
> >> state of the resource changed, here is the new value."
> >>
> >> <Ranjit-Feb23> In case of CDIV, the notifier triggers a=20
> NOTIFY when=20
> >> ever a diversion occurs and the filter criteria matches.=20
> So there is=20
> >> no resource as such whose value changes - except that notifier can=20
> >> maintain the count of number of diversions that have=20
> occurred and how=20
> >> many times the filter criteria matched and how many=20
> notifications were
> >=20
> >> sent. So every time a notification is sent, these values get
> > incremented.
> >=20
> > I'm trying to find another way to conceptualize your situation - one
> > that works better with the 3265 way of describing events.
> >=20
> >> In the simplistic form, you could view that state as=20
> exactly the data=20
> >> needed to populate the body of your NOTIFY.
> >> <Ranjit-Feb23> Then the state needs to contain the caller details,=20
> >> diversion time, reason for diversion and if filter=20
> criteria matched or
> >=20
> >> not.
> >>
> >> It gets a more complicated if there can be multiple=20
> subscribers, and=20
> >> you don't allow them all to see all of the data. (I'm not=20
> certain if=20
> >> you have restrictions like that or not.) If you do, then=20
> there might=20
> >> be data in the state that come subscribers get and others do not.
> >> <Ranjit-Feb23> No need of such a thing because a=20
> particular subscriber
> >=20
> >> can only subscribe to his or her diversions and not others.
> >=20
> > OK. Then we won't worry about that.
> >=20
> >> In your case, it appears from your xml, that your state=20
> consists of=20
> >> zero or more diversions. (Specifically it can be more than=20
> one.) And=20
> >> for each there an originating URI, a diverting URI, a=20
> diverted-to URI,
> >=20
> >> a diverting time, and a diverting reason.
> >> <Ranjit-Feb23> could be.=20
> >>
> >> But the diverting-URI is redundant - its equivalent to the=20
> resource=20
> >> that is the target of the subscription, and so will be constant=20
> >> throughout the subscription. I don't understand why it is=20
> present. My=20
> >> suspicion is that you have in mind that the server will serve more=20
> >> than one diverting user and so the *server* state consists=20
> of info for
> >=20
> >> each diverting user it serves. But unless you allow a=20
> subscription to=20
> >> address the *server* rather than a particular diverting=20
> user, this is=20
> >> invisible to the state of a given resource subscription target.
> >> <Ranjit-Feb23> Yes there is no need of diverting-URI.=20
> >=20
> > OK. Then it could be removed from the notification body,=20
> and the filter
> > too, because it is redundant. Right?
> >=20
> > Its being there confuses people like me, who assume that if=20
> its there it
> > must have a purpose.
> >=20
> >> If I have that right, then this state changes over time as=20
> calls are=20
> >> diverted. In particular, a new diversion is added. Then,=20
> >> simplistically, each notify would contain not only the most recent=20
> >> diversion, but all the prior diversions as well.
> >> <Ranjit-Feb23> the set of previous diversions can be taken=20
> care of by=20
> >> use of History-Info header.
> >=20
> > You are thinking of something different than I am.
> > You are talking about previous diversions of the *same call*.
> > I'm talking about diversions of previous calls by the same diverting
> > user.
> >=20
> > IOW,
> > - at 2:00 a call from alice was diverted to bob.
> > - at 3:00 a call from carol was diverted to ted.
> > ...
> > <Ranjit-Feb23-2> Ok here as you said, the notification=20
> always contains
> > the most recent and latest diversion that occurred and the one that
> > matched the subscriber's filter criteria.=20
>=20
> Except for what you say below
>=20
> > Its certainly possible to *imagine* such a state, gradually=20
> increasing
> > with time as other calls are diverted.
> >=20
> >> Each notification will contain information about only one=20
> diversion=20
> >> which gets triggered when the diversion occurs.
> >=20
> > So, if my phone has been *off*, when I turn it on I won't get a
> > notification about a diversion that occurred while it was=20
> off? Is that a
> > feature, or a limitation of the design?
> > <Ranjit-Feb23-2> The notifications are temporarily stored=20
> in the server
> > when the phone is *off* and they would be sent one by one=20
> when the phone
> > is *on*. So yes it's a feature in the server to monitor=20
> phone's status
> > and store the notifications if required - the amount of=20
> storage could
> > depend on server policy and some may or may not. No hard=20
> rule that they
> > should. If they do not store, then only the latest=20
> notification would be
> > sent.
>=20
> See, you really *do* have state!!!
>=20
> IIUC, you are saying that the server might store history of multiple=20
> diversions, or just the most recent one, or none.
>=20
> > If notifications of diversions are only delivered in=20
> realtime as they
> > occur, then why do you bother to include the time of the=20
> diversion in
> > the notification?
> > <Ranjit-Feb23-2> true. This makes sense only when notifications are
> > stored and sent later (phone off scenario)
>=20
> When you say "phone off", I presume you mean "phone not subscribed to=20
> this event package". Is that right?
>=20
> What do you do about multiple phones (extensions) on the same=20
> AOR? E.g.=20
> one at the office and another at home. Consider that case, and that I=20
> have been out of the office for some time, and arrive home without=20
> visiting the office. My home phone has been off, but the office phone=20
> has been on the whole time. Will I see the diversions that=20
> have occurred=20
> while I was out?
>=20
> If you clear the state as soon as you send one notification with it,=20
> then when I get home I will see no diversions.
>=20
> It would all be much cleaner if you kept the most recent N=20
> diversions in=20
> the server, where N>=3D1. When a new one comes in, if that=20
> makes N+1, then=20
> discard the oldest, and *then* send the notification to whoever is=20
> subscribed. If somebody else subscribes later, send what you have.
>=20
> 	Thanks,
> 	Paul
>=20
> >> Or you could define it so that older diversions are=20
> removed, using any
> >=20
> >> of a variety of algorithms. E.g. you might keep the most=20
> recent N of=20
> >> them, or all that have occurred in the last N minutes.=20
> (E.g. you might
> >=20
> >> keep only the single most recent one.) But that does imply that=20
> >> somebody subscribing *after* a diversion has occurred will get a=20
> >> notification immediately with whatever that state is - say=20
> the most=20
> >> recent diversion, even if that occurred a long time ago.=20
> (This could=20
> >> be a feature, but the implementation needs to support it=20
> by retaining=20
> >> the state.)
> >>
> >> It starts to get messy if you don't want to retain the=20
> state. Then you
> >=20
> >> must pretend that the new diversion was added to the (previously=20
> >> empty) state, caused notification(s), and then was=20
> immediately removed
> >=20
> >> from the state. If viewed that way, the removal would also cause a
> > notification.
> >> I suggested how that might be handled, but Adam thought it=20
> improper.
> >> <Ranjit-Feb23> ok.
> >=20
> > The trick here is to find a way to model the state, and the=20
> implicit and
> > explicit filters, to achieve the sort of results you want=20
> and permit the
> > sort of implementation you want. There are no doubt several=20
> ways that
> > may work.
> >=20
> > I think what you need is something like:
> >=20
> > - the state consists (abstractly) of x, y, z
> > - the state is adjusted in the following way when
> >    another call is diverted
> > - the manner of matching a particular component in a
> >    filter against the state is ...
> > - the manner of dealing with multiple components in
> >    a filter is ...
> > - the manner of populating the notification body from
> >    the state is ...
> >=20
> > <Ranjit-Feb23-2> need to think further on this.
> >=20
> > 	Thanks,
> > 	Paul
> >=20
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> =

