
From adam@nostrum.com  Mon Apr  2 08:40:43 2012
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From: Adam Roach - SIPCORE Chair <adam@nostrum.com>
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Subject: [sipcore] 3rd WGLC: draft-ietf-sipcore-rfc4244bis-07
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[as chair]

SIPCORE working group:

This message is notice of a third (and hopefully final) working group 
last call for the History-Info document (draft-ietf-sipcore-rfc4244bis-07):

http://www.ietf.org/id/draft-ietf-sipcore-rfc4244bis-07.txt

This working group last call will conclude on Monday, April 16th. 
Interested parties should carefully review the associated draft and 
provide feedback on the SIPCORE list prior to that date.

Thank you.

/a

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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Subject: Re: [sipcore] Confirming Consensus: WebSockets
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[as a participant]

Hadriel:

I understand the practicalities you're trying to highlight, but I don't 
think it's reasonable to work around the not-formally-defined and 
constantly shifting behavior of arguably noncompliant intermediaries at 
this point. Maybe STRAW can take on an item to define out-of-dialog 
request routing behaviors, and discuss GRUU interaction; then we can 
talk in terms of intermediaries that implement those specific procedures.

For the time being, I think we need to work with the standardized tools 
that have been defined -- tools that work just fine in an actual 
standards-compliant network -- and let those entities that break proper 
routing deal with the consequences through whatever proprietary 
band-aids amuse them.

In this context, that would (IMHO) involve a discussion of the problems 
solved by GRUU, and a pointer to the document. I agree with Christer 
that -- like outbound -- we don't need to mandate GRUU; we simply need 
to describe what happens without it, and let implementors make their own 
evaluation about whether those properties warrant the additional 
implementation complexity.

/a

On 3/29/12 12:40 PM, Hadriel Kaplan wrote:
> On Mar 28, 2012, at 10:56 PM, Paul Kyzivat wrote:
>
>> Please correct me if I'm wrong, but I *think* what you are trying to say is that out of dialog requests to the contact address from a dialog will mostly never work, unless somebody puts the phone-number-style URI in the contact address, or if the contact is one that was substituted by the last SBC.
> I'm not quite sure what *you're* saying, so I can't tell if it's what I'm also trying to say. :)
>
>
>> I would think, *if they wanted to*, SBCs could make whatever substitution is necessary to make a gruu that they receive in a contact address be usable on the other side. This would be in their traditional role of fixing all the interop problems of the networks they connect.
>
> Sort of.  There are some folks in the IETF who don't want the SBCs to change them at all if they're GRUUs.  And there are also other SDOs/groups that seem to think they will be getting GRUUs, that they MUST get GRUUs, and that it will somehow make things magically work.
>
> -hadriel
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From adam@nostrum.com  Mon Apr  2 09:09:05 2012
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Subject: Re: [sipcore] Some considerations on draft-ibc-sipcore-sip-websocket (Outbound and more) - GRUU
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On 3/26/12 3:46 PM, Hadriel Kaplan wrote:
> On Mar 26, 2012, at 2:25 PM, Kevin P. Fleming wrote:
>
>> For nearly *all* WebSocket use cases (that we've agreed to address with this draft), SIP endpoints will have this issue. Since the WebSocket transport is being documented *after* the creation of the solution (GRUU), it seems reasonable to me for the WebSocket transport document to refer to that solution as the preferred mechanism. It shouldn't be mandated, but preferring it seems practical and useful.
> I would agree, if it weren't for that I wrote a draft of problems with using GRUUs.  :)
>
> See here:
> http://tools.ietf.org/html/draft-kaplan-dispatch-gruu-problematic-00
>
> And before Paul says anything, the draft is NOT blaming GRUUs.  It's not their fault the Internet is a nasty place. ;)
>

Well, be honest about the contents of the document, though. It boils 
down to: (a) here are some poorly-thought-out things that B2BUAs can do 
to ruin the beneficial properties of GRUUs, and (b) broken deployments 
are broken.

It's more of a "worst common practices" document than an indictment of 
GRUUs. The solution to pretty much all the issues that draft highlights 
is: "don't do that, then." It's not the GRUUs that are broken -- it's 
the deployments and poorly-designed B2BUAs.

/a

From victor.pascual.avila@gmail.com  Mon Apr  2 09:10:26 2012
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Subject: Re: [sipcore] Confirming Consensus: WebSockets
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On Mon, Apr 2, 2012 at 5:52 PM, Adam Roach <adam@nostrum.com> wrote:
> I agree with Christer that -- like outbound -- we don't need to mandate
> GRUU; we simply need to describe what happens without it, and let
> implementors make their own evaluation about whether those properties
> warrant the additional implementation complexity.

Agreed -- this is what we're actually planning to do for the next
revision of the document

-Victor

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Subject: Re: [sipcore] Some considerations on draft-ibc-sipcore-sip-websocket (Outbound and more) - GRUU
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2012/4/2 Adam Roach <adam@nostrum.com>:
> Well, be honest about the contents of the document, though. It boils down
> to: (a) here are some poorly-thought-out things that B2BUAs can do to rui=
n
> the beneficial properties of GRUUs, and (b) broken deployments are broken=
.
>
> It's more of a "worst common practices" document than an indictment of
> GRUUs. The solution to pretty much all the issues that draft highlights i=
s:
> "don't do that, then." It's not the GRUUs that are broken -- it's the
> deployments and poorly-designed B2BUAs.

Completely agreed. I don't understand why we should discuss here about
the fact that non standard devices (B2BUAs, SBCs) do not play well
with real standard specifications (Outbound, GRUU). Neither we should
consider here (when talking about standards) the fact that some SIP
servers implement their proprietary "version" of Outbound and/or
Connection Reuse.

Outbound and GRUU do work in well designed standard networks.

Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
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Date: Mon, 2 Apr 2012 12:53:25 -0400
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Bruno Chatras has pointed out that a number of examples in various SIP-rela=
ted
RFCs show Content-Length header fields in the headers of body-parts of mult=
ipart
entities.  MIME does not define the meaining of such header fields, as the =
MIME
framing of body parts is done by the boundary markers.

As Dale Moberg has pointed out, these header fields might not be truly inco=
rrect
but merely redundant or "to be ignored".

But given that MIME multipart already frames the body-parts, Content-Length
headers on body-parts are never necessary.

So it seems to me that we should decide on a "best practices" for the use o=
f
Content-Length in the headers of a body part, to clarify the ambiguities in=
 the
specifications, and to state whether it's preferable to include such a Cont=
ent-Length
or not.

I've appended a list of all the examples carrying such Content-Length heade=
rs
that I could find by automated means.

Dale
---------------------------------------------------------------------------=
-----------------------------
RFC 2848

                       The PINT Service Protocol:
   Extensions to SIP and SDP for IP Access to Telephone Call Services

Suspicious line at rfc2848.txt:2195:       Content-Length: 236
Suspicious line at rfc2848.txt:2209:       Content-Length:50
Suspicious line at rfc2848.txt:2268:       Content-Length: 316
Suspicious line at rfc2848.txt:2284:       Content-Length: 172
Suspicious line at rfc2848.txt:2372:    Content-Length: 325
Suspicious line at rfc2848.txt:2387:    Content-Length: 352

Several of these body-part headers are not separated from the
body-part contents by CR-LF-CR-LF.

RFC 3261

                    SIP: Session Initiation Protocol

Suspicious line at rfc3261.txt:11846:         Content-Length: 231

RFC 3892

      The Session Initiation Protocol (SIP) Referred-By Mechanism

Suspicious line at rfc3892.txt:642:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:687:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:702:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:879:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:887:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:938:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:964:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:972:       Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:1056:      Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:1106:      Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:1168:      Content-Length: (appropriate valu=
e)
Suspicious line at rfc3892.txt:1196:      Content-Length: (appropriate valu=
e)

RFC 3893

                   Session Initiation Protocol (SIP)
                Authenticated Identity Body (AIB) Format

Suspicious line at rfc3893.txt:249:    Content-Length: 147
(This has an extra CR-LF before the body-part headers.)
Suspicious line at rfc3893.txt:263:    Content-Length: 608

RFC 4463

                A Media Resource Control Protocol (MRCP)
              Developed by Cisco, Nuance, and Speechworks

Suspicious line at rfc4463.txt:1705:        Content-Length:176
Suspicious line at rfc4463.txt:1714:        Content-Length:104
Suspicious line at rfc4463.txt:2977:       Content-Length:176
(Also missing the CR-LF after the body part headers.)
Suspicious line at rfc4463.txt:2985:       Content-Length:104

RFC 5547

      A Session Description Protocol (SDP) Offer/Answer Mechanism
                        to Enable File Transfer

Suspicious line at rfc5547.txt:1645:    Content-Length: [length of SDP]
Suspicious line at rfc5547.txt:1670:    Content-Length: [length of image]
Suspicious line at rfc5547.txt:2093:    Content-Length: [length of SDP]
Suspicious line at rfc5547.txt:2118:    Content-Length: [length of image]

RFC 5589

       Session Initiation Protocol (SIP) Call Control - Transfer

Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
(An extra CR-LF in the middle of the body part headers.)
Suspicious line at rfc5589.txt:2677:    Content-Length: 156
(An extra CR-LF in the middle of the body part headers.)
Suspicious line at rfc5589.txt:2696:    Content-Length: 2961

RFC 6086

  Session Initiation Protocol (SIP) INFO Method and Package Framework

Suspicious line at rfc6086.txt:1594:    Content-length: 59
Suspicious line at rfc6086.txt:1620:    Content-length: 59
Suspicious line at rfc6086.txt:1637:    Content-length: 59
Suspicious line at rfc6086.txt:1664:    Content-length: 59

From brett@broadsoft.com  Mon Apr  2 10:52:39 2012
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From: Brett Tate <brett@broadsoft.com>
To: SIPCORE <sipcore@ietf.org>, "draft-ietf-sipcore-rfc4244bis@tools.ietf.org" <draft-ietf-sipcore-rfc4244bis@tools.ietf.org>
Date: Mon, 2 Apr 2012 10:52:35 -0700
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Howdy,

The following are some draft-ietf-sipcore-rfc4244bis-07 comments.

Thanks,
Brett

------

Section 16.1: "changed rom" should be "changed from".

Appendix B:

- If depicting an RFC 3261 compliant UAS, the 302/486/ACK examples must inc=
lude To tags.

- If depicting an RFC 3261 compliant UAS, the Via headers using an FQDN mus=
t cause the UAS to add a received parameter to the Via.

Section B.1:

- In case not intentional, the 302's ACK is not depicted within the call fl=
ow.

- F3: Missing top Via entry of F2.

- F4: The top unique via branch should be changed since it was already used=
 within F2.

- F5: Missing top Via entry of F4.  The Contact should be changed since it =
reflects the proxy instead of UAS location.

- F6: The top unique via branch should be changed since it was already used=
 within F2.

- F7: Missing top Via entry of F6.  The Contact should be changed since it =
reflects the proxy instead of UAS location.

Section B.2:

- In case not intentional, the 302's ACK is not depicted within the call fl=
ow.

- F3: Missing top Via entry of F2.

- F4: The top unique via branch should be changed since it was already used=
 within F2.

- F5: Missing top Via entry of F4.  The Contact should be changed since it =
reflects the proxy instead of UAS location.

- F6: The top unique via branch should be changed since it was already used=
 within F2.

- F7: Missing top Via entry of F6.  The Contact should be changed since it =
reflects the proxy instead of UAS location.


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From pkyzivat@alum.mit.edu  Mon Apr  2 11:12:29 2012
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[reposting with a subject line]

On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
> Bruno Chatras has pointed out that a number of examples in various SIP-related
> RFCs show Content-Length header fields in the headers of body-parts of multipart
> entities.  MIME does not define the meaining of such header fields, as the MIME
> framing of body parts is done by the boundary markers.
>
> As Dale Moberg has pointed out, these header fields might not be truly incorrect
> but merely redundant or "to be ignored".
>
> But given that MIME multipart already frames the body-parts, Content-Length
> headers on body-parts are never necessary.
>
> So it seems to me that we should decide on a "best practices" for the use of
> Content-Length in the headers of a body part, to clarify the ambiguities in the
> specifications, and to state whether it's preferable to include such a Content-Length
> or not.
>
> I've appended a list of all the examples carrying such Content-Length headers
> that I could find by automated means.
>
> Dale
> --------------------------------------------------------------------------------------------------------
> RFC 2848
>
>                         The PINT Service Protocol:
>     Extensions to SIP and SDP for IP Access to Telephone Call Services
>
> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
> Suspicious line at rfc2848.txt:2209:       Content-Length:50
> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
>
> Several of these body-part headers are not separated from the
> body-part contents by CR-LF-CR-LF.
>
> RFC 3261
>
>                      SIP: Session Initiation Protocol
>
> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
>
> RFC 3892
>
>        The Session Initiation Protocol (SIP) Referred-By Mechanism
>
> Suspicious line at rfc3892.txt:642:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:687:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:702:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:879:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:887:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:938:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:964:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:972:       Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:1056:      Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:1106:      Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:1168:      Content-Length: (appropriate value)
> Suspicious line at rfc3892.txt:1196:      Content-Length: (appropriate value)
>
> RFC 3893
>
>                     Session Initiation Protocol (SIP)
>                  Authenticated Identity Body (AIB) Format
>
> Suspicious line at rfc3893.txt:249:    Content-Length: 147
> (This has an extra CR-LF before the body-part headers.)
> Suspicious line at rfc3893.txt:263:    Content-Length: 608
>
> RFC 4463
>
>                  A Media Resource Control Protocol (MRCP)
>                Developed by Cisco, Nuance, and Speechworks
>
> Suspicious line at rfc4463.txt:1705:        Content-Length:176
> Suspicious line at rfc4463.txt:1714:        Content-Length:104
> Suspicious line at rfc4463.txt:2977:       Content-Length:176
> (Also missing the CR-LF after the body part headers.)
> Suspicious line at rfc4463.txt:2985:       Content-Length:104
>
> RFC 5547
>
>        A Session Description Protocol (SDP) Offer/Answer Mechanism
>                          to Enable File Transfer
>
> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of SDP]
> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of image]
> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of SDP]
> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of image]
>
> RFC 5589
>
>         Session Initiation Protocol (SIP) Call Control - Transfer
>
> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
> (An extra CR-LF in the middle of the body part headers.)
> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
> (An extra CR-LF in the middle of the body part headers.)
> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
>
> RFC 6086
>
>    Session Initiation Protocol (SIP) INFO Method and Package Framework
>
> Suspicious line at rfc6086.txt:1594:    Content-length: 59
> Suspicious line at rfc6086.txt:1620:    Content-length: 59
> Suspicious line at rfc6086.txt:1637:    Content-length: 59
> Suspicious line at rfc6086.txt:1664:    Content-length: 59
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From HKaplan@acmepacket.com  Mon Apr  2 15:21:55 2012
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "Dale R (Dale) Worley" <dworley@avaya.com>
Thread-Topic: [sipcore] Content-Length in multipart body parts
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Date: Mon, 2 Apr 2012 22:21:38 +0000
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Subject: Re: [sipcore] Content-Length in multipart body parts
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Ugh.  Yeah, that's not good for examples to show.
So what's the solution?  Errata?