From salvatore.loreto@ericsson.com  Wed Feb 24 05:27:49 2010
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Date: Wed, 24 Feb 2010 15:28:57 +0200
From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [dispatch] Suggestion on draft-loreto-dispatch-disaggregated-media-01.txt
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Hi Gao,

can you provide us some simple use cases for your two proposed scenarios?

thanks
Sal

On 02/24/2010 08:14 AM, gao.yang2@zte.com.cn wrote:
>
> Hi Salvatore and Gonzalo,
>
> I also think there are requirements to express a particular 
> relationship between two or more of streams(m= lines) in different 
> sessions/dialogs, while people do not want to combine there m= lines 
> in one SDP or doing so is too hard for them.
> And further, I think we should allow;
> 1. combine only part of session1's streams and part of session2's 
> streams as one group;
> 2. one session's differet streams can be in differet groups.
> And as we have the mechanism to express a particular relationship 
> between two or more of streams(m= lines) in one session(RFC3388), we 
> may do extension on this base.
>
> If we can do the extension along this way, I have some suggestion:
> 1. To group media streams in different session(SDP), then we need the 
> placeholder of session1's stream in session2;
> 2. While grouping streams in different sessions, session ID may be 
> needed as part of the index.
>
> Thanks,
>
> Gao
>
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
> ===================================
> --------------------------------------------------------
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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 http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#ffffff">
Hi Gao,<br>
<br>
can you provide us some simple use cases for your two proposed
scenarios?<br>
<br>
thanks<br>
Sal<br>
<br>
On 02/24/2010 08:14 AM, <a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a> wrote:
<blockquote
 cite="mid:OFDD5F94DA.2FD39BD1-ON482576D4.001F0731-482576D4.00226889@zte.com.cn"
 type="cite">
  <meta http-equiv="Context-Type" content="text/html; charset=us-ascii">
  <br>
Hi Salvatore and Gonzalo, <br>
  <br>
I also think there are requirements
to express a particular relationship between two or more of streams(m=
lines) in different sessions/dialogs, while people do not want to
combine
there m= lines in one SDP or doing so is too hard for them. <br>
And further, I think we should allow; <br>
1. combine only part of session1's streams
and part of session2's streams as one group; <br>
2. one session's differet streams can
be in differet groups.<br>
</blockquote>
<blockquote
 cite="mid:OFDD5F94DA.2FD39BD1-ON482576D4.001F0731-482576D4.00226889@zte.com.cn"
 type="cite"> And as we have the mechanism to express
a particular relationship between two or more of streams(m= lines) in
one
session(RFC3388), we may do extension on this base. <br>
  <br>
If we can do the extension along this
way, I have some suggestion: <br>
1. To group media streams in different
session(SDP), then we need the placeholder of session1's stream in
session2; <br>
2. While grouping streams in different
sessions, session ID may be needed as part of the index. <br>
  <br>
Thanks, <br>
  <br>
Gao <br>
  <br>
===================================<br>
Zip &nbsp; &nbsp;: 210012<br>
Tel &nbsp; &nbsp;: 87211<br>
Tel2 &nbsp; :(+86)-025-52877211<br>
e_mail : <a class="moz-txt-link-abbreviated" href="mailto:gao.yang2@zte.com.cn">gao.yang2@zte.com.cn</a><br>
=================================== <br>
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--=_alternative 000B4D9F482576D5_=--


From ranjit@motorola.com  Wed Feb 24 20:55:04 2010
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thread-topic: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi Paul

My comments <Ranjit-Feb25>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Wednesday, February 24, 2010 12:18 AM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]



Avasarala Ranjit-A20990 wrote:
> Hi Paul.
>=20
> Thanks for the comments. <Ranjit-Feb23-2>
>=20
>=20
> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, February 23, 2010 7:40 PM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
> Inline. I've trimmed out irrelevant stuff
>=20
> Avasarala Ranjit-A20990 wrote:
>> Hi Paul
>>
>> My comments <Ranjit-Feb23>
>>
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>=20
>> I think you are not understanding the kind of state model we are=20
>> talking about.
>>
>> Think of the state as a collection of data that may change over time,

>> that is visible to the notifier. When something in the state changes=20
>> in that state, the notifier sends a NOTIFY to subscribers, saying=20
>> "the
>=20
>> state of the resource changed, here is the new value."
>>
>> <Ranjit-Feb23> In case of CDIV, the notifier triggers a NOTIFY when=20
>> ever a diversion occurs and the filter criteria matches. So there is=20
>> no resource as such whose value changes - except that notifier can=20
>> maintain the count of number of diversions that have occurred and how

>> many times the filter criteria matched and how many notifications=20
>> were
>=20
>> sent. So every time a notification is sent, these values get
> incremented.
>=20
> I'm trying to find another way to conceptualize your situation - one=20
> that works better with the 3265 way of describing events.
>=20
>> In the simplistic form, you could view that state as exactly the data

>> needed to populate the body of your NOTIFY.
>> <Ranjit-Feb23> Then the state needs to contain the caller details,=20
>> diversion time, reason for diversion and if filter criteria matched=20
>> or
>=20
>> not.
>>
>> It gets a more complicated if there can be multiple subscribers, and=20
>> you don't allow them all to see all of the data. (I'm not certain if=20
>> you have restrictions like that or not.) If you do, then there might=20
>> be data in the state that come subscribers get and others do not.
>> <Ranjit-Feb23> No need of such a thing because a particular=20
>> subscriber
>=20
>> can only subscribe to his or her diversions and not others.
>=20
> OK. Then we won't worry about that.
<Ranjit-Feb25> Yes
>=20
>> In your case, it appears from your xml, that your state consists of=20
>> zero or more diversions. (Specifically it can be more than one.) And=20
>> for each there an originating URI, a diverting URI, a diverted-to=20
>> URI,
>=20
>> a diverting time, and a diverting reason.
>> <Ranjit-Feb23> could be.=20
>>
>> But the diverting-URI is redundant - its equivalent to the resource=20
>> that is the target of the subscription, and so will be constant=20
>> throughout the subscription. I don't understand why it is present. My

>> suspicion is that you have in mind that the server will serve more=20
>> than one diverting user and so the *server* state consists of info=20
>> for
>=20
>> each diverting user it serves. But unless you allow a subscription to

>> address the *server* rather than a particular diverting user, this is

>> invisible to the state of a given resource subscription target.
>> <Ranjit-Feb23> Yes there is no need of diverting-URI.=20
>=20
> OK. Then it could be removed from the notification body, and the=20
> filter too, because it is redundant. Right?
<Ranjit-Feb25> yes we do not need diverting-URI in the notification
body.
>=20
> Its being there confuses people like me, who assume that if its there=20
> it must have a purpose.
>=20
>> If I have that right, then this state changes over time as calls are=20
>> diverted. In particular, a new diversion is added. Then,=20
>> simplistically, each notify would contain not only the most recent=20
>> diversion, but all the prior diversions as well.
>> <Ranjit-Feb23> the set of previous diversions can be taken care of by

>> use of History-Info header.
>=20
> You are thinking of something different than I am.
> You are talking about previous diversions of the *same call*.
> I'm talking about diversions of previous calls by the same diverting=20
> user.
>=20
> IOW,
> - at 2:00 a call from alice was diverted to bob.
> - at 3:00 a call from carol was diverted to ted.
> ...
> <Ranjit-Feb23-2> Ok here as you said, the notification always contains

> the most recent and latest diversion that occurred and the one that=20
> matched the subscriber's filter criteria.

Except for what you say below

> Its certainly possible to *imagine* such a state, gradually increasing

> with time as other calls are diverted.
>=20
>> Each notification will contain information about only one diversion=20
>> which gets triggered when the diversion occurs.
>=20
> So, if my phone has been *off*, when I turn it on I won't get a=20
> notification about a diversion that occurred while it was off? Is that

> a feature, or a limitation of the design?
> <Ranjit-Feb23-2> The notifications are temporarily stored in the=20
> server when the phone is *off* and they would be sent one by one when=20
> the phone is *on*. So yes it's a feature in the server to monitor=20
> phone's status and store the notifications if required - the amount of

> storage could depend on server policy and some may or may not. No hard

> rule that they should. If they do not store, then only the latest=20
> notification would be sent.

See, you really *do* have state!!!


IIUC, you are saying that the server might store history of multiple
diversions, or just the most recent one, or none.

> If notifications of diversions are only delivered in realtime as they=20
> occur, then why do you bother to include the time of the diversion in=20
> the notification?
> <Ranjit-Feb23-2> true. This makes sense only when notifications are=20
> stored and sent later (phone off scenario)

When you say "phone off", I presume you mean "phone not subscribed to
this event package". Is that right?
<Ranjit-Feb25> True. This is ths phone state - being ON or OFF i.e
registered with the network or not and then if it
Is subscribed to CDIVN service or not.=20

What do you do about multiple phones (extensions) on the same AOR? E.g.=20
one at the office and another at home. Consider that case, and that I
have been out of the office for some time, and arrive home without
visiting the office. My home phone has been off, but the office phone
has been on the whole time. Will I see the diversions that have occurred
while I was out?

<Ranjit-Feb25> Here it depends on the target of the call. i.e if the
call was targetted for your home phone and it was off, and there was a
rule that
Diverts all incoming calls to home be diverted, then you would get a
notification.=20

If you clear the state as soon as you send one notification with it,
then when I get home I will see no diversions.
<Ranjit-Feb25> True. Once a notification about a particular diversion is
sent, then the server's state is back to idle as if no diversions have
occurred and nothing to do.=20

It would all be much cleaner if you kept the most recent N diversions in
the server, where N>=3D1. When a new one comes in, if that makes N+1, =
then
discard the oldest, and *then* send the notification to whoever is
subscribed. If somebody else subscribes later, send what you have.
<Ranjit-Feb25> This actually depends on how many diversions u want to
store, server storage per subscriber - subscription types like silver.
Gold where gold subscriber gets higher storage than silver subscriber

	Thanks,
	Paul

>> Or you could define it so that older diversions are removed, using=20
>> any
>=20
>> of a variety of algorithms. E.g. you might keep the most recent N of=20
>> them, or all that have occurred in the last N minutes. (E.g. you=20
>> might
>=20
>> keep only the single most recent one.) But that does imply that=20
>> somebody subscribing *after* a diversion has occurred will get a=20
>> notification immediately with whatever that state is - say the most=20
>> recent diversion, even if that occurred a long time ago. (This could=20
>> be a feature, but the implementation needs to support it by retaining

>> the state.)
>>
>> It starts to get messy if you don't want to retain the state. Then=20
>> you
>=20
>> must pretend that the new diversion was added to the (previously
>> empty) state, caused notification(s), and then was immediately=20
>> removed
>=20
>> from the state. If viewed that way, the removal would also cause a
> notification.
>> I suggested how that might be handled, but Adam thought it improper.
>> <Ranjit-Feb23> ok.
>=20
> The trick here is to find a way to model the state, and the implicit=20
> and explicit filters, to achieve the sort of results you want and=20
> permit the sort of implementation you want. There are no doubt several

> ways that may work.
>=20
> I think what you need is something like:
>=20
> - the state consists (abstractly) of x, y, z
> - the state is adjusted in the following way when
>    another call is diverted
> - the manner of matching a particular component in a
>    filter against the state is ...
> - the manner of dealing with multiple components in
>    a filter is ...
> - the manner of populating the notification body from
>    the state is ...
>=20
> <Ranjit-Feb23-2> need to think further on this.
>=20
> 	Thanks,
> 	Paul
>=20