-hadriel
p.s. I like the "Suspicious Line" thing - all RFC lines are suspicious! ;)


On Apr 2, 2012, at 2:12 PM, Paul Kyzivat wrote:

> [reposting with a subject line]
>=20
> On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
>> Bruno Chatras has pointed out that a number of examples in various SIP-r=
elated
>> RFCs show Content-Length header fields in the headers of body-parts of m=
ultipart
>> entities.  MIME does not define the meaining of such header fields, as t=
he MIME
>> framing of body parts is done by the boundary markers.
>>=20
>> As Dale Moberg has pointed out, these header fields might not be truly i=
ncorrect
>> but merely redundant or "to be ignored".
>>=20
>> But given that MIME multipart already frames the body-parts, Content-Len=
gth
>> headers on body-parts are never necessary.
>>=20
>> So it seems to me that we should decide on a "best practices" for the us=
e of
>> Content-Length in the headers of a body part, to clarify the ambiguities=
 in the
>> specifications, and to state whether it's preferable to include such a C=
ontent-Length
>> or not.
>>=20
>> I've appended a list of all the examples carrying such Content-Length he=
aders
>> that I could find by automated means.
>>=20
>> Dale
>> ------------------------------------------------------------------------=
--------------------------------
>> RFC 2848
>>=20
>>                        The PINT Service Protocol:
>>    Extensions to SIP and SDP for IP Access to Telephone Call Services
>>=20
>> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
>> Suspicious line at rfc2848.txt:2209:       Content-Length:50
>> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
>> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
>> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
>> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
>>=20
>> Several of these body-part headers are not separated from the
>> body-part contents by CR-LF-CR-LF.
>>=20
>> RFC 3261
>>=20
>>                     SIP: Session Initiation Protocol
>>=20
>> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
>>=20
>> RFC 3892
>>=20
>>       The Session Initiation Protocol (SIP) Referred-By Mechanism
>>=20
>> Suspicious line at rfc3892.txt:642:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:687:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:702:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:879:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:887:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:938:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:964:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:972:       Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:1056:      Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:1106:      Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:1168:      Content-Length: (appropriate v=
alue)
>> Suspicious line at rfc3892.txt:1196:      Content-Length: (appropriate v=
alue)
>>=20
>> RFC 3893
>>=20
>>                    Session Initiation Protocol (SIP)
>>                 Authenticated Identity Body (AIB) Format
>>=20
>> Suspicious line at rfc3893.txt:249:    Content-Length: 147
>> (This has an extra CR-LF before the body-part headers.)
>> Suspicious line at rfc3893.txt:263:    Content-Length: 608
>>=20
>> RFC 4463
>>=20
>>                 A Media Resource Control Protocol (MRCP)
>>               Developed by Cisco, Nuance, and Speechworks
>>=20
>> Suspicious line at rfc4463.txt:1705:        Content-Length:176
>> Suspicious line at rfc4463.txt:1714:        Content-Length:104
>> Suspicious line at rfc4463.txt:2977:       Content-Length:176
>> (Also missing the CR-LF after the body part headers.)
>> Suspicious line at rfc4463.txt:2985:       Content-Length:104
>>=20
>> RFC 5547
>>=20
>>       A Session Description Protocol (SDP) Offer/Answer Mechanism
>>                         to Enable File Transfer
>>=20
>> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of SDP]
>> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of image=
]
>> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of SDP]
>> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of image=
]
>>=20
>> RFC 5589
>>=20
>>        Session Initiation Protocol (SIP) Call Control - Transfer
>>=20
>> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
>> (An extra CR-LF in the middle of the body part headers.)
>> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
>> (An extra CR-LF in the middle of the body part headers.)
>> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
>>=20
>> RFC 6086
>>=20
>>   Session Initiation Protocol (SIP) INFO Method and Package Framework
>>=20
>> Suspicious line at rfc6086.txt:1594:    Content-length: 59
>> Suspicious line at rfc6086.txt:1620:    Content-length: 59
>> Suspicious line at rfc6086.txt:1637:    Content-length: 59
>> Suspicious line at rfc6086.txt:1664:    Content-length: 59
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From pkyzivat@alum.mit.edu  Mon Apr  2 15:37:15 2012
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Cc: "Dale R \(Dale\) Worley" <dworley@avaya.com>, "<sipcore@ietf.org>" <sipcore@ietf.org>
Subject: Re: [sipcore] Content-Length in multipart body parts
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On 4/2/12 6:21 PM, Hadriel Kaplan wrote:
>
> Ugh.  Yeah, that's not good for examples to show.
> So what's the solution?  Errata?

Yes, I think the errata process is the right one for this.

	Thanks,
	Paul

> -hadriel
> p.s. I like the "Suspicious Line" thing - all RFC lines are suspicious! ;)
>
>
> On Apr 2, 2012, at 2:12 PM, Paul Kyzivat wrote:
>
>> [reposting with a subject line]
>>
>> On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
>>> Bruno Chatras has pointed out that a number of examples in various SIP-related
>>> RFCs show Content-Length header fields in the headers of body-parts of multipart
>>> entities.  MIME does not define the meaining of such header fields, as the MIME
>>> framing of body parts is done by the boundary markers.
>>>
>>> As Dale Moberg has pointed out, these header fields might not be truly incorrect
>>> but merely redundant or "to be ignored".
>>>
>>> But given that MIME multipart already frames the body-parts, Content-Length
>>> headers on body-parts are never necessary.
>>>
>>> So it seems to me that we should decide on a "best practices" for the use of
>>> Content-Length in the headers of a body part, to clarify the ambiguities in the
>>> specifications, and to state whether it's preferable to include such a Content-Length
>>> or not.
>>>
>>> I've appended a list of all the examples carrying such Content-Length headers
>>> that I could find by automated means.
>>>
>>> Dale
>>> --------------------------------------------------------------------------------------------------------
>>> RFC 2848
>>>
>>>                         The PINT Service Protocol:
>>>     Extensions to SIP and SDP for IP Access to Telephone Call Services
>>>
>>> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
>>> Suspicious line at rfc2848.txt:2209:       Content-Length:50
>>> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
>>> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
>>> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
>>> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
>>>
>>> Several of these body-part headers are not separated from the
>>> body-part contents by CR-LF-CR-LF.
>>>
>>> RFC 3261
>>>
>>>                      SIP: Session Initiation Protocol
>>>
>>> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
>>>
>>> RFC 3892
>>>
>>>        The Session Initiation Protocol (SIP) Referred-By Mechanism
>>>
>>> Suspicious line at rfc3892.txt:642:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:687:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:702:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:879:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:887:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:938:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:964:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:972:       Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:1056:      Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:1106:      Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:1168:      Content-Length: (appropriate value)
>>> Suspicious line at rfc3892.txt:1196:      Content-Length: (appropriate value)
>>>
>>> RFC 3893
>>>
>>>                     Session Initiation Protocol (SIP)
>>>                  Authenticated Identity Body (AIB) Format
>>>
>>> Suspicious line at rfc3893.txt:249:    Content-Length: 147
>>> (This has an extra CR-LF before the body-part headers.)
>>> Suspicious line at rfc3893.txt:263:    Content-Length: 608
>>>
>>> RFC 4463
>>>
>>>                  A Media Resource Control Protocol (MRCP)
>>>                Developed by Cisco, Nuance, and Speechworks
>>>
>>> Suspicious line at rfc4463.txt:1705:        Content-Length:176
>>> Suspicious line at rfc4463.txt:1714:        Content-Length:104
>>> Suspicious line at rfc4463.txt:2977:       Content-Length:176
>>> (Also missing the CR-LF after the body part headers.)
>>> Suspicious line at rfc4463.txt:2985:       Content-Length:104
>>>
>>> RFC 5547
>>>
>>>        A Session Description Protocol (SDP) Offer/Answer Mechanism
>>>                          to Enable File Transfer
>>>
>>> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of SDP]
>>> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of image]
>>> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of SDP]
>>> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of image]
>>>
>>> RFC 5589
>>>
>>>         Session Initiation Protocol (SIP) Call Control - Transfer
>>>
>>> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
>>> (An extra CR-LF in the middle of the body part headers.)
>>> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
>>> (An extra CR-LF in the middle of the body part headers.)
>>> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
>>>
>>> RFC 6086
>>>
>>>    Session Initiation Protocol (SIP) INFO Method and Package Framework
>>>
>>> Suspicious line at rfc6086.txt:1594:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1620:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1637:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1664:    Content-length: 59
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
>


From shida@ntt-at.com  Mon Apr  2 17:39:49 2012
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To: Brett Tate <brett@broadsoft.com>
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Cc: "draft-ietf-sipcore-rfc4244bis@tools.ietf.org" <draft-ietf-sipcore-rfc4244bis@tools.ietf.org>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-07: comments
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Brett;

 Many thanks for having a close look at the example messages.

 We will incorporate these changes.

 Many Thanks
  Shida

On Apr 2, 2012, at 7:52 PM, Brett Tate wrote:

> Howdy,
>=20
> The following are some draft-ietf-sipcore-rfc4244bis-07 comments.
>=20
> Thanks,
> Brett
>=20
> ------
>=20
> Section 16.1: "changed rom" should be "changed from".
>=20
> Appendix B:
>=20
> - If depicting an RFC 3261 compliant UAS, the 302/486/ACK examples =
must include To tags.
>=20
> - If depicting an RFC 3261 compliant UAS, the Via headers using an =
FQDN must cause the UAS to add a received parameter to the Via.
>=20
> Section B.1:
>=20
> - In case not intentional, the 302's ACK is not depicted within the =
call flow.
>=20
> - F3: Missing top Via entry of F2.
>=20
> - F4: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F5: Missing top Via entry of F4.  The Contact should be changed =
since it reflects the proxy instead of UAS location.
>=20
> - F6: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F7: Missing top Via entry of F6.  The Contact should be changed =
since it reflects the proxy instead of UAS location.
>=20
> Section B.2:
>=20
> - In case not intentional, the 302's ACK is not depicted within the =
call flow.
>=20
> - F3: Missing top Via entry of F2.
>=20
> - F4: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F5: Missing top Via entry of F4.  The Contact should be changed =
since it reflects the proxy instead of UAS location.
>=20
> - F6: The top unique via branch should be changed since it was already =
used within F2.
>=20
> - F7: Missing top Via entry of F6.  The Contact should be changed =
since it reflects the proxy instead of UAS location.
>=20
>=20
> This email is intended solely for the person or entity to which it is =
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From pkyzivat@alum.mit.edu  Mon Apr  2 21:20:47 2012
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Date: Tue, 03 Apr 2012 00:20:43 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Confirming Consensus: WebSockets
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[as individual]

On 3/29/12 1:40 PM, Hadriel Kaplan wrote:
>
> On Mar 28, 2012, at 10:56 PM, Paul Kyzivat wrote:
>
>> Please correct me if I'm wrong, but I *think* what you are trying to say is that out of dialog requests to the contact address from a dialog will mostly never work, unless somebody puts the phone-number-style URI in the contact address, or if the contact is one that was substituted by the last SBC.
>
> I'm not quite sure what *you're* saying, so I can't tell if it's what I'm also trying to say. :)

OK, I'll try again, and at the same time back off from some assumptions 
I was making about what you were saying.

The point of a UA using a Gruu (a gruu (globally routable ua uri) 
assigned by a registrar according to 5627) as its contact address in a 
dialog is so that it knows that it can be reached out-of-dialog by any 
UA that receives its contact address.

When you disparage using Gruus in this way, without offering a reference 
to an alternative, I guess you are saying the UA should just use its 
own, non-gruu URI, and rely on "somebody else" to "make it better" in 
some unspecified, magic way. Is that right? How is the UA to know this 
will happen? How does it know that it will work in all the cases that 
matter to it?

>> I would think, *if they wanted to*, SBCs could make whatever substitution is necessary to make a gruu that they receive in a contact address be usable on the other side. This would be in their traditional role of fixing all the interop problems of the networks they connect.
>
> Sort of.  There are some folks in the IETF who don't want the SBCs to change them at all if they're GRUUs.

Yes. I think that if they are constructed to be globally routable then 
they should not need to be changed in any 3261 conforming network.

I guess I would not object if an SBC did something to a Gruu to make it 
work in an environment that was *not* conforming and where the Gruu 
would otherwise not work. I would also expect that network to *not* 
advertise itself as conforming to sip.

> And there are also other SDOs/groups that seem to think they will be getting GRUUs,
> that they MUST get GRUUs, and that it will somehow make things magically work.

Well, 3261 says that UAs MUST use a contact URI with global scope. 5627 
was created because many UAs were not following this because they had no 
access to such a URI.

The 3261 requirement is not to use a Gruu assigned by a registrar 
according to 5627. The UA can get one any way it wants. The registrar 
mechanism is just one way to achieve that.

But its pretty sad if an SBC goes out of its way to break a UA that is 
trying hard to do what it is required to do.

Now this doesn't mean I'm blaming *you*, or *Acme*, or any SBC vendor, 
for doing this. As you repeatedly stress, and I understand, you are just 
doing what your customers ask you to do. So its *them* that I blame for 
this.

If they are going to require non-conforming behavior, or "fix" 
conforming behavior to be non-conforming, they they should make it clear 
to *their* customers that they are operating a closed environment that 
is not sip-conforming.

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Mon Apr  2 23:39:44 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, Hadriel Kaplan <HKaplan@acmepacket.com>
Date: Tue, 3 Apr 2012 08:39:38 +0200
Thread-Topic: [sipcore] Content-Length in multipart body parts
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Cc: "Dale R \(Dale\) Worley" <dworley@avaya.com>, "<sipcore@ietf.org>" <sipcore@ietf.org>
Subject: Re: [sipcore] Content-Length in multipart body parts
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Hi,

>> Ugh.  Yeah, that's not good for examples to show.
>> So what's the solution?  Errata?
>
> Yes, I think the errata process is the right one for this.

Agree, eventhough the C-L should cause no conflicts (as long as the value i=
s correct).

And, even if a specific MIME contains another (nested) multipart entry, C-L=
 should not be needed, as the MIME borders are still present.

Regards,

Christer


> -hadriel
> p.s. I like the "Suspicious Line" thing - all RFC lines are=20
> suspicious! ;)
>
>
> On Apr 2, 2012, at 2:12 PM, Paul Kyzivat wrote:
>
>> [reposting with a subject line]
>>
>> On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
>>> Bruno Chatras has pointed out that a number of examples in various=20
>>> SIP-related RFCs show Content-Length header fields in the headers of=20
>>> body-parts of multipart entities.  MIME does not define the meaining=20
>>> of such header fields, as the MIME framing of body parts is done by the=
 boundary markers.
>>>
>>> As Dale Moberg has pointed out, these header fields might not be=20
>>> truly incorrect but merely redundant or "to be ignored".
>>>
>>> But given that MIME multipart already frames the body-parts,=20
>>> Content-Length headers on body-parts are never necessary.
>>>
>>> So it seems to me that we should decide on a "best practices" for=20
>>> the use of Content-Length in the headers of a body part, to clarify=20
>>> the ambiguities in the specifications, and to state whether it's=20
>>> preferable to include such a Content-Length or not.
>>>
>>> I've appended a list of all the examples carrying such=20
>>> Content-Length headers that I could find by automated means.
>>>
>>> Dale
>>> --------------------------------------------------------------------
>>> ------------------------------------
>>> RFC 2848
>>>
>>>                         The PINT Service Protocol:
>>>     Extensions to SIP and SDP for IP Access to Telephone Call=20
>>> Services
>>>
>>> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
>>> Suspicious line at rfc2848.txt:2209:       Content-Length:50
>>> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
>>> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
>>> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
>>> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
>>>
>>> Several of these body-part headers are not separated from the=20
>>> body-part contents by CR-LF-CR-LF.
>>>
>>> RFC 3261
>>>
>>>                      SIP: Session Initiation Protocol
>>>
>>> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
>>>
>>> RFC 3892
>>>
>>>        The Session Initiation Protocol (SIP) Referred-By Mechanism
>>>
>>> Suspicious line at rfc3892.txt:642:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:687:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:702:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:879:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:887:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:938:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:964:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:972:       Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:1056:      Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:1106:      Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:1168:      Content-Length: (appropriate =
value)
>>> Suspicious line at rfc3892.txt:1196:      Content-Length: (appropriate =
value)
>>>
>>> RFC 3893
>>>
>>>                     Session Initiation Protocol (SIP)
>>>                  Authenticated Identity Body (AIB) Format
>>>
>>> Suspicious line at rfc3893.txt:249:    Content-Length: 147
>>> (This has an extra CR-LF before the body-part headers.)
>>> Suspicious line at rfc3893.txt:263:    Content-Length: 608
>>>
>>> RFC 4463
>>>
>>>                  A Media Resource Control Protocol (MRCP)
>>>                Developed by Cisco, Nuance, and Speechworks
>>>
>>> Suspicious line at rfc4463.txt:1705:        Content-Length:176
>>> Suspicious line at rfc4463.txt:1714:        Content-Length:104
>>> Suspicious line at rfc4463.txt:2977:       Content-Length:176
>>> (Also missing the CR-LF after the body part headers.)
>>> Suspicious line at rfc4463.txt:2985:       Content-Length:104
>>>
>>> RFC 5547
>>>
>>>        A Session Description Protocol (SDP) Offer/Answer Mechanism
>>>                          to Enable File Transfer
>>>
>>> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of SDP]
>>> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of imag=
e]
>>> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of SDP]
>>> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of imag=
e]
>>>
>>> RFC 5589
>>>
>>>         Session Initiation Protocol (SIP) Call Control - Transfer
>>>
>>> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
>>> (An extra CR-LF in the middle of the body part headers.)
>>> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
>>> (An extra CR-LF in the middle of the body part headers.)
>>> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
>>>
>>> RFC 6086
>>>
>>>    Session Initiation Protocol (SIP) INFO Method and Package=20
>>> Framework
>>>
>>> Suspicious line at rfc6086.txt:1594:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1620:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1637:    Content-length: 59
>>> Suspicious line at rfc6086.txt:1664:    Content-length: 59
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>
>

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

From bruno.chatras@orange.com  Tue Apr  3 02:09:14 2012
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From: <bruno.chatras@orange.com>
To: <christer.holmberg@ericsson.com>, <pkyzivat@alum.mit.edu>, <HKaplan@acmepacket.com>
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Cc: dworley@avaya.com, sipcore@ietf.org
Subject: Re: [sipcore] Content-Length in multipart body parts
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I assume you mean a set of errata to fix the "Suspicious" examples. If =
Yes, I do agree as well.