From pkyzivat@cisco.com  Thu Feb 25 13:05:46 2010
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Avasarala Ranjit-A20990 wrote:

>>> But the diverting-URI is redundant - its equivalent to the resource 
>>> that is the target of the subscription, and so will be constant 
>>> throughout the subscription. I don't understand why it is present. My
> 
>>> suspicion is that you have in mind that the server will serve more 
>>> than one diverting user and so the *server* state consists of info 
>>> for
>>> each diverting user it serves. But unless you allow a subscription to
>>> address the *server* rather than a particular diverting user, this is
> 
>>> invisible to the state of a given resource subscription target.
>>> <Ranjit-Feb23> Yes there is no need of diverting-URI. 
>> OK. Then it could be removed from the notification body, and the 
>> filter too, because it is redundant. Right?
> <Ranjit-Feb25> yes we do not need diverting-URI in the notification
> body.
>> Its being there confuses people like me, who assume that if its there 
>> it must have a purpose.

If you want to leave it there, even though its redundant, the main issue 
is making it *clear* that it is redundant and will always match the 
subscribed-to URI.

The big problem is the potential that it might differ, and what it would 
mean if it did.

> IIUC, you are saying that the server might store history of multiple
> diversions, or just the most recent one, or none.
> 
>> If notifications of diversions are only delivered in realtime as they 
>> occur, then why do you bother to include the time of the diversion in 
>> the notification?
>> <Ranjit-Feb23-2> true. This makes sense only when notifications are 
>> stored and sent later (phone off scenario)
> 
> When you say "phone off", I presume you mean "phone not subscribed to
> this event package". Is that right?
> <Ranjit-Feb25> True. This is ths phone state - being ON or OFF i.e
> registered with the network or not and then if it
> Is subscribed to CDIVN service or not. 

Well, in IMS I guess if you aren't registered you can't be subscribed.
But presumably you can be registered and not subscribed. Its the 
subscription that is the gating factor here, right?

> What do you do about multiple phones (extensions) on the same AOR? E.g. 
> one at the office and another at home. Consider that case, and that I
> have been out of the office for some time, and arrive home without
> visiting the office. My home phone has been off, but the office phone
> has been on the whole time. Will I see the diversions that have occurred
> while I was out?
> 
> <Ranjit-Feb25> Here it depends on the target of the call. i.e if the
> call was targetted for your home phone and it was off, and there was a
> rule that
> Diverts all incoming calls to home be diverted, then you would get a
> notification. 

I don't quite understand, and think it may be significant.
I was talking about two phones registered to the same URI. E.g. AOR 
sip:alice@atlanta.org with two contacts: 
sip:line1@alice.office.atlanta.com and sip:line2@alice.home.atlanta.com.

Hence both phones would be subscribing to the CDIV service at 
sip:alice@atlanta.org.

In that arrangement a caller can't (easily, without callerprefs or 
gruus) target a call to the home or the office. Rather the calls just 
ring both places. And then diversion rules will apply to calls to that 
AOR in general, so that both phones need to know about the diversion so 
that alice will know about it regardless of where she is.

But I think you are talking about some other arrangement, with multiple 
AORs that have some relationship to one another here, that isn't clear 
to me from the text. If that is so, can you explain?

> If you clear the state as soon as you send one notification with it,
> then when I get home I will see no diversions.
> <Ranjit-Feb25> True. Once a notification about a particular diversion is
> sent, then the server's state is back to idle as if no diversions have
> occurred and nothing to do. 
> 
> It would all be much cleaner if you kept the most recent N diversions in
> the server, where N>=1. When a new one comes in, if that makes N+1, then
> discard the oldest, and *then* send the notification to whoever is
> subscribed. If somebody else subscribes later, send what you have.
> <Ranjit-Feb25> This actually depends on how many diversions u want to
> store, server storage per subscriber - subscription types like silver.
> Gold where gold subscriber gets higher storage than silver subscriber

In the above I was assuming that the value of N is configurable. A given 
AOR has some value. (Or maybe the whole CDIV service has a common 
value.) And maybe its enough for that value to always be one.

The problem case is where the value of N is zero.

	Thanks,
	Paul

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From: "Avasarala Ranjit-A20990" <ranjit@motorola.com>
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Hi Paul

My comments <Ranjit-Feb26>

[1] diverting-URI: I plan to remove this field from the list of elements
that can be part of the notification body and filter criteria.=20

[2] phone-off and not subscribed-to: Here I mean, phone is off implies
the subscriber is not registered to the IMS network at that moment. But
the
   subcription could still be valid. So when the subscriber's phone is
switched off, the notifier will not be able to send the notification to
the
   subscriber and could store the notifications until the subscriber
re-registers with the IMS network. The number of notifications that the
   notifier would store would be policy dependent.=20

   But if the subscriber is registered and not subscribed, then there is
no way the notifier can send any notifications. A case of no
subscription.

[3] in the example u suggested where sip:alice@atlanta.org with two
contacts: sip:line1@alice.office.atlanta.com and
sip:line2@alice.home.atlanta.com and
    both the phones ring, then it would be a case of parallel forking.
Now say one or both the phones have set diversion rules, to divert the
call to say=20
    sip:voicemail@alice.atlanta.com, then the notifications would get
generated with respect to both the contacts.

[4] Yes the value of N is configurable and depends on how many
diversions the subscribe wants to keep or could be dependent on server
policy (some maximum
    number). If the server policy is to send notifications as and when
they occur and not keep any divsersions information, then the value of
last N=20
    diversions will be zero. But if the server chooses to retain the
last diversion always, then the value of N will be 1 and so on.

Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Friday, February 26, 2010 2:38 AM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]



Avasarala Ranjit-A20990 wrote:

>>> But the diverting-URI is redundant - its equivalent to the resource=20
>>> that is the target of the subscription, and so will be constant=20
>>> throughout the subscription. I don't understand why it is present.=20
>>> My
>=20
>>> suspicion is that you have in mind that the server will serve more=20
>>> than one diverting user and so the *server* state consists of info=20
>>> for each diverting user it serves. But unless you allow a=20
>>> subscription to address the *server* rather than a particular=20
>>> diverting user, this is
>=20
>>> invisible to the state of a given resource subscription target.
>>> <Ranjit-Feb23> Yes there is no need of diverting-URI.=20
>> OK. Then it could be removed from the notification body, and the=20
>> filter too, because it is redundant. Right?
> <Ranjit-Feb25> yes we do not need diverting-URI in the notification=20
> body.
>> Its being there confuses people like me, who assume that if its there

>> it must have a purpose.