> -----Message d'origine-----
> De=A0: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] De =
la
> part de Christer Holmberg
> Envoy=E9=A0: mardi 3 avril 2012 08:40
> =C0=A0: Paul Kyzivat; Hadriel Kaplan
> Cc=A0: Dale R (Dale) Worley; <sipcore@ietf.org>
> Objet=A0: Re: [sipcore] Content-Length in multipart body parts
>=20
> Hi,
>=20
> >> Ugh.  Yeah, that's not good for examples to show.
> >> So what's the solution?  Errata?
> >
> > Yes, I think the errata process is the right one for this.
>=20
> Agree, eventhough the C-L should cause no conflicts (as long as the
> value is correct).
>=20
> And, even if a specific MIME contains another (nested) multipart =
entry,
> C-L should not be needed, as the MIME borders are still present.
>=20
> Regards,
>=20
> Christer
>=20
>=20
> > -hadriel
> > p.s. I like the "Suspicious Line" thing - all RFC lines are
> > suspicious! ;)
> >
> >
> > On Apr 2, 2012, at 2:12 PM, Paul Kyzivat wrote:
> >
> >> [reposting with a subject line]
> >>
> >> On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
> >>> Bruno Chatras has pointed out that a number of examples in various
> >>> SIP-related RFCs show Content-Length header fields in the headers
> of
> >>> body-parts of multipart entities.  MIME does not define the
> meaining
> >>> of such header fields, as the MIME framing of body parts is done =
by
> the boundary markers.
> >>>
> >>> As Dale Moberg has pointed out, these header fields might not be
> >>> truly incorrect but merely redundant or "to be ignored".
> >>>
> >>> But given that MIME multipart already frames the body-parts,
> >>> Content-Length headers on body-parts are never necessary.
> >>>
> >>> So it seems to me that we should decide on a "best practices" for
> >>> the use of Content-Length in the headers of a body part, to =
clarify
> >>> the ambiguities in the specifications, and to state whether it's
> >>> preferable to include such a Content-Length or not.
> >>>
> >>> I've appended a list of all the examples carrying such
> >>> Content-Length headers that I could find by automated means.
> >>>
> >>> Dale
> >>> =
-------------------------------------------------------------------
> -
> >>> ------------------------------------
> >>> RFC 2848
> >>>
> >>>                         The PINT Service Protocol:
> >>>     Extensions to SIP and SDP for IP Access to Telephone Call
> >>> Services
> >>>
> >>> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
> >>> Suspicious line at rfc2848.txt:2209:       Content-Length:50
> >>> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
> >>> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
> >>> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
> >>> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
> >>>
> >>> Several of these body-part headers are not separated from the
> >>> body-part contents by CR-LF-CR-LF.
> >>>
> >>> RFC 3261
> >>>
> >>>                      SIP: Session Initiation Protocol
> >>>
> >>> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
> >>>
> >>> RFC 3892
> >>>
> >>>        The Session Initiation Protocol (SIP) Referred-By Mechanism
> >>>
> >>> Suspicious line at rfc3892.txt:642:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:687:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:702:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:879:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:887:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:938:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:964:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:972:       Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:1056:      Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:1106:      Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:1168:      Content-Length:
> (appropriate value)
> >>> Suspicious line at rfc3892.txt:1196:      Content-Length:
> (appropriate value)
> >>>
> >>> RFC 3893
> >>>
> >>>                     Session Initiation Protocol (SIP)
> >>>                  Authenticated Identity Body (AIB) Format
> >>>
> >>> Suspicious line at rfc3893.txt:249:    Content-Length: 147
> >>> (This has an extra CR-LF before the body-part headers.)
> >>> Suspicious line at rfc3893.txt:263:    Content-Length: 608
> >>>
> >>> RFC 4463
> >>>
> >>>                  A Media Resource Control Protocol (MRCP)
> >>>                Developed by Cisco, Nuance, and Speechworks
> >>>
> >>> Suspicious line at rfc4463.txt:1705:        Content-Length:176
> >>> Suspicious line at rfc4463.txt:1714:        Content-Length:104
> >>> Suspicious line at rfc4463.txt:2977:       Content-Length:176
> >>> (Also missing the CR-LF after the body part headers.)
> >>> Suspicious line at rfc4463.txt:2985:       Content-Length:104
> >>>
> >>> RFC 5547
> >>>
> >>>        A Session Description Protocol (SDP) Offer/Answer Mechanism
> >>>                          to Enable File Transfer
> >>>
> >>> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of
> SDP]
> >>> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of
> image]
> >>> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of
> SDP]
> >>> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of
> image]
> >>>
> >>> RFC 5589
> >>>
> >>>         Session Initiation Protocol (SIP) Call Control - Transfer
> >>>
> >>> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
> >>> (An extra CR-LF in the middle of the body part headers.)
> >>> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
> >>> (An extra CR-LF in the middle of the body part headers.)
> >>> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
> >>>
> >>> RFC 6086
> >>>
> >>>    Session Initiation Protocol (SIP) INFO Method and Package
> >>> Framework
> >>>
> >>> Suspicious line at rfc6086.txt:1594:    Content-length: 59
> >>> Suspicious line at rfc6086.txt:1620:    Content-length: 59
> >>> Suspicious line at rfc6086.txt:1637:    Content-length: 59
> >>> Suspicious line at rfc6086.txt:1664:    Content-length: 59
> >>> _______________________________________________
> >>> sipcore mailing list
> >>> sipcore@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/sipcore
> >>>
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> >
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From pkyzivat@alum.mit.edu  Tue Apr  3 06:52:48 2012
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Date: Tue, 03 Apr 2012 09:52:45 -0400
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Subject: Re: [sipcore] Content-Length in multipart body parts
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On 4/3/12 5:09 AM, bruno.chatras@orange.com wrote:
> I assume you mean a set of errata to fix the "Suspicious" examples. If Yes, I do agree as well.

Yes, that is what I meant.

	Thanks,
	Paul

>> -----Message d'origine-----
>> De : sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] De la
>> part de Christer Holmberg
>> Envoyé : mardi 3 avril 2012 08:40
>> Ą : Paul Kyzivat; Hadriel Kaplan
>> Cc : Dale R (Dale) Worley;<sipcore@ietf.org>
>> Objet : Re: [sipcore] Content-Length in multipart body parts
>>
>> Hi,
>>
>>>> Ugh.  Yeah, that's not good for examples to show.
>>>> So what's the solution?  Errata?
>>>
>>> Yes, I think the errata process is the right one for this.
>>
>> Agree, eventhough the C-L should cause no conflicts (as long as the
>> value is correct).
>>
>> And, even if a specific MIME contains another (nested) multipart entry,
>> C-L should not be needed, as the MIME borders are still present.
>>
>> Regards,
>>
>> Christer
>>
>>
>>> -hadriel
>>> p.s. I like the "Suspicious Line" thing - all RFC lines are
>>> suspicious! ;)
>>>
>>>
>>> On Apr 2, 2012, at 2:12 PM, Paul Kyzivat wrote:
>>>
>>>> [reposting with a subject line]
>>>>
>>>> On 4/2/12 6:53 PM, Worley, Dale R (Dale) wrote:
>>>>> Bruno Chatras has pointed out that a number of examples in various
>>>>> SIP-related RFCs show Content-Length header fields in the headers
>> of
>>>>> body-parts of multipart entities.  MIME does not define the
>> meaining
>>>>> of such header fields, as the MIME framing of body parts is done by
>> the boundary markers.
>>>>>
>>>>> As Dale Moberg has pointed out, these header fields might not be
>>>>> truly incorrect but merely redundant or "to be ignored".
>>>>>
>>>>> But given that MIME multipart already frames the body-parts,
>>>>> Content-Length headers on body-parts are never necessary.
>>>>>
>>>>> So it seems to me that we should decide on a "best practices" for
>>>>> the use of Content-Length in the headers of a body part, to clarify
>>>>> the ambiguities in the specifications, and to state whether it's
>>>>> preferable to include such a Content-Length or not.
>>>>>
>>>>> I've appended a list of all the examples carrying such
>>>>> Content-Length headers that I could find by automated means.
>>>>>
>>>>> Dale
>>>>> -------------------------------------------------------------------
>> -
>>>>> ------------------------------------
>>>>> RFC 2848
>>>>>
>>>>>                          The PINT Service Protocol:
>>>>>      Extensions to SIP and SDP for IP Access to Telephone Call
>>>>> Services
>>>>>
>>>>> Suspicious line at rfc2848.txt:2195:       Content-Length: 236
>>>>> Suspicious line at rfc2848.txt:2209:       Content-Length:50
>>>>> Suspicious line at rfc2848.txt:2268:       Content-Length: 316
>>>>> Suspicious line at rfc2848.txt:2284:       Content-Length: 172
>>>>> Suspicious line at rfc2848.txt:2372:    Content-Length: 325
>>>>> Suspicious line at rfc2848.txt:2387:    Content-Length: 352
>>>>>
>>>>> Several of these body-part headers are not separated from the
>>>>> body-part contents by CR-LF-CR-LF.
>>>>>
>>>>> RFC 3261
>>>>>
>>>>>                       SIP: Session Initiation Protocol
>>>>>
>>>>> Suspicious line at rfc3261.txt:11846:         Content-Length: 231
>>>>>
>>>>> RFC 3892
>>>>>
>>>>>         The Session Initiation Protocol (SIP) Referred-By Mechanism
>>>>>
>>>>> Suspicious line at rfc3892.txt:642:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:687:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:702:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:879:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:887:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:938:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:964:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:972:       Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:1056:      Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:1106:      Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:1168:      Content-Length:
>> (appropriate value)
>>>>> Suspicious line at rfc3892.txt:1196:      Content-Length:
>> (appropriate value)
>>>>>
>>>>> RFC 3893
>>>>>
>>>>>                      Session Initiation Protocol (SIP)
>>>>>                   Authenticated Identity Body (AIB) Format
>>>>>
>>>>> Suspicious line at rfc3893.txt:249:    Content-Length: 147
>>>>> (This has an extra CR-LF before the body-part headers.)
>>>>> Suspicious line at rfc3893.txt:263:    Content-Length: 608
>>>>>
>>>>> RFC 4463
>>>>>
>>>>>                   A Media Resource Control Protocol (MRCP)
>>>>>                 Developed by Cisco, Nuance, and Speechworks
>>>>>
>>>>> Suspicious line at rfc4463.txt:1705:        Content-Length:176
>>>>> Suspicious line at rfc4463.txt:1714:        Content-Length:104
>>>>> Suspicious line at rfc4463.txt:2977:       Content-Length:176
>>>>> (Also missing the CR-LF after the body part headers.)
>>>>> Suspicious line at rfc4463.txt:2985:       Content-Length:104
>>>>>
>>>>> RFC 5547
>>>>>
>>>>>         A Session Description Protocol (SDP) Offer/Answer Mechanism
>>>>>                           to Enable File Transfer
>>>>>
>>>>> Suspicious line at rfc5547.txt:1645:    Content-Length: [length of
>> SDP]
>>>>> Suspicious line at rfc5547.txt:1670:    Content-Length: [length of
>> image]
>>>>> Suspicious line at rfc5547.txt:2093:    Content-Length: [length of
>> SDP]
>>>>> Suspicious line at rfc5547.txt:2118:    Content-Length: [length of
>> image]
>>>>>
>>>>> RFC 5589
>>>>>
>>>>>          Session Initiation Protocol (SIP) Call Control - Transfer
>>>>>
>>>>> Suspicious line at rfc5589.txt:2609:    Content-Length: 2961
>>>>> (An extra CR-LF in the middle of the body part headers.)
>>>>> Suspicious line at rfc5589.txt:2677:    Content-Length: 156
>>>>> (An extra CR-LF in the middle of the body part headers.)
>>>>> Suspicious line at rfc5589.txt:2696:    Content-Length: 2961
>>>>>
>>>>> RFC 6086
>>>>>
>>>>>     Session Initiation Protocol (SIP) INFO Method and Package
>>>>> Framework
>>>>>
>>>>> Suspicious line at rfc6086.txt:1594:    Content-length: 59
>>>>> Suspicious line at rfc6086.txt:1620:    Content-length: 59
>>>>> Suspicious line at rfc6086.txt:1637:    Content-length: 59
>>>>> Suspicious line at rfc6086.txt:1664:    Content-length: 59
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>>
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>> _______________________________________________
>> sipcore mailing list
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>> https://www.ietf.org/mailman/listinfo/sipcore
>


From kpfleming@digium.com  Tue Apr  3 14:47:04 2012
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Subject: Re: [sipcore] Editorial (minor) comment on RFC 5627
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On 03/31/2012 12:58 PM, Nataraju A.B wrote:
> Hi All,
>
>
> I feel following is an editorial comment on the RFC 5627...
>
>
>
> 4.4.  Using One's Own GRUUs
>
>
>      ....A UA compliant to this specification SHOULD use one of its GRUUs as its remote target. This includes..
>
>
>     . . . .
>
>     o  a 2xx response to NOTIFY
>     o  the UPDATE request
>     o  a 2xx response to *NOTIFY*
>
>
> The last point must have been mentioned as
>
>     o  a 2xx response to **UPDATE**

Please check to see whether an errata has been filed already, and if 
not, you should file one yourself so this will be tracked for a future 
revision of the document.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From nataraju.sip@gmail.com  Tue Apr  3 18:04:02 2012
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I have filed the bug against rfc 5627

*Errata ID: 3173
*

Thanks,
Nataraju A.B.

On Wed, Apr 4, 2012 at 3:17 AM, Kevin P. Fleming <kpfleming@digium.com>wrote:

> On 03/31/2012 12:58 PM, Nataraju A.B wrote:
>
>> Hi All,
>>
>>
>> I feel following is an editorial comment on the RFC 5627...
>>
>>
>>
>> 4.4.  Using One's Own GRUUs
>>
>>
>>     ....A UA compliant to this specification SHOULD use one of its GRUUs
>> as its remote target. This includes..
>>
>>
>>    . . . .
>>
>>    o  a 2xx response to NOTIFY
>>    o  the UPDATE request
>>    o  a 2xx response to *NOTIFY*
>>
>>
>> The last point must have been mentioned as
>>
>>    o  a 2xx response to **UPDATE**
>>
>
> Please check to see whether an errata has been filed already, and if not,
> you should file one yourself so this will be tracked for a future revision
> of the document.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> ______________________________**_________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/**listinfo/sipcore<https://www.ietf.org/mailman/listinfo/sipcore>
>



-- 
Thanks,
Nataraju A.B.

--e89a8f22bd899e90ca04bccfff5b
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<div style=3D"font-family:arial,sans-serif;font-size:14px;background-color:=
rgb(255,255,255)">I have filed the bug against rfc 5627</div><div style=3D"=
font-family:arial,sans-serif;font-size:14px;background-color:rgb(255,255,25=
5)">
<br></div><div style=3D"font-family:arial,sans-serif;font-size:14px;backgro=
und-color:rgb(255,255,255)"><b><span style=3D"font-family:&#39;Times New Ro=
man&#39;;font-size:18px">Errata ID: 3173</span>=A0<br></b></div><div><br></=
div>
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace">Nataraju A.B.</font></font></div><br class=3D"Apple-int=
erchange-newline">
<div class=3D"gmail_quote">On Wed, Apr 4, 2012 at 3:17 AM, Kevin P. Fleming=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:kpfleming@digium.com">kpfleming@di=
gium.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><div></div><div class=3D"h5">On 03/31/2012 12:58 PM, Nataraju A.B wrot=
e:<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class=3D"h5=
">
Hi All,<br>
<br>
<br>
I feel following is an editorial comment on the RFC 5627...<br>
<br>
<br>
<br>
4.4. =A0Using One&#39;s Own GRUUs<br>
<br>
<br>
 =A0 =A0 ....A UA compliant to this specification SHOULD use one of its GRU=
Us as its remote target. This includes..<br>
<br>
<br>
 =A0 =A0. . . .<br>
<br>
 =A0 =A0o =A0a 2xx response to NOTIFY<br>
 =A0 =A0o =A0the UPDATE request<br>
 =A0 =A0o =A0a 2xx response to *NOTIFY*<br>
<br>
<br>
The last point must have been mentioned as<br>
<br></div></div>
 =A0 =A0o =A0a 2xx response to **UPDATE**<br>
</blockquote>
<br>
Please check to see whether an errata has been filed already, and if not, y=
ou should file one yourself so this will be tracked for a future revision o=
f the document.<br><font color=3D"#888888">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href=3D"mailto:kfleming@digium.com" target=3D"_blank">kfleming@d=
igium.com</a> | SIP: <a href=3D"mailto:kpfleming@digium.com" target=3D"_bla=
nk">kpfleming@digium.com</a> | Skype: kpfleming<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at <a href=3D"http://www.digium.com" target=3D"_blank">www.dig=
ium.com</a> &amp; <a href=3D"http://www.asterisk.org" target=3D"_blank">www=
.asterisk.org</a></font><div><div></div><div class=3D"h5"><br>
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace">Nataraju A.B.</font></font></div>
<br>

--e89a8f22bd899e90ca04bccfff5b--

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Subject: Re: [sipcore] Editorial (minor) comment on RFC 5627
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On 4/3/12 9:03 PM, Nataraju A.B wrote:
> I have filed the bug against rfc 5627
>
> *Errata ID: 3173
> *

Thanks!