If you want to leave it there, even though its redundant, the main issue
is making it *clear* that it is redundant and will always match the
subscribed-to URI.

The big problem is the potential that it might differ, and what it would
mean if it did.
<Ranjit-Feb26> [1]

> IIUC, you are saying that the server might store history of multiple=20
> diversions, or just the most recent one, or none.
>=20
>> If notifications of diversions are only delivered in realtime as they

>> occur, then why do you bother to include the time of the diversion in

>> the notification?
>> <Ranjit-Feb23-2> true. This makes sense only when notifications are=20
>> stored and sent later (phone off scenario)
>=20
> When you say "phone off", I presume you mean "phone not subscribed to=20
> this event package". Is that right?
> <Ranjit-Feb25> True. This is ths phone state - being ON or OFF i.e=20
> registered with the network or not and then if it Is subscribed to=20
> CDIVN service or not.

Well, in IMS I guess if you aren't registered you can't be subscribed.
But presumably you can be registered and not subscribed. Its the
subscription that is the gating factor here, right?
<Ranjit-Feb26> [2]

> What do you do about multiple phones (extensions) on the same AOR?
E.g.=20
> one at the office and another at home. Consider that case, and that I=20
> have been out of the office for some time, and arrive home without=20
> visiting the office. My home phone has been off, but the office phone=20
> has been on the whole time. Will I see the diversions that have=20
> occurred while I was out?
>=20
> <Ranjit-Feb25> Here it depends on the target of the call. i.e if the=20
> call was targetted for your home phone and it was off, and there was a

> rule that Diverts all incoming calls to home be diverted, then you=20
> would get a notification.

I don't quite understand, and think it may be significant.
I was talking about two phones registered to the same URI. E.g. AOR
sip:alice@atlanta.org with two contacts:=20
sip:line1@alice.office.atlanta.com and sip:line2@alice.home.atlanta.com.

Hence both phones would be subscribing to the CDIV service at
sip:alice@atlanta.org.

In that arrangement a caller can't (easily, without callerprefs or
gruus) target a call to the home or the office. Rather the calls just
ring both places. And then diversion rules will apply to calls to that
AOR in general, so that both phones need to know about the diversion so
that alice will know about it regardless of where she is.

But I think you are talking about some other arrangement, with multiple
AORs that have some relationship to one another here, that isn't clear
to me from the text. If that is so, can you explain?

<Ranjit-Feb26> [3]

> If you clear the state as soon as you send one notification with it,=20
> then when I get home I will see no diversions.
> <Ranjit-Feb25> True. Once a notification about a particular diversion=20
> is sent, then the server's state is back to idle as if no diversions=20
> have occurred and nothing to do.
>=20
> It would all be much cleaner if you kept the most recent N diversions=20
> in the server, where N>=3D1. When a new one comes in, if that makes =
N+1,

> then discard the oldest, and *then* send the notification to whoever=20
> is subscribed. If somebody else subscribes later, send what you have.
> <Ranjit-Feb25> This actually depends on how many diversions u want to=20
> store, server storage per subscriber - subscription types like silver.
> Gold where gold subscriber gets higher storage than silver subscriber

In the above I was assuming that the value of N is configurable. A given
AOR has some value. (Or maybe the whole CDIV service has a common
value.) And maybe its enough for that value to always be one.

The problem case is where the value of N is zero.

<Ranjit-Feb26> [4]

	Thanks,
	Paul

From pkyzivat@cisco.com  Fri Feb 26 06:34:19 2010
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Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert	review	ofdraft-avasarala-dispatch-comm-div-notification-03]
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Avasarala Ranjit-A20990 wrote:
> Hi Paul
> 
> My comments <Ranjit-Feb26>
> 
> [1] diverting-URI: I plan to remove this field from the list of elements
> that can be part of the notification body and filter criteria. 

OK.

> [2] phone-off and not subscribed-to: Here I mean, phone is off implies
> the subscriber is not registered to the IMS network at that moment. But
> the
>    subcription could still be valid. So when the subscriber's phone is
> switched off, the notifier will not be able to send the notification to
> the
>    subscriber and could store the notifications until the subscriber
> re-registers with the IMS network. The number of notifications that the
>    notifier would store would be policy dependent. 
> 
>    But if the subscriber is registered and not subscribed, then there is
> no way the notifier can send any notifications. A case of no
> subscription.

I'm still a little confused here.
If phone is on and subscription is active, then in general notification 
should be possible, regardless of registration. I realize IMS doesn't 
permit that, but from a general standard perspective we shouldn't assume 
it. The manifestation in this case is that the NOTIFY would fail and the 
subscription would eventually be torn down.

So I think the cases are:
- there is a subscription and notification works
- there is a subscription but notification fails
- there is no subscription.

> [3] in the example u suggested where sip:alice@atlanta.org with two
> contacts: sip:line1@alice.office.atlanta.com and
> sip:line2@alice.home.atlanta.com and
>     both the phones ring, then it would be a case of parallel forking.
> Now say one or both the phones have set diversion rules, to divert the

In this case, what does it mean for "one or both phones have set 
diversion rules"? Does that somehow imply distinct rules for each phone? 
I don't see how that could work. Aren't the diversion rules for the *AOR*?

I see they could be set/changed from either phone. But its a single set 
of rules isn't it? So I might set the rules from the office, and then 
cancel them from home.

> call to say 
>     sip:voicemail@alice.atlanta.com, then the notifications would get
> generated with respect to both the contacts.

Again I'm confused by the terminology you use, and how it relates 
operationally. The diversion happens on the AOR, and the subscriptions 
are on the AOR, and the notifications are sent to the subscribers. I 
don't see where the registered contacts enter into the picture at all.

If you have something else in mind, I hope you can explain it better in 
the draft.