	Paul

> Thanks,
> Nataraju A.B.
>
> On Wed, Apr 4, 2012 at 3:17 AM, Kevin P. Fleming <kpfleming@digium.com
> <mailto:kpfleming@digium.com>> wrote:
>
>     On 03/31/2012 12:58 PM, Nataraju A.B wrote:
>
>         Hi All,
>
>
>         I feel following is an editorial comment on the RFC 5627...
>
>
>
>         4.4.  Using One's Own GRUUs
>
>
>              ....A UA compliant to this specification SHOULD use one of
>         its GRUUs as its remote target. This includes..
>
>
>             . . . .
>
>             o  a 2xx response to NOTIFY
>             o  the UPDATE request
>             o  a 2xx response to *NOTIFY*
>
>
>         The last point must have been mentioned as
>
>             o  a 2xx response to **UPDATE**
>
>
>     Please check to see whether an errata has been filed already, and if
>     not, you should file one yourself so this will be tracked for a
>     future revision of the document.
>
>     --
>     Kevin P. Fleming
>     Digium, Inc. | Director of Software Technologies
>     Jabber: kfleming@digium.com <mailto:kfleming@digium.com> | SIP:
>     kpfleming@digium.com <mailto:kpfleming@digium.com> | Skype: kpfleming
>     445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>     Check us out at www.digium.com <http://www.digium.com> &
>     www.asterisk.org <http://www.asterisk.org>
>
>     _________________________________________________
>     sipcore mailing list
>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/sipcore
>     <https://www.ietf.org/mailman/listinfo/sipcore>
>
>
>
>
> --
> Thanks,
> Nataraju A.B.
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nataraju.sip@gmail.com  Thu Apr  5 11:42:43 2012
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Subject: [sipcore] clarification needed : GRUU - Contact mapping - RFC 5627 (GRUU)
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--f46d04083b0f8e120704bcf2e76f
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Hi All,

AFAIU, the GRUU is always associated with a specific UA instance (only
one). But the following text from RFC5627 seems to convey single GRUU
can be mapped on to more than one contact/UA instance.


Following is the excerpt from RFC5627.


<5627>

3.4.  Dereferencing a GRUU

   Because a GRUU is simply a URI, a UA dereferences it in exactly the
   same way as it would any other URI.  However, once the request has
   been routed to the appropriate proxy, the behavior is slightly
   different.  The proxy will map the GRUU to the AOR and determine the
   set of contacts that the particular UA instance has registered.  The
   GRUU is then mapped to **those contacts**, and the request is routed
   towards the UA.

</5627>

I feel the text must rephrased as follows

<>
   Because a GRUU is simply a URI, a UA dereferences it in exactly the
   same way as it would any other URI.  However, once the request has
   been routed to the appropriate proxy, the behavior is slightly
   different.  The proxy will map the GRUU to the AOR and determine the
   set of contacts that are registered with the AOR. Then
   GRUU is mapped to specific contact based on matching instance id
   (+sip.instance), and then request is routed towards the specific UA.
</>

I appreciate your opinion on the same issue.

Thanks,
Nataraju A.B.

--f46d04083b0f8e120704bcf2e76f
Content-Type: text/html; charset=ISO-8859-1
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<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" size=3D"1">Hi All,</font></pre><pre style=3D"=
word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39;courier new&#=
39;, monospace" size=3D"1">
</font></pre><pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font=
 size=3D"1" face=3D"&#39;courier new&#39;, monospace">AFAIU, the GRUU is al=
ways associated with a specific UA instance (only one). But the following t=
ext from=A0RFC5627 seems to convey single GRUU can be mapped on to more tha=
n one contact/UA instance.=A0</font></pre>
<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" size=3D"1"><br></font></pre><pre style=3D"wor=
d-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39;courier new&#39;=
, monospace" size=3D"1">Following is the excerpt from RFC5627.</font></pre>

<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" size=3D"1"><br></font></pre><pre style=3D"wor=
d-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39;courier new&#39;=
, monospace" size=3D"1">&lt;5627&gt;</font></pre>

<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" size=3D"1">3.4.  Dereferencing a GRUU</font><=
/pre><pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D=
"&#39;courier new&#39;, monospace" size=3D"1">   Because a GRUU is simply a=
 URI, a UA dereferences it in exactly the
   same way as it would any other URI.  However, once the request has
   been routed to the appropriate proxy, the behavior is slightly
   different.  <font color=3D"#cc0000">The proxy will map the GRUU to the A=
OR and determine the
   set of contacts that the particular UA instance has registered.  The
   GRUU is then mapped to **those contacts**, and the request is routed
   towards the UA.
</font></font></pre><pre style=3D"word-wrap:break-word;white-space:pre-wrap=
"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">&lt;/5627&gt;<=
/font></pre><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"1"=
>I feel the text must rephrased as follows</font></div>
<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"1"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">&lt;&=
gt;</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=
=3D"1"><div>
=A0 =A0Because a GRUU is simply a URI, a UA dereferences it in exactly the<=
/div><div>=A0 =A0same way as it would any other URI. =A0However, once the r=
equest has</div><div>=A0 =A0been routed to the appropriate proxy, the behav=
ior is slightly</div>
<div>=A0 =A0different. =A0<font color=3D"#3333ff">The proxy will map the GR=
UU to the AOR and determine the</font></div><div><font color=3D"#3333ff">=
=A0 =A0set of contacts that are registered with the AOR. Then=A0</font></di=
v><div><font color=3D"#3333ff">=A0 =A0GRUU is mapped to specific contact ba=
sed on matching instance id=A0</font></div>
<div><font color=3D"#3333ff">=A0 =A0(<span style=3D"white-space:pre-wrap">+=
sip.instance)</span>, and then request is routed=A0towards the specific UA.=
</font></div><div>&lt;/&gt;</div><div><br></div></font></div>
<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">I appreciat=
e your opinion on the same issue.</font></div><div><font face=3D"&#39;couri=
er new&#39;, monospace" size=3D"1"><br></font></div><font color=3D"#000099"=
><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Thanks,</font><=
/font><div>

<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Nataraju A.B.</font></font></div>

--f46d04083b0f8e120704bcf2e76f--

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Subject: Re: [sipcore] clarification needed : GRUU - Contact mapping - RFC 5627 (GRUU)
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On 4/5/12 2:42 PM, Nataraju A.B wrote:
> Hi All,
>
>
> AFAIU, the GRUU is always associated with a specific UA instance (only one). But the following text from RFC5627 seems to convey single GRUU can be mapped on to more than one contact/UA instance.

The text below covers that case where a particular UA is fault tolerant 
and replicated or has multiple interfaces.  (Such as spelled out in 
outbound - RFC5626.

	Thanks,
	Paul

> Following is the excerpt from RFC5627.
>
>
> <5627>
>
> 3.4.  Dereferencing a GRUU
>
>     Because a GRUU is simply a URI, a UA dereferences it in exactly the
>     same way as it would any other URI.  However, once the request has
>     been routed to the appropriate proxy, the behavior is slightly
>     different.The proxy will map the GRUU to the AOR and determine the
>     set of contacts that the particular UA instance has registered.  The
>     GRUU is then mapped to **those contacts**, and the request is routed
>     towards the UA.
>
> </5627>
>
> I feel the text must rephrased as follows
>
> <>
>     Because a GRUU is simply a URI, a UA dereferences it in exactly the
>     same way as it would any other URI.  However, once the request has
>     been routed to the appropriate proxy, the behavior is slightly
>     different. The proxy will map the GRUU to the AOR and determine the
>     set of contacts that are registered with the AOR. Then
>     GRUU is mapped to specific contact based on matching instance id
>     (+sip.instance), and then request is routed towards the specific UA.
> </>
>
> I appreciate your opinion on the same issue.
>
> Thanks,
> Nataraju A.B.
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From adam@nostrum.com  Wed Apr 11 07:38:34 2012
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Date: Wed, 11 Apr 2012 09:38:32 -0500
From: Adam Roach - SIPCORE Chair <adam@nostrum.com>
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Subject: [sipcore] New Milestone: WebSockets transport
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[as chair]

Based on feedback at the meeting in Paris and a lack of on-list 
objection, I'm declaring working group consensus for adding a milestone 
to the SIPCORE charter for the definition of a new WebSockets transport 
for SIP. The new milestone I am requesting is:


Apr 2013  WebSockets transport definition for SIP to the IESG (proposed 
standard)


In terms of document adoption: we did not call for any specific document 
to be adopted in Paris, and response to the call for consensus on the 
email list has been tepid (only six people have responded to that call: 
one requested holding off adoption until changes have been made to 
reflect the discussion to date; an author spoke in favor of adoption; 
and the remaining four participants voiced neither support or objection).

Absent an indication of consensus to adopt the document, I request that 
the authors submit a new version reflecting the discussions we have had 
on the email list and in Paris regarding the outbound and GRUU 
mechanisms. I will then issue a second formal call for adoption of the 
document to satisfy the milestone.

Thank you.

/a

From internet-drafts@ietf.org  Wed Apr 11 15:09:36 2012
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc3265bis-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : SIP-Specific Event Notification
	Author(s)       : Adam Roach
	Filename        : draft-ietf-sipcore-rfc3265bis-08.txt
	Pages           : 52
	Date            : 2012-04-11

   This document describes an extension to the Session Initiation
   Protocol (SIP).  The purpose of this extension is to provide an
   extensible framework by which SIP nodes can request notification from
   remote nodes indicating that certain events have occurred.

   Note that the event notification mechanisms defined herein are NOT
   intended to be a general-purpose infrastructure for all classes of
   event subscription and notification.

   This document represents a backwards-compatible improvement on the
   original mechanism described by RFC 3265, taking into account several
   years of implementation experience.  Accordingly, this document
   obsoletes RFC 3265.  This document also updates RFC 4660 slightly to
   accommodate some small changes to the mechanism that were discussed
   in that document.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-08.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-08.txt


From adam@nostrum.com  Thu Apr 12 10:19:46 2012
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Date: Thu, 12 Apr 2012 12:19:40 -0500
From: Adam Roach <adam@nostrum.com>
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Subject: [sipcore] RFC3265bis: reason code registration policy
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[as document author]

As you probably know, RFC3265bis has been handed off to the IESG for 
pre-publication evaluation.

Section 7.2 of the RFC3265bis document discusses the new IANA registry 
for the "reason" code. The gen-art review of this document turned up the 
following discrepancy:

  * 'Following the policies outlined in "Guidelines for Writing an IANA
    Considerations Section in RFCs" [RFC5226], new reason codes require
    a Standards Action.'

  * ' Insertion of such values takes place as part of the RFC
    publication process or as the result of inter-SDO liaison activity.'


The problem is that the "Standards Action" requirement is defined: 
"Values are assigned only for Standards Track RFCs approved by the 
IESG," which implies that inter-SDO liaising isn't sufficient for adding 
values.

I can't find any relic of this issue being discussed on the mailing 
list, and have no specific recollection how we arrived at the current 
text. Keep in mind that the need to expand these reason codes should be 
very slight (we've only had one in nine years since 3265 was published), 
and will almost certainly be paired with protocol changes that RFC 5727 
would require an RFC for anyway.

Based on those two factors (and especially the second one), I think what 
we probably want to do is remove the phrase "of as the result of 
inter-SDO liaison activity," and leave the policy "Standards Action" as 
currently stated.

Please raise any objections to this proposed course of action on the 
SIPCORE mailing list. I will make sure the results of any on-list 
discussion are brought to the attention of the IESG.

/a


From jmillan@aliax.net  Thu Apr 12 13:44:36 2012
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Subject: Re: [sipcore] New Milestone: WebSockets transport
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El d=EDa 11 de abril de 2012 16:38, Adam Roach - SIPCORE Chair
<adam@nostrum.com> escribi=F3:
> [as chair]
>
> Based on feedback at the meeting in Paris and a lack of on-list objection=
,
> I'm declaring working group consensus for adding a milestone to the SIPCO=
RE
> charter for the definition of a new WebSockets transport for SIP. The new
> milestone I am requesting is:
>
>
> Apr 2013 =A0WebSockets transport definition for SIP to the IESG (proposed
> standard)
>
>
> In terms of document adoption: we did not call for any specific document =
to
> be adopted in Paris, and response to the call for consensus on the email
> list has been tepid (only six people have responded to that call: one
> requested holding off adoption until changes have been made to reflect th=
e
> discussion to date; an author spoke in favor of adoption; and the remaini=
ng
> four participants voiced neither support or objection).
>
> Absent an indication of consensus to adopt the document, I request that t=
he
> authors submit a new version reflecting the discussions we have had on th=
e
> email list and in Paris regarding the outbound and GRUU mechanisms. I wil=
l
> then issue a second formal call for adoption of the document to satisfy t=
he
> milestone.
>

A new version of the document addressing the discussions we have had
will be submitted in the next few days.

Thank you.

> Thank you.
>
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From nataraju.sip@gmail.com  Fri Apr 13 10:23:07 2012
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: Adam Roach <adam@nostrum.com>, sipcore@ietf.org
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Subject: [sipcore] draft-ietf-sipcore-rfc3265bis-08.txt - Review comments (Nataraju A B)
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Adam,

Please find some editorial comments on the draft
"draft draft-ietf-sipcore-rfc3265bis-08.txt"

For simplicity, I have mentioned both the current text and proposed text.

The comments / corrective text can be found with prefix and suffix  --- **

-----------------------------------------------------
1.1.  Overview of Operation
CurrentText:
   The general concept is that entities in the network can subscribe to
   resource or call state for various resources or calls in the network,
   and those entities (or entities acting on their behalf) can send
   notifications when those states change.

ReplacementText:
   The general concept is that entities in the network can subscribe to
   resource or call state **at / of** various resources or calls in the
network,
   and those entities (or entities acting on their behalf) **who** can send
   notifications when those states change.
-----------------------------------------------------

1.1.  Overview of Operation

CurrentText:
   Subscriptions are expired and must be refreshed by subsequent
   SUBSCRIBE requests.

ReplacementText:
   Subscriptions are expired and must be refreshed by subsequent
   SUBSCRIBE requests **before expiration**.
-----------------------------------------------------

2.  Definitions

CurrentText:
   Subscriber:  A subscriber is a user agent which receives NOTIFY
      requests from notifiers; these NOTIFY requests contain information
      about the state of a resource in which the subscriber is
      interested.  Subscribers typically also generate SUBSCRIBE
      requests and send them to notifiers to create subscriptions.

Comment:
After this information, we can add a normative text which indicates
 when a SIP entity might get NOTIFY messages without actually
 sending SUBSCRIBE request. For example, REFER request might end up with
implicit subscription. Also mention any other requests which may lead to
 implicit subscriptions (other than REFER).
-----------------------------------------------------
2.  Definitions

CurrentText:
   Subscription:  A subscription is a set of application state
      associated with a dialog.  This application state includes a
      pointer to the associated dialog, the event package name, and
      possibly an identification token.  Event packages will define
      additional subscription state information.  By definition,
      subscriptions exist in both a subscriber and a notifier.

ReplacementText:
   Subscription:  A subscription is a set of application **states**
      associated with a dialog.  **The** application state includes a
      pointer to the associated dialog, the event package name, and
      possibly an identification token.  Event packages will define
      additional subscription state information.  By definition,
      subscriptions exist in both a subscriber and a notifier.
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:

   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions on a periodic basis using a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].

ReplacementText
3.1.1.  Subscription Duration

   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions **before expiration** on a periodic basis using
a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:
      In addition to being a request to unsubscribe, a SUBSCRIBE request
      with "Expires" of 0 also causes a fetch of state; see
      Section 4.4.3.