> [4] Yes the value of N is configurable and depends on how many
> diversions the subscribe wants to keep or could be dependent on server
> policy (some maximum
>     number). If the server policy is to send notifications as and when
> they occur and not keep any divsersions information, then the value of
> last N 
>     diversions will be zero. But if the server chooses to retain the
> last diversion always, then the value of N will be 1 and so on.

All is straightforward for N>=1.

For N=0 things are messy. If we are modeling this as state, and 
notifications are about state changes, then the new diversion is added 
to the (previously empty) state and that triggers notifications to all 
current subscriptions. Then if you don't want to retain even this much 
state (because N=0) you immediately remove that version from the state. 
But that is again a state change, so you end up with another 
notification of that state change.

I presume you don't want that behavior. If you can live with N>=1 then 
we don't have the problem at all. If you need to support N=0 and don't 
want the extra notify, then we have to figure out if there is some trick 
to how things are modeled to get the desired effect. I've suggested 
something previously, but Adam didn't like it.

This needs sorting out.

	Thanks,
	Paul

> Regards
> Ranjit
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Friday, February 26, 2010 2:38 AM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
> ofdraft-avasarala-dispatch-comm-div-notification-03]
> 
> 
> 
> Avasarala Ranjit-A20990 wrote:
> 
>>>> But the diverting-URI is redundant - its equivalent to the resource 
>>>> that is the target of the subscription, and so will be constant 
>>>> throughout the subscription. I don't understand why it is present. 
>>>> My
>>>> suspicion is that you have in mind that the server will serve more 
>>>> than one diverting user and so the *server* state consists of info 
>>>> for each diverting user it serves. But unless you allow a 
>>>> subscription to address the *server* rather than a particular 
>>>> diverting user, this is
>>>> invisible to the state of a given resource subscription target.
>>>> <Ranjit-Feb23> Yes there is no need of diverting-URI. 
>>> OK. Then it could be removed from the notification body, and the 
>>> filter too, because it is redundant. Right?
>> <Ranjit-Feb25> yes we do not need diverting-URI in the notification 
>> body.
>>> Its being there confuses people like me, who assume that if its there
> 
>>> it must have a purpose.
> 
> If you want to leave it there, even though its redundant, the main issue
> is making it *clear* that it is redundant and will always match the
> subscribed-to URI.
> 
> The big problem is the potential that it might differ, and what it would
> mean if it did.
> <Ranjit-Feb26> [1]
> 
>> IIUC, you are saying that the server might store history of multiple 
>> diversions, or just the most recent one, or none.
>>
>>> If notifications of diversions are only delivered in realtime as they
> 
>>> occur, then why do you bother to include the time of the diversion in
> 
>>> the notification?
>>> <Ranjit-Feb23-2> true. This makes sense only when notifications are 
>>> stored and sent later (phone off scenario)
>> When you say "phone off", I presume you mean "phone not subscribed to 
>> this event package". Is that right?
>> <Ranjit-Feb25> True. This is ths phone state - being ON or OFF i.e 
>> registered with the network or not and then if it Is subscribed to 
>> CDIVN service or not.
> 
> Well, in IMS I guess if you aren't registered you can't be subscribed.
> But presumably you can be registered and not subscribed. Its the
> subscription that is the gating factor here, right?
> <Ranjit-Feb26> [2]
> 
>> What do you do about multiple phones (extensions) on the same AOR?
> E.g. 
>> one at the office and another at home. Consider that case, and that I 
>> have been out of the office for some time, and arrive home without 
>> visiting the office. My home phone has been off, but the office phone 
>> has been on the whole time. Will I see the diversions that have 
>> occurred while I was out?
>>
>> <Ranjit-Feb25> Here it depends on the target of the call. i.e if the 
>> call was targetted for your home phone and it was off, and there was a
> 
>> rule that Diverts all incoming calls to home be diverted, then you 
>> would get a notification.
> 
> I don't quite understand, and think it may be significant.
> I was talking about two phones registered to the same URI. E.g. AOR
> sip:alice@atlanta.org with two contacts: 
> sip:line1@alice.office.atlanta.com and sip:line2@alice.home.atlanta.com.
> 
> Hence both phones would be subscribing to the CDIV service at
> sip:alice@atlanta.org.
> 
> In that arrangement a caller can't (easily, without callerprefs or
> gruus) target a call to the home or the office. Rather the calls just
> ring both places. And then diversion rules will apply to calls to that
> AOR in general, so that both phones need to know about the diversion so
> that alice will know about it regardless of where she is.
> 
> But I think you are talking about some other arrangement, with multiple
> AORs that have some relationship to one another here, that isn't clear
> to me from the text. If that is so, can you explain?
> 
> <Ranjit-Feb26> [3]
> 
>> If you clear the state as soon as you send one notification with it, 
>> then when I get home I will see no diversions.
>> <Ranjit-Feb25> True. Once a notification about a particular diversion 
>> is sent, then the server's state is back to idle as if no diversions 
>> have occurred and nothing to do.
>>
>> It would all be much cleaner if you kept the most recent N diversions 
>> in the server, where N>=1. When a new one comes in, if that makes N+1,
> 
>> then discard the oldest, and *then* send the notification to whoever 
>> is subscribed. If somebody else subscribes later, send what you have.
>> <Ranjit-Feb25> This actually depends on how many diversions u want to 
>> store, server storage per subscriber - subscription types like silver.
>> Gold where gold subscriber gets higher storage than silver subscriber
> 
> In the above I was assuming that the value of N is configurable. A given
> AOR has some value. (Or maybe the whole CDIV service has a common
> value.) And maybe its enough for that value to always be one.
> 
> The problem case is where the value of N is zero.
> 
> <Ranjit-Feb26> [4]
> 
> 	Thanks,
> 	Paul
> 

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To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi Paul

Comments <Ranjit-Feb26-2>=20


Regards
Ranjit

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: Friday, February 26, 2010 8:07 PM
To: Avasarala Ranjit-A20990
Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review
ofdraft-avasarala-dispatch-comm-div-notification-03]



Avasarala Ranjit-A20990 wrote:
> Hi Paul
>=20
> My comments <Ranjit-Feb26>
>=20
> [1] diverting-URI: I plan to remove this field from the list of=20
> elements that can be part of the notification body and filter
criteria.

OK.
<Ranjit-Feb26-2> OK

> [2] phone-off and not subscribed-to: Here I mean, phone is off implies

> the subscriber is not registered to the IMS network at that moment.=20
> But the
>    subcription could still be valid. So when the subscriber's phone is

> switched off, the notifier will not be able to send the notification=20
> to the
>    subscriber and could store the notifications until the subscriber=20
> re-registers with the IMS network. The number of notifications that
the
>    notifier would store would be policy dependent.=20
>=20
>    But if the subscriber is registered and not subscribed, then there=20
> is no way the notifier can send any notifications. A case of no=20
> subscription.

I'm still a little confused here.
If phone is on and subscription is active, then in general notification
should be possible, regardless of registration. I realize IMS doesn't
permit that, but from a general standard perspective we shouldn't assume
it. The manifestation in this case is that the NOTIFY would fail and the
subscription would eventually be torn down.

So I think the cases are:
- there is a subscription and notification works
- there is a subscription but notification fails
- there is no subscription.

<Ranjit-Feb26-2> Agreed. Actually instead of calling it as Notification
fails, we can say - not sending notification due to (1) no match of
filter criteria (2) phone not registered or outside coverage area due to
which there is no connectivity and hence notification cannot be sent=20
Anyway if no subscription, no notification.=20

> [3] in the example u suggested where sip:alice@atlanta.org with two
> contacts: sip:line1@alice.office.atlanta.com and=20
> sip:line2@alice.home.atlanta.com and
>     both the phones ring, then it would be a case of parallel forking.
> Now say one or both the phones have set diversion rules, to divert the

In this case, what does it mean for "one or both phones have set
diversion rules"? Does that somehow imply distinct rules for each phone?

I don't see how that could work. Aren't the diversion rules for the
*AOR*?

<Ranjit-Feb26-2> Yes the diversion rules are for AoR. So when there is a
diversion rule set for the AoR and when diversions occur, the
notifications are
Sent to the AoR.

I see they could be set/changed from either phone. But its a single set
of rules isn't it? So I might set the rules from the office, and then
cancel them from home.

<Ranjit-Feb26-2> yes single set and could be set from either of the
phones. True.

> call to say=20
>     sip:voicemail@alice.atlanta.com, then the notifications would get=20
> generated with respect to both the contacts.

Again I'm confused by the terminology you use, and how it relates
operationally. The diversion happens on the AOR, and the subscriptions
are on the AOR, and the notifications are sent to the subscribers. I
don't see where the registered contacts enter into the picture at all.

<Ranjit-Feb26-2> Ya agree with you on this. Sorry for the confusion.

If you have something else in mind, I hope you can explain it better in
the draft.

> [4] Yes the value of N is configurable and depends on how many=20
> diversions the subscribe wants to keep or could be dependent on server

> policy (some maximum
>     number). If the server policy is to send notifications as and when

> they occur and not keep any divsersions information, then the value of

> last N
>     diversions will be zero. But if the server chooses to retain the=20
> last diversion always, then the value of N will be 1 and so on.

All is straightforward for N>=3D1.

For N=3D0 things are messy. If we are modeling this as state, and
notifications are about state changes, then the new diversion is added
to the (previously empty) state and that triggers notifications to all
current subscriptions. Then if you don't want to retain even this much
state (because N=3D0) you immediately remove that version from the =
state.=20
But that is again a state change, so you end up with another
notification of that state change.

<Ranjit-Feb26-2> True. Actually the notifications are for informing the
subscriber of the call diversions that happen. The notifications are not
for informing the subscriber of state changes in the notifier or the
number of diversions that occurred. As I mentioned in the abstract as
well as 24.604, the main intent of CDIVN service is to inform the
subscribers of communication diversions that occur on their behalf in
the network and this information will help them manage their rules
better.=20

<Ranjit-Feb26-2> But if we want to include the information related to
how many diversions have happened till time (say from 12 AM ) till the
time the notification is being sent, then as you say we can store the
diversions. This would be useful when the subscriber specifies in the
filter criteria to send notifications only during certain time like
between 5 and 6 PM and not whenever a diversion occurs. Then N is grater
than 1.

I presume you don't want that behavior. If you can live with N>=3D1 then
we don't have the problem at all. If you need to support N=3D0 and don't
want the extra notify, then we have to figure out if there is some trick
to how things are modeled to get the desired effect. I've suggested
something previously, but Adam didn't like it.

This needs sorting out.

	Thanks,
	Paul

> Regards
> Ranjit
>=20
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, February 26, 2010 2:38 AM
> To: Avasarala Ranjit-A20990
> Cc: Anders Kristensen; Adam Roach; dispatch@ietf.org
> Subject: Re: [dispatch] [Fwd: Re: Preliminary version ofExpert review=20
> ofdraft-avasarala-dispatch-comm-div-notification-03]
>=20
>=20
>=20
> Avasarala Ranjit-A20990 wrote:
>=20
>>>> But the diverting-URI is redundant - its equivalent to the resource

>>>> that is the target of the subscription, and so will be constant=20
>>>> throughout the subscription. I don't understand why it is present.
>>>> My
>>>> suspicion is that you have in mind that the server will serve more=20
>>>> than one diverting user and so the *server* state consists of info=20
>>>> for each diverting user it serves. But unless you allow a=20
>>>> subscription to address the *server* rather than a particular=20
>>>> diverting user, this is invisible to the state of a given resource=20
>>>> subscription target.
>>>> <Ranjit-Feb23> Yes there is no need of diverting-URI.=20
>>> OK. Then it could be removed from the notification body, and the=20
>>> filter too, because it is redundant. Right?
>> <Ranjit-Feb25> yes we do not need diverting-URI in the notification=20
>> body.
>>> Its being there confuses people like me, who assume that if its=20
>>> there
>=20
>>> it must have a purpose.
>=20
> If you want to leave it there, even though its redundant, the main=20
> issue is making it *clear* that it is redundant and will always match=20
> the subscribed-to URI.
>=20
> The big problem is the potential that it might differ, and what it=20
> would mean if it did.
> <Ranjit-Feb26> [1]
>=20
>> IIUC, you are saying that the server might store history of multiple=20
>> diversions, or just the most recent one, or none.
>>
>>> If notifications of diversions are only delivered in realtime as=20
>>> they
>=20
>>> occur, then why do you bother to include the time of the diversion=20
>>> in
>=20
>>> the notification?
>>> <Ranjit-Feb23-2> true. This makes sense only when notifications are=20
>>> stored and sent later (phone off scenario)
>> When you say "phone off", I presume you mean "phone not subscribed to