Comment:
     The above text has been mentioned to be normative text. But I feel
this
     shall be written as a regualr text in the draft/rfc. because this is
the
     expected behavior from the notifier.
-----------------------------------------------------
3.1.2.  Identification of Subscribed Events and Event Classes

CurrentText:
   Subscribers MUST include exactly one "Event" header field in
   SUBSCRIBE requests, indicating to which event or class of events they
   are subscribing.  The "Event" header field will contain a token which
   indicates the type of state for which a subscription is being
   requested.  This token will be registered with the IANA and will
   correspond to an event package which further describes the semantics
   of the event or event class.


ReplacementText:
   Subscribers MUST include exactly one "Event" header field in
   SUBSCRIBE requests, indicating to which event or class of events they
   are subscribing.  The "Event" header field **MUST** contain a token which
   indicates the type of state for which a subscription is being
   requested.  This token will be registered with the IANA and will
   correspond to an event package which further describes the semantics
   of the event or event class.
-----------------------------------------------------
3.1.  SUBSCRIBE

CurrentText:
   The SUBSCRIBE method is used to request current state and state
   updates from a remote node.  SUBSCRIBE requests are target refresh


ReplacementText:
   The SUBSCRIBE method is used to request current state and **subsequent**
state
   updates from a remote node.  SUBSCRIBE requests are target refresh


-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:

   SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
   in [RFC3261]).  This expires value indicates the duration of the
   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions on a periodic basis using a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].

Comment:
We can add a normative text saying, the refresh time will be
 normally 50% to 75% of subscription duration.

-----------------------------------------------------
3.2.1.

CurrentText:
   This body will contain either the state of the subscribed resource or
   a pointer to such state in the form of a URI (see Section 5.4.13).

ReplacementText/Comment:
   This body **SHALL** contain either the state of the subscribed resource
or
   a pointer to such state in the form of a **URL** (see Section 5.4.13).
-----------------------------------------------------
4.1.2.  Creating and Maintaining Subscriptions


CurrentText:

   From the subscriber's perspective, a subscription proceeds according
   to the following state diagram:

Comment:
Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
 received in notify_wait state (because very often this is possible).
 If no change in state of FSM, then some normative text shall convey the
same.
        It may be no-operation, but better to mention the same.
-----------------------------------------------------
4.1.2.1.  Requesting a Subscription

CurrentText:
   When a subscriber wishes to subscribe to a particular state for a
   resource, it forms a SUBSCRIBE request.  If the initial SUBSCRIBE

ReplacementText:
   When a subscriber wishes to subscribe to a particular state **of** a
   resource, it forms a SUBSCRIBE request **according to sec 3.1 and send
it across to notifier**.  If the initial SUBSCRIBE

-----------------------------------------------------
4.1.2.1.  Requesting a Subscription

CurrentText:
   [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
   an "Accept" header field, but that field does not indicate a media
   type that the notifier is capable of generating in its NOTIFY
   requests, then the proper error response is 406 (Not Acceptable).

COMMENT:
It is better to mention it as a normative text (indented).

-----------------------------------------------------
4.1.3.  Receiving and Processing State Information

CurrentText:

   If the "Subscription-State" value is "terminated", the subscriber
   MUST consider the subscription terminated.  The "expires" parameter
   has no semantics for "terminated" -- notifiers SHOULD NOT include an

ReplacementText:

   If the "Subscription-State" value is "terminated", the subscriber
   MUST consider the subscription terminated.  The "expires" parameter
   has no semantics for "terminated" **state**. Notifiers SHOULD NOT
include an
-----------------------------------------------------

CurrentText:
   deactivated:  The subscription has been terminated, but the
      subscriber SHOULD retry immediately with a new subscription.  One

ReplacementText:
   deactivated:  The subscription has been terminated, but the
      subscriber SHOULD retry with a new subscription.  One

   Is It mandatory to resubscribe immediately. if not remove the word
"immediately".
   If it is delayed, what are the consequences ????

-----------------------------------------------------
4.2.1.  Subscription Establishment and Maintenance

CurrentText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty performing one
      action without the other.  Additionally, the proper exchange of

ReplacementText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty **in** performing one
      action without the other.  Additionally, the proper exchange of

-----------------------------------------------------
4.2.1.4.  Refreshing of Subscriptions

CurrentText:
   If no refresh for a notification address is received before its
   expiration time, the subscription is removed.  When removing a

ReplacementText:
   If no refresh for a notification is received before its
   expiration time, the subscription is removed.  When removing a
 Comment: Word "address" needs to be removed.
-----------------------------------------------------
4.3.  Proxy Behavior

CurrentText:
   Proxies that did not add a Record-Route header field to the initial
   SUBSCRIBE request MUST NOT add a Record-Route header field to any of
   the associated NOTIFY requests.

ReplacementText:
   Proxies that did not add a Record-Route header field to the initial
   SUBSCRIBE request MUST NOT add a Record-Route header field to any of
   the associated NOTIFY requests **or any SUBSCRIBE refresh requests**.
-----------------------------------------------------
4.4.1.  Dialog Creation and Termination

CurrentText:

      A subscriber may send a SUBSCRIBE request with an "Expires" header
      field of 0 in order to trigger the sending of such a NOTIFY
      request; however, for the purposes of subscription and dialog
      lifetime, the subscription is not considered terminated until the
      NOTIFY transaction with a "Subscription-State" of "terminated"
      completes.

Comment:
We may have to mention how long the dialog must be kept alive, before
 being cleaned up. Do we delete the dialog immediately after 200-SUB
or wait till NOTIFY with "Subscription-State"="terminated". If we always
 wait till Last Notify for dialog termination, then 200-SUB don't have any
impact on the dialog termination. Atleast some normative text is of
 necessity, for better understanding of the topic.
-----------------------------------------------------
4.4.3.  Polling Resource State

CurrentText:

   Note that the NOTIFY requests triggered by SUBSCRIBE requests with
   "Expires" header fields of 0 will contain a "Subscription-State"
   value of "terminated", and a "reason" parameter of "timeout".

ReplacementText:
   Note that the NOTIFY requests triggered by SUBSCRIBE requests with
   "Expires" header fields of 0 **shall** contain a "Subscription-State"
   value of "terminated", and a "reason" parameter of "timeout".
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
      for a dialog (e.g., route set handling).  In particular, multiple
      subscriptions within a dialog are expire independently, and
      require independent subscription refreshes.

ReplacementText:
      for a dialog (e.g., route set handling).  In particular, multiple
      subscriptions within a dialog **operates** independently, and
      require independent subscription refreshes.
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
   o  Subscribers MAY include an "id" parameter in SUBSCRIBE request
      "Event" header field to allow differentiation between multiple

ReplacementText:
   o  Subscribers MAY include an "id" parameter in SUBSCRIBE **request's**
      "Event" header field to allow differentiation between multiple
OR
   o  Subscribers MAY include an "id" parameter in **"Event" header field
      of SUBSCRIBE request** to allow differentiation between multiple
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
   o  When a subscriber refreshes a the subscription timer, the
      SUBSCRIBE request MUST contain the same "Event" header field "id"
      parameter as was present in the SUBSCRIBE request that created the
      subscription.  (Otherwise, the notifier will interpret the

ReplacementText:
   o  When a subscriber refreshes a the subscription timer, the
      SUBSCRIBE request MUST contain the same **"id" parameter in "Event"
      header field, which** was present in the SUBSCRIBE request that
created the
      subscription.  (Otherwise, the notifier will interpret the
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:

   o  When a subscription is created in the notifier, it stores any
      "Event" header field "id" parameter as part of the subscription
      information (along with the event package name).

ReplacementText:
   o  When a subscription is created in the notifier, it stores **the**
      **"id" parameter in "Event" header field** as part of the subscription
      information (along with the event package name).
-----------------------------------------------------
4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions

CurrentText:

   Neither SUBSCRIBE nor NOTIFY requests can be canceled.  If a UAS
   receives a CANCEL request that matches a known SUBSCRIBE or NOTIFY
   transaction, it MUST respond to the CANCEL request, but otherwise
   ignore it.  In particular, the CANCEL request MUST NOT affect
   processing of the SUBSCRIBE or NOTIFY request in any way.

ReplacementText/Comment:
What should be the response code for the CANCEL's reply ???
 Preferably 200-OK shall be used.
-----------------------------------------------------
4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions

CurrentText:

   UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY
   transactions.

ReplacementText:
   UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY
   transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY
   or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**
-----------------------------------------------------
5.3.2.  State Deltas

CurrentText:

   In the case that the state information to be conveyed is large, the
   event package may choose to detail a scheme by which NOTIFY requests
   contain state deltas instead of complete state.


ReplacementText:
   In the case that the state information to be conveyed is large, the
   event package may choose a scheme by which NOTIFY requests
   contain state deltas instead of complete state.

comment: deleted words "to detail" in 2nd line.

-----------------------------------------------------
5.3.2.  State Deltas

CurrentText:

   If a NOTIFY request arrives that has a version number that is
   incremented by more than one, the subscriber knows that a state delta
   has been missed; it ignores the NOTIFY request containing the state
   delta (except for the version number, which it retains to detect
   message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
   request containing a complete state snapshot.

Comment:
   From this text, it infers that if any state delta lost, then subscriber
   sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
request inturn lead
   to fetching the complete state information.

   It may be a good diea to differentiate the re-SUBSCRIBE triggered by
different scenarios.
   a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE duration.
In this
      case it is *not *necessary to fetch the complete state information
   b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec 5.3.2.
      VERSION number in received NOTIFY is increased by greater than ONE).
      In this case subscriber wants to fetch the complete state information.

   If we don't differentiate these two cases, then natural re-SUBSCRIBE
request
   unnecessarily fetch the complete state information, which may not be
necessary / expected.
   But uses higher network bandwidth to fetch the complete state
information.
-----------------------------------------------------
5.4.7.  Notifier generation of NOTIFY requests

CurrentText:
   This section may optionally describe the behavior used to process the
   subsequent response.

Comment:
   In this section 5.4.7, the above statement is not clear, what is really
meant to be.

-----------------------------------------------------
5.4.13.  Use of URIs to Retrieve State

CurrentText:

ReplacementText:
   In this context, I feel Use of "URLs" is more appropriate than "URIs".
-----------------------------------------------------


Thanks,
Nataraju A.B.
E-ID : nataraju.ab@gmail.com / nataraju.sip@gmail.com
+91-98455-95744

--f46d0430855a93ad1604bd92b957
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<font color=3D"#000099">Adam,</font><div><font color=3D"#000099"><br></font=
></div><div><font color=3D"#000099">Please find some editorial comments on =
the draft &quot;draft=A0draft-ietf-sipcore-rfc3265bis-08.txt&quot;<br clear=
=3D"all">
</font><div><font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">For simplicity, I have mentioned =
both the current text and proposed text.</font></div><div><font color=3D"#0=
00099"><br></font></div><div><font color=3D"#000099">The comments / correct=
ive text can be found with prefix and suffix =A0--- **</font></div>


<div><font color=3D"#000099"><br></font></div><div><div><font color=3D"#000=
099">-----------------------------------------------------</font></div><div=
><font color=3D"#000099">1.1. =A0Overview of Operation</font></div><div><fo=
nt color=3D"#000099">CurrentText:</font></div>

<div><font color=3D"#000099">=A0 =A0The general concept is that entities in=
 the network can subscribe to</font></div>
<div><font color=3D"#000099">=A0 =A0resource or call state for various reso=
urces or calls in the network,</font></div><div><font color=3D"#000099">=A0=
 =A0and those entities (or entities acting on their behalf) can send</font>=
</div><div>

<font color=3D"#000099">=A0 =A0notifications when those states change.</fon=
t></div><div><font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">ReplacementText:</font></div><div=
><font color=3D"#000099">=A0 =A0The general concept is that entities in the=
 network can subscribe to</font></div><div><font color=3D"#000099">=A0 =A0r=
esource or call state **at / of** various resources or calls in the network=
,</font></div>

<div><font color=3D"#000099">=A0 =A0and those entities (or entities acting =
on their behalf) **who** can send</font></div>
<div><font color=3D"#000099">=A0 =A0notifications when those states change.=
</font></div><div><font color=3D"#000099">---------------------------------=
--------------------</font></div><div><font color=3D"#000099"><br></font></=
div><div>

<font color=3D"#000099">1.1. =A0Overview of Operation</font></div><div><fon=
t color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentTe=
xt:</font></div><div><font color=3D"#000099">=A0 =A0Subscriptions are expir=
ed and must be refreshed by subsequent</font></div>


<div><font color=3D"#000099">=A0 =A0SUBSCRIBE requests.</font></div><div><f=
ont color=3D"#000099"><br></font></div><div><font color=3D"#000099">Replace=
mentText:</font></div><div><font color=3D"#000099">=A0 =A0Subscriptions are=
 expired and must be refreshed by subsequent</font></div>

<div><font color=3D"#000099">=A0 =A0SUBSCRIBE requests **before expiration*=
*.</font></div><div><font color=3D"#000099">-------------------------------=
----------------------</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
2. =A0Definitions</font></div><div><font color=3D"#000099"><br></font></div=
><div><font color=3D"#000099">CurrentText:</font></div><div><font color=3D"=
#000099">=A0 =A0Subscriber: =A0A subscriber is a user agent which receives =
NOTIFY</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 requests from notifiers; these NOT=
IFY requests contain information</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 about the state of a resource in w=
hich the subscriber is</font></div><div><font color=3D"#000099">=A0 =A0 =A0=
 interested. =A0Subscribers typically also generate SUBSCRIBE</font></div><=
div><font color=3D"#000099">=A0 =A0 =A0 requests and send them to notifiers=
 to create subscriptions.</font></div>


<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
Comment:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>After this information, we can add a normative text whi=
ch indicates=A0</font></div>

<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>w=
hen a SIP entity might get NOTIFY messages without actually=A0</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>s=
ending SUBSCRIBE request. For example, REFER request might end up with=A0</=
font></div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap=
">	</span>implicit subscription. Also mention any other requests which may =
lead to=A0</font></div>


<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>i=
mplicit subscriptions (other than REFER).</font></div><div><font color=3D"#=
000099">-----------------------------------------------------</font></div><=
div>

<font color=3D"#000099">2. =A0Definitions</font></div><div><font color=3D"#=
000099"><br></font></div><div><font color=3D"#000099">
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0Subscription: =
=A0A subscription is a set of application state</font></div><div><font colo=
r=3D"#000099">=A0 =A0 =A0 associated with a dialog. =A0This application sta=
te includes a</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 pointer to the associated dialog, =
the event package name, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 possibly an identification token. =
=A0Event packages will define</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 additional subscription state information. =A0By definition,</font>=
</div><div><font color=3D"#000099">=A0 =A0 =A0 subscriptions exist in both =
a subscriber and a notifier.</font></div>


<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0Subscripti=
on: =A0A subscription is a set of application **states**</font></div><div><=
font color=3D"#000099">=A0 =A0 =A0 associated with a dialog. =A0**The** app=
lication state includes a</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 pointer to the associated dialog, =
the event package name, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 possibly an identification token. =
=A0Event packages will define</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 additional subscription state information. =A0By definition,</font>=
</div><div><font color=3D"#000099">=A0 =A0 =A0 subscriptions exist in both =
a subscriber and a notifier.</font></div>


<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">3.1.1. =A0Subscription Dur=
ation</font></div><div><font color=3D"#000099"><br></font></div><div><font =
color=3D"#000099">CurrentText:</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0subscription. =A0In order to keep subscriptions effective beyond the=
</font></div>
<div><font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expi=
res&quot; header field, subscribers need</font></div><div><font color=3D"#0=
00099">=A0 =A0to refresh subscriptions on a periodic basis using a new SUBS=
CRIBE</font></div>

<div><font color=3D"#000099">=A0 =A0request on the same dialog as defined i=
n [RFC3261].</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText</font></div><div><font color=3D"#000099">3.1.1. =A0Subscrip=
tion Duration</font></div><div><font color=3D"#000099"><br></font></div><di=
v><font color=3D"#000099">=A0 =A0subscription. =A0In order to keep subscrip=
tions effective beyond the</font></div>

<div><font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expi=
res&quot; header field, subscribers need</font></div>
<div><font color=3D"#000099">=A0 =A0to refresh subscriptions **before expir=
ation** on a periodic basis using a new SUBSCRIBE</font></div><div><font co=
lor=3D"#000099">=A0 =A0request on the same dialog as defined in [RFC3261].<=
/font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">3.1.1. =A0Subscription Duration</font></div><d=
iv><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Cu=
rrentText:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 In addition=
 to being a request to unsubscribe, a SUBSCRIBE request</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 with &quot;Expires&quot; of 0 also=
 causes a fetch of state; see</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 Section 4.4.3.</font></div><div><f=
ont color=3D"#000099"><br></font></div><div><font color=3D"#000099">Comment=
:</font></div><div><font color=3D"#000099">=A0 =A0 =A0The above text has be=
en mentioned to be normative text. But I feel this=A0</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0shall be written as a regualr text =
in the draft/rfc. because this is the=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0expected behavior from the notifier=
.</font></div><div><font color=3D"#000099">--------------------------------=
---------------------</font></div><div><font color=3D"#000099">3.1.2. =A0Id=
entification of Subscribed Events and Event Classes</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div>
<div><font color=3D"#000099">=A0 =A0Subscribers MUST include exactly one &q=
uot;Event&quot; header field in</font></div><div><font color=3D"#000099">=
=A0 =A0SUBSCRIBE requests, indicating to which event or class of events the=
y</font></div>

<div><font color=3D"#000099">=A0 =A0are subscribing. =A0The &quot;Event&quo=
t; header field will contain a token which</font></div>
<div><font color=3D"#000099">=A0 =A0indicates the type of state for which a=
 subscription is being</font></div><div><font color=3D"#000099">=A0 =A0requ=
ested. =A0This token will be registered with the IANA and will</font></div>=
<div><font color=3D"#000099">=A0 =A0correspond to an event package which fu=
rther describes the semantics</font></div>


<div><font color=3D"#000099">=A0 =A0of the event or event class.</font></di=
v><div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099=
"><br></font></div><div><font color=3D"#000099">ReplacementText:</font></di=
v><div>

<font color=3D"#000099">=A0 =A0Subscribers MUST include exactly one &quot;E=
vent&quot; header field in</font></div><div><font color=3D"#000099">=A0 =A0=
SUBSCRIBE requests, indicating to which event or class of events they</font=
></div>

<div><font color=3D"#000099">=A0 =A0are subscribing. =A0The &quot;Event&quo=
t; header field **MUST** contain a token which</font></div><div><font color=
=3D"#000099">=A0 =A0indicates the type of state for which a subscription is=
 being</font></div>

<div><font color=3D"#000099">=A0 =A0requested. =A0This token will be regist=
ered with the IANA and will</font></div>
<div><font color=3D"#000099">=A0 =A0correspond to an event package which fu=
rther describes the semantics</font></div><div><font color=3D"#000099">=A0 =
=A0of the event or event class.</font></div><div><font color=3D"#000099">--=
---------------------------------------------------</font></div>

<div><font color=3D"#000099">3.1. =A0SUBSCRIBE</font></div><div><font color=
=3D"#000099"><br>
</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0The SUBSCRIBE method is used to request current=
 state and state</font></div><div><font color=3D"#000099">=A0 =A0updates fr=
om a remote node. =A0SUBSCRIBE requests are target refresh</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
<br></font></div><div><font color=3D"#000099">
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0The SUBSCR=
IBE method is used to request current state and **subsequent** state</font>=
</div><div><font color=3D"#000099">=A0 =A0updates from a remote node. =A0SU=
BSCRIBE requests are target refresh</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
<br></font></div>
<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">3.1.1. =A0Subscription Dur=
ation</font></div><div><font color=3D"#000099"><br></font></div><div><font =
color=3D"#000099">CurrentText:</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0SUBSCRIBE requests SHOULD contain an &quot;Expires&quot; header fiel=
d (defined</font></div>
<div><font color=3D"#000099">=A0 =A0in [RFC3261]). =A0This expires value in=
dicates the duration of the</font></div><div><font color=3D"#000099">=A0 =
=A0subscription. =A0In order to keep subscriptions effective beyond the</fo=
nt></div><div>

<font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expires&q=
uot; header field, subscribers need</font></div>
<div><font color=3D"#000099">=A0 =A0to refresh subscriptions on a periodic =
basis using a new SUBSCRIBE</font></div><div><font color=3D"#000099">=A0 =
=A0request on the same dialog as defined in [RFC3261].</font></div><div><fo=
nt color=3D"#000099"><br>

</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>We can add a =
normative text saying, the refresh time will be</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>n=
ormally 50% to 75% of subscription duration.</font></div><div><font color=
=3D"#000099"><br></font></div><div><font color=3D"#000099">----------------=
-------------------------------------</font></div>

<div><font color=3D"#000099">3.2.1.=A0</font></div><div><font color=3D"#000=
099"><br>
</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0This body will contain either the state of the =
subscribed resource or</font></div><div><font color=3D"#000099">=A0 =A0a po=
inter to such state in the form of a URI (see Section 5.4.13).</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText/Comment:</font></div>
<div><font color=3D"#000099">=A0 =A0This body **SHALL** contain either the =
state of the subscribed resource or</font></div><div><font color=3D"#000099=
">=A0 =A0a pointer to such state in the form of a **URL** (see Section 5.4.=
13).</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.1.2. =A0Creating and Maintaining Subscriptio=
ns</font></div><div><font color=3D"#000099"><br></font></div><div><font col=
or=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentText:</=
font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0From the subscriber&#39;s perspective, a subscription proceeds accor=
ding</font></div><div><font color=3D"#000099">=A0 =A0to the following state=
 diagram:</font></div>


<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
Comment:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>Here it is better to mention, what happens to FSM when =
200-SUBSCRIBE=A0</font></div>

<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>r=
eceived in notify_wait state (because very often this is possible).</font><=
/div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>I=
f no change in state of FSM, then some normative text shall convey the same=
.</font></div><div><font color=3D"#000099">=A0 =A0 =A0 =A0 It may be no-ope=
ration, but better to mention the same.</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">4.1.2.1. =A0Requesting a S=
ubscription</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0When a subscri=
ber wishes to subscribe to a particular state for a</font></div><div><font =
color=3D"#000099">=A0 =A0resource, it forms a SUBSCRIBE request. =A0If the =
initial SUBSCRIBE</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div>
<div><font color=3D"#000099">=A0 =A0When a subscriber wishes to subscribe t=
o a particular state **of** a</font></div><div><font color=3D"#000099">=A0 =
=A0resource, it forms a SUBSCRIBE request **according to sec 3.1 and send i=
t across to notifier**. =A0If the initial SUBSCRIBE</font></div>

<div>
<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">-----=
------------------------------------------------</font></div><div><font col=
or=3D"#000099">4.1.2.1. =A0Requesting a Subscription</font></div><div><font=
 color=3D"#000099"><br>

</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0[RFC3261]. =A0For the sake of clarity: if a SUB=
SCRIBE request contains</font></div>
<div><font color=3D"#000099">=A0 =A0an &quot;Accept&quot; header field, but=
 that field does not indicate a media</font></div><div><font color=3D"#0000=
99">=A0 =A0type that the notifier is capable of generating in its NOTIFY</f=
ont></div>

<div><font color=3D"#000099">=A0 =A0requests, then the proper error respons=
e is 406 (Not Acceptable).</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
COMMENT:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>It is better to mention it as a normative text (indente=
d).</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
-----------------------------------------------------</font></div>
<div><font color=3D"#000099">4.1.3. =A0Receiving and Processing State Infor=
mation</font></div><div><font color=3D"#000099"><br></font></div><div><font=
 color=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099"><b=
r></font></div>

<div><font color=3D"#000099">=A0 =A0If the &quot;Subscription-State&quot; v=
alue is &quot;terminated&quot;, the subscriber</font></div><div><font color=
=3D"#000099">=A0 =A0MUST consider the subscription terminated. =A0The &quot=
;expires&quot; parameter</font></div>


<div><font color=3D"#000099">=A0 =A0has no semantics for &quot;terminated&q=
uot; -- notifiers SHOULD NOT include an</font></div><div><font color=3D"#00=
0099"><br></font></div><div><font color=3D"#000099">ReplacementText:</font>=
</div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If the &quot;Subscription-State&quot; value is &quot;terminated&quot=
;, the subscriber</font></div>
<div><font color=3D"#000099">=A0 =A0MUST consider the subscription terminat=
ed. =A0The &quot;expires&quot; parameter</font></div><div><font color=3D"#0=
00099">=A0 =A0has no semantics for &quot;terminated&quot; **state**. Notifi=
ers SHOULD NOT include an</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0deactivated: =
=A0The subscription has been terminated, but the</font></div><div><font col=
or=3D"#000099">=A0 =A0 =A0 subscriber SHOULD retry immediately with a new s=
ubscription. =A0One</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div>
<div><font color=3D"#000099">=A0 =A0deactivated: =A0The subscription has be=
en terminated, but the</font></div><div><font color=3D"#000099">=A0 =A0 =A0=
 subscriber SHOULD retry with a new subscription. =A0One</font></div><div><=
font color=3D"#000099"><br>

</font></div><div><font color=3D"#000099">=A0 =A0Is It mandatory to resubsc=
ribe immediately. if not remove the word &quot;immediately&quot;.</font></d=
iv>
<div><font color=3D"#000099">=A0 =A0If it is delayed, what are the conseque=
nces ????</font></div><div><font color=3D"#000099"><br></font></div><div><f=
ont color=3D"#000099">-----------------------------------------------------=
</font></div>

<div><font color=3D"#000099">4.2.1. =A0Subscription Establishment and Maint=
enance</font></div><div><font color=3D"#000099"><br></font></div><div><font=
 color=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099">=
=A0 =A0 =A0 inability to signal the success of a REFER request while signal=
ing</font></div>


<div><font color=3D"#000099">=A0 =A0 =A0 a problem with the subscription; a=
nd difficulty performing one</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 action without the other. =A0Additionally, the proper exchange of</=
font></div><div>

<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Repla=
cementText:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 inability =
to signal the success of a REFER request while signaling</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 a problem with the subscription; a=
nd difficulty **in** performing one</font></div><div><font color=3D"#000099=
">=A0 =A0 =A0 action without the other. =A0Additionally, the proper exchang=
e of</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
-----------------------------------------------------</font></div>
<div><font color=3D"#000099">4.2.1.4. =A0Refreshing of Subscriptions</font>=
</div><div><font color=3D"#000099"><br></font></div><div><font color=3D"#00=
0099">CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0If no re=
fresh for a notification address is received before its</font></div>

<div><font color=3D"#000099">=A0 =A0expiration time, the subscription is re=
moved. =A0When removing a</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0If no refr=
esh for a notification is received before its</font></div><div><font color=
=3D"#000099">=A0 =A0expiration time, the subscription is removed. =A0When r=
emoving a</font></div>

<div><span style=3D"white-space:pre-wrap"><font color=3D"#000099">	</font><=
/span></div>
<div><font color=3D"#000099">Comment:<span style=3D"white-space:pre-wrap">	=
</span>Word &quot;address&quot; needs to be removed.</font></div><div><font=
 color=3D"#000099">-----------------------------------------------------</f=
ont></div>

<div><font color=3D"#000099">4.3. =A0Proxy Behavior</font></div><div>
<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Curre=
ntText:</font></div><div><font color=3D"#000099">=A0 =A0Proxies that did no=
t add a Record-Route header field to the initial</font></div><div><font col=
or=3D"#000099">=A0 =A0SUBSCRIBE request MUST NOT add a Record-Route header =
field to any of</font></div>

<div><font color=3D"#000099">=A0 =A0the associated NOTIFY requests.</font><=
/div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0Proxies th=
at did not add a Record-Route header field to the initial</font></div><div>=
<font color=3D"#000099">=A0 =A0SUBSCRIBE request MUST NOT add a Record-Rout=
e header field to any of</font></div>

<div><font color=3D"#000099">=A0 =A0the associated NOTIFY requests **or any=
 SUBSCRIBE refresh requests**.</font></div>
<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">4.4.1. =A0Dialog Creation =
and Termination</font></div><div><font color=3D"#000099"><br></font></div><=
div>

<font color=3D"#000099">CurrentText:</font></div><div><font color=3D"#00009=
9"><br></font></div><div><font color=3D"#000099">=A0 =A0 =A0 A subscriber m=
ay send a SUBSCRIBE request with an &quot;Expires&quot; header</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 field of 0 in order to trigger the=
 sending of such a NOTIFY</font></div><div><font color=3D"#000099">=A0 =A0 =
=A0 request; however, for the purposes of subscription and dialog</font></d=
iv><div><font color=3D"#000099">=A0 =A0 =A0 lifetime, the subscription is n=
ot considered terminated until the</font></div>


<div><font color=3D"#000099">=A0 =A0 =A0 NOTIFY transaction with a &quot;Su=
bscription-State&quot; of &quot;terminated&quot;</font></div><div><font col=
or=3D"#000099">=A0 =A0 =A0 completes.</font></div><div><font color=3D"#0000=
99"><br></font></div>

<div><font color=3D"#000099">Comment:</font></div><div><font color=3D"#0000=
99"><span style=3D"white-space:pre-wrap">	</span>We may have to mention how=
 long the dialog must be kept alive, before=A0</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>b=
eing cleaned up. Do we delete the dialog immediately after 200-SUB=A0</font=
></div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	<=
/span>or wait till NOTIFY with &quot;Subscription-State&quot;=3D&quot;termi=
nated&quot;. If we always=A0</font></div>


<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>w=
ait till Last Notify for dialog termination, then 200-SUB don&#39;t have an=
y=A0</font></div><div><font color=3D"#000099"><span style=3D"white-space:pr=
e-wrap">	</span>impact on the dialog termination. Atleast some normative te=
xt is of=A0</font></div>


<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>n=
ecessity, for better understanding of the topic.</font></div><div><font col=
or=3D"#000099">-----------------------------------------------------</font>=
</div>

<div><font color=3D"#000099">4.4.3. =A0Polling Resource State</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0Note that the NOTIFY requests triggered by =
SUBSCRIBE requests with</font></div>

<div><font color=3D"#000099">=A0 =A0&quot;Expires&quot; header fields of 0 =
will contain a &quot;Subscription-State&quot;</font></div>
<div><font color=3D"#000099">=A0 =A0value of &quot;terminated&quot;, and a =
&quot;reason&quot; parameter of &quot;timeout&quot;.</font></div><div><font=
 color=3D"#000099"><br></font></div><div><font color=3D"#000099">Replacemen=
tText:</font></div>

<div><font color=3D"#000099">=A0 =A0Note that the NOTIFY requests triggered=
 by SUBSCRIBE requests with</font></div>
<div><font color=3D"#000099">=A0 =A0&quot;Expires&quot; header fields of 0 =
**shall** contain a &quot;Subscription-State&quot;</font></div><div><font c=
olor=3D"#000099">=A0 =A0value of &quot;terminated&quot;, and a &quot;reason=
&quot; parameter of &quot;timeout&quot;.</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentT=
ext:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 for a dialog (e.g=
., route set handling). =A0In particular, multiple</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 subscriptions within a dialog are =
expire independently, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 require independent subscription r=
efreshes.</font></div><div><font color=3D"#000099"><br></font></div><div><f=
ont color=3D"#000099">ReplacementText:</font></div><div><font color=3D"#000=
099">=A0 =A0 =A0 for a dialog (e.g., route set handling). =A0In particular,=
 multiple</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 subscriptions within a dialog **op=
erates** independently, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 require independent subscription r=
efreshes.</font></div><div><font color=3D"#000099">------------------------=
-----------------------------</font></div><div><font color=3D"#000099">4.5.=
2. =A0Sharing Dialogs</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0o =A0Subscribe=
rs MAY include an &quot;id&quot; parameter in SUBSCRIBE request</font></div=
>
<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field to =
allow differentiation between multiple</font></div><div><font color=3D"#000=
099"><br></font></div><div><font color=3D"#000099">ReplacementText:</font><=
/div><div>

<font color=3D"#000099">=A0 =A0o =A0Subscribers MAY include an &quot;id&quo=
t; parameter in SUBSCRIBE **request&#39;s**</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field to =
allow differentiation between multiple</font></div><div><font color=3D"#000=
099"><span style=3D"white-space:pre-wrap">				</span>OR</font></div><div><f=
ont color=3D"#000099">=A0 =A0o =A0Subscribers MAY include an &quot;id&quot;=
 parameter in **&quot;Event&quot; header field</font></div>


<div><font color=3D"#000099">=A0 =A0 =A0 of SUBSCRIBE request** to allow di=
fferentiation between multiple</font></div><div><font color=3D"#000099">---=
--------------------------------------------------</font></div><div><font c=
olor=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">
=A0 =A0o =A0When a subscriber refreshes a the subscription timer, the</font=
></div><div><font color=3D"#000099">=A0 =A0 =A0 SUBSCRIBE request MUST cont=
ain the same &quot;Event&quot; header field &quot;id&quot;</font></div><div=
><font color=3D"#000099">=A0 =A0 =A0 parameter as was present in the SUBSCR=
IBE request that created the</font></div>