>> this event package". Is that right?
>> <Ranjit-Feb25> True. This is ths phone state - being ON or OFF i.e=20
>> registered with the network or not and then if it Is subscribed to=20
>> CDIVN service or not.
>=20
> Well, in IMS I guess if you aren't registered you can't be subscribed.
> But presumably you can be registered and not subscribed. Its the=20
> subscription that is the gating factor here, right?
> <Ranjit-Feb26> [2]
>=20
>> What do you do about multiple phones (extensions) on the same AOR?
> E.g.=20
>> one at the office and another at home. Consider that case, and that I

>> have been out of the office for some time, and arrive home without=20
>> visiting the office. My home phone has been off, but the office phone

>> has been on the whole time. Will I see the diversions that have=20
>> occurred while I was out?
>>
>> <Ranjit-Feb25> Here it depends on the target of the call. i.e if the=20
>> call was targetted for your home phone and it was off, and there was=20
>> a
>=20
>> rule that Diverts all incoming calls to home be diverted, then you=20
>> would get a notification.
>=20
> I don't quite understand, and think it may be significant.
> I was talking about two phones registered to the same URI. E.g. AOR=20
> sip:alice@atlanta.org with two contacts:
> sip:line1@alice.office.atlanta.com and
sip:line2@alice.home.atlanta.com.
>=20
> Hence both phones would be subscribing to the CDIV service at=20
> sip:alice@atlanta.org.
>=20
> In that arrangement a caller can't (easily, without callerprefs or
> gruus) target a call to the home or the office. Rather the calls just=20
> ring both places. And then diversion rules will apply to calls to that

> AOR in general, so that both phones need to know about the diversion=20
> so that alice will know about it regardless of where she is.
>=20
> But I think you are talking about some other arrangement, with=20
> multiple AORs that have some relationship to one another here, that=20
> isn't clear to me from the text. If that is so, can you explain?
>=20
> <Ranjit-Feb26> [3]
>=20
>> If you clear the state as soon as you send one notification with it,=20
>> then when I get home I will see no diversions.
>> <Ranjit-Feb25> True. Once a notification about a particular diversion

>> is sent, then the server's state is back to idle as if no diversions=20
>> have occurred and nothing to do.
>>
>> It would all be much cleaner if you kept the most recent N diversions

>> in the server, where N>=3D1. When a new one comes in, if that makes=20
>> N+1,
>=20
>> then discard the oldest, and *then* send the notification to whoever=20
>> is subscribed. If somebody else subscribes later, send what you have.
>> <Ranjit-Feb25> This actually depends on how many diversions u want to

>> store, server storage per subscriber - subscription types like
silver.
>> Gold where gold subscriber gets higher storage than silver subscriber
>=20
> In the above I was assuming that the value of N is configurable. A=20
> given AOR has some value. (Or maybe the whole CDIV service has a=20
> common
> value.) And maybe its enough for that value to always be one.
>=20
> The problem case is where the value of N is zero.
>=20
> <Ranjit-Feb26> [4]
>=20
> 	Thanks,
> 	Paul
>=20

From andrew.hutton@siemens-enterprise.com  Fri Feb 26 08:30:34 2010
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To: DISPATCH <dispatch@ietf.org>
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I have submitted the following draft which relates to the hopefully soon to=
 be formed media recording working group.

---------------------
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org] =
On Behalf Of Internet-Drafts@ietf.org
Sent: 26 February 2010 15:30
To: i-d-announce@ietf.org
Subject: I-D Action:draft-hutton-dispatch-session-recording-arch-00.txt=20

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : An Architecture for Media Recording using the Session In=
itiation Protocol
	Author(s)       : A. Hutton, et al.
	Filename        : draft-hutton-dispatch-session-recording-arch-00.txt
	Pages           : 14
	Date            : 2010-02-26

Session recording is a critical requirement in many communications
environments such as call centers and financial trading.  In some of
these environments, all calls must be recorded for regulatory,
compliance, and consumer protection reasons.  Recording of a session
is typically performed by sending a copy of a media stream to a
recording device.  This document describes architectures for
deploying session recording solutions in an environment which is
based on the Session Initiation Protocol (SIP).  This is an initial
draft which explores possible architectural solutions and has many
open issues.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-hutton-dispatch-session-recording=
-arch-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From D.Hancock@CableLabs.com  Fri Feb 26 11:55:20 2010
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Folks,

The following I-D has been updated based on previously received expert revi=
ew feedback from Milan Patel. Additional comments, if any, from interested =
participants are welcome and will be appreciated.

http://tools.ietf.org/html/draft-yu-tel-dai-08


>From the I-D:
"   This document defines a "dai" parameter for the "tel" Uniform
   Resource Identifier (URI) to support the Dial Around Indicator (DAI).
   The "dai" parameter is associated with the "cic" parameter, defined
   in [RFC4694], and indicates how the carrier identified in the "cic"
   parameter was selected.  This document also expands the use of the
   "cic" parameter to support pre-subscribed and dialed long-distance
   carrier requirements."

Thanks,
David


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<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Folks,<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>The following I-D has =
been
updated based on previously received expert review feedback from Milan Pate=
l.
Additional comments, if any, from interested participants are welcome and w=
ill
be appreciated.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><a
href=3D"http://tools.ietf.org/html/draft-yu-tel-dai-08">http://tools.ietf.o=
rg/html/draft-yu-tel-dai-08</a><o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>From the I-D:<o:p></o:=
p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&quot;&nbsp;&nbsp; Thi=
s
document defines a &quot;dai&quot; parameter for the &quot;tel&quot; Unifor=
m<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; Resource
Identifier (URI) to support the Dial Around Indicator (DAI).<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; The
&quot;dai&quot; parameter is associated with the &quot;cic&quot; parameter,
defined<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; in [RFC46=
94],
and indicates how the carrier identified in the &quot;cic&quot;<o:p></o:p><=
/span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; parameter=
 was
selected.&nbsp; This document also expands the use of the<o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; &quot;cic=
&quot;
parameter to support pre-subscribed and dialed long-distance<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp; carrier
requirements.&quot;<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'><o:p>&nbsp;</o:p></spa=
n></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>Thanks,<o:p></o:p></sp=
an></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 face=3D"C=
ourier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>David<o:p></o:p></span=
></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

</div>

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