<div><font color=3D"#000099">=A0 =A0 =A0 subscription. =A0(Otherwise, the n=
otifier will interpret the</font></div><div><font color=3D"#000099"><br></f=
ont></div><div><font color=3D"#000099">ReplacementText:</font></div><div><f=
ont color=3D"#000099">=A0 =A0o =A0When a subscriber refreshes a the subscri=
ption timer, the</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 SUBSCRIBE request MUST contain the=
 same **&quot;id&quot; parameter in &quot;Event&quot;=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 header field, which** was present =
in the SUBSCRIBE request that created the</font></div><div><font color=3D"#=
000099">=A0 =A0 =A0 subscription. =A0(Otherwise, the notifier will interpre=
t the</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentT=
ext:</font></div><div><font color=3D"#000099"><br></font></div><div><font c=
olor=3D"#000099">=A0 =A0o =A0When a subscription is created in the notifier=
, it stores any</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field &qu=
ot;id&quot; parameter as part of the subscription</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 information (along with the event =
package name).</font></div><div><font color=3D"#000099"><br></font></div><d=
iv><font color=3D"#000099">ReplacementText:</font></div><div><font color=3D=
"#000099">=A0 =A0o =A0When a subscription is created in the notifier, it st=
ores **the**</font></div>

<div><font color=3D"#000099">=A0 =A0 =A0 **&quot;id&quot; parameter in &quo=
t;Event&quot; header field** as part of the subscription</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 information (along with the event =
package name).</font></div><div><font color=3D"#000099">-------------------=
----------------------------------</font></div><div><font color=3D"#000099"=
>4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transactions</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0Neither SUBSCRIBE nor NOTIFY requests can be canceled. =A0If a UAS</=
font></div><div><font color=3D"#000099">=A0 =A0receives a CANCEL request th=
at matches a known SUBSCRIBE or NOTIFY</font></div>

<div><font color=3D"#000099">=A0 =A0transaction, it MUST respond to the CAN=
CEL request, but otherwise</font></div>
<div><font color=3D"#000099">=A0 =A0ignore it. =A0In particular, the CANCEL=
 request MUST NOT affect</font></div><div><font color=3D"#000099">=A0 =A0pr=
ocessing of the SUBSCRIBE or NOTIFY request in any way.</font></div><div><f=
ont color=3D"#000099"><br>

</font></div><div><font color=3D"#000099">ReplacementText/Comment:</font></=
div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</sp=
an>What should be the response code for the CANCEL&#39;s reply ???</font></=
div>


<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>P=
referably 200-OK shall be used.</font></div><div><font color=3D"#000099">--=
---------------------------------------------------</font></div><div><font =
color=3D"#000099">4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transact=
ions</font></div>


<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0UACs SHOULD NOT send CANCEL requests for SU=
BSCRIBE or NOTIFY</font></div>

<div><font color=3D"#000099">=A0 =A0transactions.</font></div><div><font co=
lor=3D"#000099"><br></font></div><div><font color=3D"#000099">ReplacementTe=
xt:</font></div><div><font color=3D"#000099">=A0 =A0UACs **MUST** NOT send =
CANCEL requests for SUBSCRIBE or NOTIFY</font></div>


<div><font color=3D"#000099">=A0 =A0transactions. **UASs SHOULD send 200OK =
for CANCEL within NOTIFY=A0</font></div><div><font color=3D"#000099">=A0 =
=A0or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**</font></d=
iv><div><font color=3D"#000099">-------------------------------------------=
----------</font></div>

<div><font color=3D"#000099">5.3.2. =A0State Deltas</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0In the case that the state information to b=
e conveyed is large, the</font></div>

<div><font color=3D"#000099">=A0 =A0event package may choose to detail a sc=
heme by which NOTIFY requests</font></div><div><font color=3D"#000099">
=A0 =A0contain state deltas instead of complete state.</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099"><br></fo=
nt></div><div><font color=3D"#000099">ReplacementText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0In the case that the state information to be co=
nveyed is large, the</font></div>

<div><font color=3D"#000099">=A0 =A0event package may choose a scheme by wh=
ich NOTIFY requests</font></div>
<div><font color=3D"#000099">=A0 =A0contain state deltas instead of complet=
e state.</font></div><div><font color=3D"#000099"><br></font></div><div><fo=
nt color=3D"#000099">comment:<span style=3D"white-space:pre-wrap">	</span>d=
eleted words &quot;to detail&quot; in 2nd line.</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">
-----------------------------------------------------</font></div><div><fon=
t color=3D"#000099">5.3.2. =A0State Deltas</font></div><div><font color=3D"=
#000099"><br></font></div><div><font color=3D"#000099">CurrentText:</font><=
/div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If a NOTIFY request arrives that has a version number that is</font>=
</div><div><font color=3D"#000099">
=A0 =A0incremented by more than one, the subscriber knows that a state delt=
a</font></div><div><font color=3D"#000099">=A0 =A0has been missed; it ignor=
es the NOTIFY request containing the state</font></div><div><font color=3D"=
#000099">=A0 =A0delta (except for the version number, which it retains to d=
etect</font></div>


<div><font color=3D"#000099">=A0 =A0message loss), and re-sends a SUBSCRIBE=
 request to force a NOTIFY</font></div><div><font color=3D"#000099">=A0 =A0=
request containing a complete state snapshot.</font></div><div><font color=
=3D"#000099"><br>

</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099">=A0 =A0From this text, it infers that if any state delta l=
ost, then subscriber=A0</font></div>
<div><font color=3D"#000099">=A0 =A0sends refresh SUBSCRIBE (re-SUBSCRIBE) =
request. This re-SUBSCRIBE request inturn lead=A0</font></div><div><font co=
lor=3D"#000099">=A0 =A0to fetching the complete state information.=A0</font=
></div><div>

<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=A0 =
=A0It may be a good diea to differentiate the re-SUBSCRIBE triggered by dif=
ferent scenarios.</font></div>
<div><font color=3D"#000099">=A0 =A0a) natural re-SUBSCRIBE request, which =
extends the SUBSCRIBE duration. In this=A0</font></div><div><font color=3D"=
#000099">=A0 =A0 =A0 case it is <b>not </b>necessary to fetch the complete =
state information</font></div>

<div><font color=3D"#000099">=A0 =A0b) re-SUBSCRIBE triggered because of lo=
st NOTIFY request (ref sec 5.3.2.=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 VERSION number in received NOTIFY =
is increased by greater than ONE).=A0</font></div><div><font color=3D"#0000=
99">=A0 =A0 =A0 In this case subscriber wants to fetch the complete state i=
nformation.</font></div>

<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If we don&#39;t differentiate these two cases, then natural re-SUBSC=
RIBE request</font></div>
<div><font color=3D"#000099">=A0 =A0unnecessarily fetch the complete state =
information, which may not be necessary / expected.=A0</font></div><div><fo=
nt color=3D"#000099">=A0 =A0But uses higher network bandwidth to fetch the =
complete state information.</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">5.4.7. =A0Notifier generation of NOTIFY reques=
ts</font></div><div><font color=3D"#000099"><br></font></div><div><font col=
or=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099">=A0 =
=A0This section may optionally describe the behavior used to process the</f=
ont></div>

<div><font color=3D"#000099">=A0 =A0subsequent response.</font></div><div><=
font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099">=A0 =A0In this section 5.4.7, the above statement is not c=
lear, what is really meant to be.</font></div><div><font color=3D"#000099">=
<br></font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">5.4.13. =A0Use of URIs to =
Retrieve State</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">ReplacementText:</font></div><div><font color=3D"#=
000099">=A0 =A0In this context, I feel Use of &quot;URLs&quot; is more appr=
opriate than &quot;URIs&quot;.=A0</font></div>

<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
</div><div><font color=3D"#000099"><br></font></div><div><font color=3D"#00=
0099"><br></font></div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace">Thanks,</font></font><div><font face=3D"&#39;courier ne=
w&#39;, monospace" color=3D"#000099">Nataraju A.B.</font></div>

<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">E-ID=
 : <a href=3D"mailto:nataraju.ab@gmail.com" target=3D"_blank">nataraju.ab@g=
mail.com</a> / <a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">=
nataraju.sip@gmail.com</a></font></div>
<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">



+91-98455-95744
</font></div>

--f46d0430855a93ad1604bd92b957--

From ibc@aliax.net  Mon Apr 16 09:49:59 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 16 Apr 2012 18:49:38 +0200
Message-ID: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com>
To: SIPCORE WG <sipcore@ietf.org>
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Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-websocket-02.txt
X-BeenThere: sipcore@ietf.org
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A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
successfully submitted by I=C3=B1aki Baz Castillo and posted to the IETF
repository.

Filename: =C2=A0 =C2=A0 =C2=A0 =C2=A0draft-ibc-sipcore-sip-websocket
Revision: =C2=A0 =C2=A0 =C2=A0 =C2=A002
Title: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 The WebSocket Protocol as a Trans=
port for the Session
Initiation Protocol (SIP)
Creation date: =C2=A0 2012-04-16
WG ID: =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Individual Submission
Number of pages: 20

Abstract:
=C2=A0 The WebSocket protocol enables two-way realtime communication betwee=
n
=C2=A0 clients and servers. =C2=A0This document specifies a new WebSocket s=
ub-
=C2=A0 protocol as a reliable transport mechanism between SIP (Session
=C2=A0 Initiation Protocol) entities and enables usage of the SIP protocol
=C2=A0 in new scenarios.


- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt


This new version incorporates improvements and changes based on the
comments and reviews on the previous version, along with the consensus
got in IETF 83 Paris.


As usual, comments are welcome. Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From adam@nostrum.com  Mon Apr 16 11:02:48 2012
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Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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[as chair]

SIPCORE Working Group:

This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02 
document to satisfy our recently added milestone to define a WebSockets 
transport for the SIP protocol. Interested parties should voice their 
support and/or concerns on the SIPCORE mailing list. The chairs plan to 
make a determination of consensus on Friday, April 27th.

Thank you. The draft announcement follows.

/a

-------- Original Message --------
Subject: [sipcore] New Version Notification for 
draft-ibc-sipcore-sip-websocket-02.txt
Date: Mon, 16 Apr 2012 18:49:38 +0200
From: IĆ±aki Baz Castillo <ibc@aliax.net>
To: SIPCORE WG <sipcore@ietf.org>

A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
successfully submitted by IĆ±aki Baz Castillo and posted to the IETF
repository.

Filename:        draft-ibc-sipcore-sip-websocket
Revision:        02
Title:           The WebSocket Protocol as a Transport for the Session
Initiation Protocol (SIP)
Creation date:   2012-04-16
WG ID:           Individual Submission
Number of pages: 20

Abstract:
   The WebSocket protocol enables two-way realtime communication between
   clients and servers.  This document specifies a new WebSocket sub-
   protocol as a reliable transport mechanism between SIP (Session
   Initiation Protocol) entities and enables usage of the SIP protocol
   in new scenarios.


- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt


This new version incorporates improvements and changes based on the
comments and reviews on the previous version, along with the consensus
got in IETF 83 Paris.


As usual, comments are welcome. Thanks a lot.


-- 
IĆ±aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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On 04/16/2012 01:02 PM, Adam Roach - SIPCORE Chair wrote:
> [as chair]
>
> SIPCORE Working Group:
>
> This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
> document to satisfy our recently added milestone to define a WebSockets
> transport for the SIP protocol. Interested parties should voice their
> support and/or concerns on the SIPCORE mailing list. The chairs plan to
> make a determination of consensus on Friday, April 27th.

I support adoption of draft-ibc-sip-websocket-02 to satisfy the newly
added milestone.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
Web:   http://ect.bell-labs.com/who/vkg/

From gsalguei@cisco.com  Mon Apr 16 11:14:13 2012
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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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+1 on draft-ibc-sip-websocket-02 being adopted as WG item to address =
WebSockets milestone

--G


On Apr 16, 2012, at 2:09 PM, Vijay K. Gurbani wrote:

> On 04/16/2012 01:02 PM, Adam Roach - SIPCORE Chair wrote:
>> [as chair]
>>=20
>> SIPCORE Working Group:
>>=20
>> This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
>> document to satisfy our recently added milestone to define a =
WebSockets
>> transport for the SIP protocol. Interested parties should voice their
>> support and/or concerns on the SIPCORE mailing list. The chairs plan =
to
>> make a determination of consensus on Friday, April 27th.
>=20
> I support adoption of draft-ibc-sip-websocket-02 to satisfy the newly
> added milestone.
>=20
> Thanks,
>=20
> - vijay
> --=20
> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
> Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
> Web:   http://ect.bell-labs.com/who/vkg/
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20


From salvatore.loreto@ericsson.com  Mon Apr 16 11:15:48 2012
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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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+1

On 4/16/12 8:14 PM, Gonzalo Salgueiro wrote:
> +1 on draft-ibc-sip-websocket-02 being adopted as WG item to address WebSockets milestone
>
> --G
>
>
> On Apr 16, 2012, at 2:09 PM, Vijay K. Gurbani wrote:
>
>> On 04/16/2012 01:02 PM, Adam Roach - SIPCORE Chair wrote:
>>> [as chair]
>>>
>>> SIPCORE Working Group:
>>>
>>> This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
>>> document to satisfy our recently added milestone to define a WebSockets
>>> transport for the SIP protocol. Interested parties should voice their
>>> support and/or concerns on the SIPCORE mailing list. The chairs plan to
>>> make a determination of consensus on Friday, April 27th.
>> I support adoption of draft-ibc-sip-websocket-02 to satisfy the newly
>> added milestone.
>>
>> Thanks,
>>
>> - vijay
>> -- 
>> Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>> 1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
>> Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
>> Web:   http://ect.bell-labs.com/who/vkg/
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



From christer.holmberg@ericsson.com  Mon Apr 16 11:34:55 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'SIPCORE Chairs' <sipcore-chairs@tools.ietf.org>, "SIPCORE (Session Initiation Protocol Core) WG" <sipcore@ietf.org>
Date: Mon, 16 Apr 2012 20:34:49 +0200
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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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I support adoption of the draft.

Regards,

Christer=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Adam Roach - SIPCORE Chair
Sent: 16. huhtikuuta 2012 21:03
To: SIPCORE (Session Initiation Protocol Core) WG
Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02

[as chair]

SIPCORE Working Group:

This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
document to satisfy our recently added milestone to define a WebSockets tra=
nsport for the SIP protocol. Interested parties should voice their support =
and/or concerns on the SIPCORE mailing list. The chairs plan to make a dete=
rmination of consensus on Friday, April 27th.

Thank you. The draft announcement follows.

/a

-------- Original Message --------
Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-webso=
cket-02.txt
Date: Mon, 16 Apr 2012 18:49:38 +0200
From: I=F1aki Baz Castillo <ibc@aliax.net>
To: SIPCORE WG <sipcore@ietf.org>

A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been succe=
ssfully submitted by I=F1aki Baz Castillo and posted to the IETF repository=
.

Filename:        draft-ibc-sipcore-sip-websocket
Revision:        02
Title:           The WebSocket Protocol as a Transport for the Session
Initiation Protocol (SIP)
Creation date:   2012-04-16
WG ID:           Individual Submission
Number of pages: 20

Abstract:
   The WebSocket protocol enables two-way realtime communication between
   clients and servers.  This document specifies a new WebSocket sub-
   protocol as a reliable transport mechanism between SIP (Session
   Initiation Protocol) entities and enables usage of the SIP protocol
   in new scenarios.


- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt


This new version incorporates improvements and changes based on the comment=
s and reviews on the previous version, along with the consensus got in IETF=
 83 Paris.


As usual, comments are welcome. Thanks a lot.


--
I=F1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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On 04/16/2012 01:34 PM, Christer Holmberg wrote:
>
> I support adoption of the draft.

I also support adoption of the draft.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From saul@ag-projects.com  Mon Apr 16 12:28:12 2012
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I do support the adoption of this draft.


Regards,

On Apr 16, 2012, at 8:02 PM, Adam Roach - SIPCORE Chair wrote:

> [as chair]
>=20
> SIPCORE Working Group:
>=20
> This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02 =
document to satisfy our recently added milestone to define a WebSockets =
transport for the SIP protocol. Interested parties should voice their =
support and/or concerns on the SIPCORE mailing list. The chairs plan to =
make a determination of consensus on Friday, April 27th.
>=20
> Thank you. The draft announcement follows.
>=20
> /a
>=20
> -------- Original Message --------
> Subject: [sipcore] New Version Notification for =
draft-ibc-sipcore-sip-websocket-02.txt
> Date: Mon, 16 Apr 2012 18:49:38 +0200
> From: I=F1aki Baz Castillo <ibc@aliax.net>
> To: SIPCORE WG <sipcore@ietf.org>
>=20
> A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
> successfully submitted by I=F1aki Baz Castillo and posted to the IETF
> repository.
>=20
> Filename:        draft-ibc-sipcore-sip-websocket
> Revision:        02
> Title:           The WebSocket Protocol as a Transport for the Session
> Initiation Protocol (SIP)
> Creation date:   2012-04-16
> WG ID:           Individual Submission
> Number of pages: 20
>=20
> Abstract:
>  The WebSocket protocol enables two-way realtime communication between
>  clients and servers.  This document specifies a new WebSocket sub-
>  protocol as a reliable transport mechanism between SIP (Session
>  Initiation Protocol) entities and enables usage of the SIP protocol
>  in new scenarios.
>=20
>=20
> - http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
> - http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
>=20
>=20
> This new version incorporates improvements and changes based on the
> comments and reviews on the previous version, along with the consensus
> got in IETF 83 Paris.
>=20
>=20
> As usual, comments are welcome. Thanks a lot.
>=20
>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From internet-drafts@ietf.org  Tue Apr 17 00:02:48 2012
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-proxy-feature-01.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : Indication of features supported by proxy
	Author(s)       : Christer Holmberg
                          Ivo Sedlacek
                          Hadriel Kaplan
	Filename        : draft-ietf-sipcore-proxy-feature-01.txt
	Pages           : 17
	Date            : 2012-04-17

   This specification creates a new IANA registry, "SIP Feature Cap
   Registry", which is used to register indicators, "SIP feature caps",
   used by SIP entities to indicate support of features and
   capabilities, in cases where the Contact header field contains a URI
   that does not represent the SIP entity that wants to indicate support
   of its features and capabilities.

   This specification also defines a new SIP header field, Feature-Caps,
   that can be used by SIP entities to convey information about
   supported features and capabilities, using SIP feature caps.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-proxy-feature-01.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-proxy-feature-01.txt


From christer.holmberg@ericsson.com  Tue Apr 17 00:05:32 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 17 Apr 2012 09:05:26 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-proxy-feature-01
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Hi,

I've submitted a new version (-01) of the proxy-feature draft.

Based on the discussions in Paris, instead of using feature tags the draft =
now creates a new IANA "feature cap" registry.

Regards,

Christer

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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span lang=3DFI>=
Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</=
o:p></span></p><p class=3DMsoNormal>I&#8217;ve submitted a new version (-01=
) of the proxy-feature draft.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp=
;</o:p></p><p class=3DMsoNormal>Based on the discussions in Paris, instead =
of using feature tags the draft now creates a new IANA &#8220;feature cap&#=
8221; registry.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>Regards,<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</=
o:p></p><p class=3DMsoNormal>Christer<o:p></o:p></p></div></body></html>=

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Subject: Re: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
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16 apr 2012 kl. 20:02 skrev Adam Roach - SIPCORE Chair:

> [as chair]
>=20
> SIPCORE Working Group:
>=20
> This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02 =
document to satisfy our recently added milestone to define a WebSockets =
transport for the SIP protocol. Interested parties should voice their =
support and/or concerns on the SIPCORE mailing list. The chairs plan to =
make a determination of consensus on Friday, April 27th.
I support adoption of this document.

/O
>=20
> Thank you. The draft announcement follows.
>=20
> /a
>=20
> -------- Original Message --------
> Subject: [sipcore] New Version Notification for =
draft-ibc-sipcore-sip-websocket-02.txt
> Date: Mon, 16 Apr 2012 18:49:38 +0200
> From: I=F1aki Baz Castillo <ibc@aliax.net>
> To: SIPCORE WG <sipcore@ietf.org>
>=20
> A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
> successfully submitted by I=F1aki Baz Castillo and posted to the IETF
> repository.
>=20
> Filename:        draft-ibc-sipcore-sip-websocket
> Revision:        02
> Title:           The WebSocket Protocol as a Transport for the Session
> Initiation Protocol (SIP)
> Creation date:   2012-04-16
> WG ID:           Individual Submission
> Number of pages: 20
>=20
> Abstract:
>  The WebSocket protocol enables two-way realtime communication between
>  clients and servers.  This document specifies a new WebSocket sub-
>  protocol as a reliable transport mechanism between SIP (Session
>  Initiation Protocol) entities and enables usage of the SIP protocol
>  in new scenarios.
>=20
>=20
> - http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
> - http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
>=20
>=20
> This new version incorporates improvements and changes based on the
> comments and reviews on the previous version, along with the consensus
> got in IETF 83 Paris.
>=20
>=20
> As usual, comments are welcome. Thanks a lot.
>=20
>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
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---
* Olle E Johansson - oej@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




From nataraju.sip@gmail.com  Tue Apr 17 09:46:30 2012
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Date: Tue, 17 Apr 2012 22:16:24 +0530
Message-ID: <CA+rAfUPREpXWSuF2erYYcGdo5F8tJXRZmfRB67sHmgzo1j0sXg@mail.gmail.com>
From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, jmillan@aliax.net,  vpascual@acmepacket.com, sipcore@ietf.org
Content-Type: multipart/alternative; boundary=bcaec554d9aae3688704bde2adcc
Subject: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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--bcaec554d9aae3688704bde2adcc
Content-Type: text/plain; charset=ISO-8859-1

Hi Authros,

Please find some comments on the draft
"draft-ibc-sipcore-sip-websocket-02.txt"

For simplicity, I have mentioned both the current text and proposed text.

The comments / corrective text can be found with prefix and suffix  --- **

-----------------------------------------------------
Section :  8.1.  Registration
CurrentText:
   Alice    (SIP WSS)    proxy.atlanta.com
   |                             |
   |REGISTER F1                  |
   |---------------------------->|
   |200 OK F2                    |
   |<----------------------------|
   |                             |

ReplacementText / Comment:

   Alice    (SIP WSS)    proxy.atlanta.com
   |                             |
   |                             |
   |WebSocket Con Req F1         |
   |---------------------------->|
   |WebSocket Con Reply F2       |
   |<----------------------------|
   |                             |
   |                             |
   |REGISTER F3                  |
   |---------------------------->|
   |200 OK F4                    |
   |<----------------------------|
   |                             |

   Prefixed the WebSocket connection request followed by SIP registrations.
   F1, F2 will be same messages mentioned in sec 4.1

-----------------------------------------------------
Section :  8.1.  Registration
CurrentText:
   F1 REGISTER  Alice -> proxy.atlanta.com (transport WSS)

   REGISTER sip:proxy.atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf

   F2 200 OK  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf

ReplacementText / Comment:

If I understand right,
          Websocket connection is mapped on to SIP URI  --> HTTP
   Secure Websocket connection is mapped on to SIPS URI --> HTTPS

HTTP --> WS --> SIP URI --> port 80

HTTPS --> WSS --> SIPS URI --> port 443

Hence in the following messages "WSS" shall be replaced with "WS"

   F1 REGISTER  Alice -> proxy.atlanta.com (transport WSS)

   REGISTER sip:proxy.atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf

WSS --> WS

   F2 200 OK  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf

WSS --> WS


Probably in the appendix A.2, we can list out the call flows for *""SECURE
WEB SOCKET"" (WSS)*

For example, REGISTER request looks like this (with WSS)...

   REGISTER sips:proxy.atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKasudf
   From: sips:alice@atlanta.com;tag=65bnmj.34asd
   To: sips:alice@atlanta.com
   Call-ID: aiuy7k9njasd
   CSeq: 1 REGISTER
   Max-Forwards: 70
   Supported: path, outbound, gruu
   Contact: <sips:alice@df7jal23ls0d.invalid;transport=ws>
     ;reg-id=1
     ;+sip.instance="<urn:uuid:f81-7dec-14a06cf1>"


   changes:
   sip --> sips in RURI, From, TO, contact headers

-----------------------------------------------------
Section : 8.2.  INVITE dialog through a proxy
CurrentText:

   F1 INVITE  Alice -> proxy.atlanta.com (transport WSS)

   INVITE sip:bob@atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   From: sip:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 70
   Supported: path, outbound, gruu
   Route: <sip:proxy.atlanta.com:443;transport=ws;lr>
   Contact: <sip:alice@atlanta.com
    ;gr=urn:uuid:f81-7dec-14a06cf1;ob>"
   Content-Type: application/sdp


ReplacementText / Comment:

   Extra closing double quote in contact header.

      Contact: <sip:alice@atlanta.com
          ;gr=urn:uuid:f81-7dec-14a06cf1;ob>

   Route: <sip:proxy.atlanta.com:80;transport=ws;lr>

   change :
        443-->80              assuming HTTP <-- WS <-- SIP URI
        This comment is applicable for messages F1, F3, F4, F5, F6, F8, F9,
F10,

   Changes
WSS --> WS            assuming HTTP <-- WS <-- SIP URI
        This comment is applicable for messages F1, F2, F3, F4, F5, F6, F7,


-----------------------------------------------------
Section : 1.  Introduction

CurrentText:
Media transport is out of the scope of this document

ReplacementText / Comment:
   Any other draft / rfc which addresses this issue. it would improve the
   readability of the document if we can is suggest reference for
completeness.

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

Section : 5.1.  General
CurrentText:
      (typically web-based applications with an strict and simple API
      for receiving WebSocket messages).  There is no need to establish


ReplacementText / Comment:
      (typically web-based applications with **a** strict and simple API
      for receiving WebSocket messages).  There is no need to establish

-----------------------------------------------------
Section :
CurrentText:
   In the absence of an explicit port and DNS SRV resource records, the
   default port for a SIP URI with "ws" transport parameter is 80 in
   case of SIP scheme and 443 in case of SIPS scheme.


ReplacementText / Comment:
   In the absence of an explicit port and DNS SRV resource records, the
   default port for a SIP URI with "ws" transport parameter is 80 in
   case of SIP scheme and 443 in case of SIPS scheme **, which is same as
   that used for http and https protocols respectively**

These ports have been used for the sake of backward compatibility with
servers
which does not support WS/WSS.

-----------------------------------------------------
Section :
CurrentText:

 _This section is non-normative._

ReplacementText / Comment:
 Some sections have been explicitly mentioned with this text. Does that mean
 other sections are normative text. Not clear though.

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

Thanks,
Nataraju A.B.
nataraju.ab@gmail.com / nataraju.sip@gmail.com
+91-98455-95744

--bcaec554d9aae3688704bde2adcc
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<font face=3D"&#39;courier new&#39;, monospace"><div style=3D"color:rgb(51,=
51,255)">Hi Authros,</div><div style=3D"color:rgb(51,51,255)"><br></div><di=
v style=3D"color:rgb(51,51,255)">Please find some comments on the draft &qu=
ot;draft-ibc-sipcore-sip-websocket-02.txt&quot;</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">For simplicity, I have mentioned both the current text and proposed =
text.</div><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"colo=
r:rgb(51,51,255)">
The comments / corrective text can be found with prefix and suffix =A0--- *=
*</div><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rg=
b(51,51,255)">-----------------------------------------------------</div><d=
iv style=3D"color:rgb(51,51,255)">
Section : =A08.1. =A0Registration</div><div style=3D"color:rgb(51,51,255)">=
CurrentText:</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Alice =A0 =A0(=
SIP WSS) =A0 =A0<a href=3D"http://proxy.atlanta.com">proxy.atlanta.com</a><=
/div><div style=3D"color:rgb(51,51,255)">
=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><di=
v style=3D"color:rgb(51,51,255)">=A0 =A0|REGISTER F1 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0|</div><div style=3D"color:rgb(51,51,255)">=A0 =A0|---------=
-------------------&gt;|</div><div style=3D"color:rgb(51,51,255)">
=A0 =A0|200 OK F2 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|</div><div style=
=3D"color:rgb(51,51,255)">=A0 =A0|&lt;----------------------------|</div><d=
iv style=3D"color:rgb(51,51,255)">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 |</div><div style=3D"color:rgb(51,51,255)">
<br></div><div style=3D"color:rgb(51,51,255)">ReplacementText / Comment:</d=
iv><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51=
,51,255)">=A0 =A0Alice =A0 =A0(SIP WSS) =A0 =A0<a href=3D"http://proxy.atla=
nta.com">proxy.atlanta.com</a></div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><div st=
yle=3D"color:rgb(51,51,255)">=A0 =A0|WebSocket Con Req F1 =A0 =A0 =A0 =A0 |=
</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0|----------------------------&gt=
;|</div><div style=3D"color:rgb(51,51,255)">=A0 =A0|WebSocket Con Reply F2 =
=A0 =A0 =A0 |</div><div style=3D"color:rgb(51,51,255)">=A0 =A0|&lt;--------=
--------------------|</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><div st=
yle=3D"color:rgb(51,51,255)">=A0 =A0|REGISTER F3 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0|</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0|----------------------------&gt=
;|</div><div style=3D"color:rgb(51,51,255)">=A0 =A0|200 OK F4 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0|</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0|&lt;----------------------------|</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 |</div><div style=3D"color:rgb(51,51,255)"><br>=
</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Prefixed the WebSocket con=
nection request followed by SIP registrations.=A0</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0F1, F2 will be same messages men=
tioned in sec 4.1</div><div style=3D"color:rgb(51,51,255)"><br></div><div s=
tyle=3D"color:rgb(51,51,255)">---------------------------------------------=
--------</div>
<div style=3D"color:rgb(51,51,255)">Section : =A08.1. =A0Registration</div>=
<div style=3D"color:rgb(51,51,255)">CurrentText:</div><div style=3D"color:r=
gb(51,51,255)">=A0 =A0F1 REGISTER =A0Alice -&gt; <a href=3D"http://proxy.at=
lanta.com">proxy.atlanta.com</a> (transport WSS)</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0REGISTER sip:<a href=3D"http://proxy.atlanta.com">proxy.atlan=
ta.com</a> SIP/2.0</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Via: SIP=
/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0F2 200 OK =A0<a href=3D"http://proxy.atlanta.com">proxy.atlan=
ta.com</a> -&gt; Alice (transport WSS)</div><div style=3D"color:rgb(51,51,2=
55)"><br></div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0SIP/2.0 200 OK</div><div style=
=3D"color:rgb(51,51,255)">=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;bran=
ch=3Dz9hG4bKasudf</div><div style=3D"color:rgb(51,51,255)"><br></div><div s=
tyle=3D"color:rgb(51,51,255)">
ReplacementText / Comment:</div><div style=3D"color:rgb(51,51,255)"><br></d=
iv><div style=3D"color:rgb(51,51,255)">If I understand right,=A0</div><div =
style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 =A0 =A0 Websocket connection is =
mapped on to SIP URI =A0--&gt; HTTP</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0Secure Websocket connection is m=
apped on to SIPS URI --&gt; HTTPS</div><div style=3D"color:rgb(51,51,255)">=
<br></div><div style=3D"color:rgb(51,51,255)">HTTP --&gt; WS --&gt; SIP URI=
 --&gt; port 80</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">HTTPS --&gt; WSS --&gt; SIPS URI --&gt; port 443</div><div style=3D"=
color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,255)">Hence i=
n the following messages &quot;WSS&quot; shall be replaced with &quot;WS&qu=
ot;</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0F1 REGISTER =A0Alice -&gt; <a href=3D"http://proxy.atlanta.co=
m">proxy.atlanta.com</a> (transport WSS)</div><div style=3D"color:rgb(51,51=
,255)"><br>
</div><div style=3D"color:rgb(51,51,255)">=A0 =A0REGISTER sip:<a href=3D"ht=
tp://proxy.atlanta.com">proxy.atlanta.com</a> SIP/2.0</div><div style=3D"co=
lor:rgb(51,51,255)">=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz=
9hG4bKasudf</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>WSS=
 --&gt; WS</div><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D=
"color:rgb(51,51,255)">
=A0 =A0F2 200 OK =A0<a href=3D"http://proxy.atlanta.com">proxy.atlanta.com<=
/a> -&gt; Alice (transport WSS)</div><div style=3D"color:rgb(51,51,255)"><b=
r></div><div style=3D"color:rgb(51,51,255)">=A0 =A0SIP/2.0 200 OK</div><div=
 style=3D"color:rgb(51,51,255)">
=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf</div><di=
v style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,25=
5)"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>WSS --=
&gt; WS</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)"><br></div><div style=3D"color:rgb(51,51,255)">Probably in the append=
ix A.2, we can list out the call flows for <b>&quot;&quot;SECURE WEB SOCKET=
&quot;&quot; (WSS)</b></div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">For example, REGISTER request looks like this (with WSS)...</div><di=
v style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,25=
5)">
=A0 =A0REGISTER sips:<a href=3D"http://proxy.atlanta.com">proxy.atlanta.com=
</a> SIP/2.0</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Via: SIP/2.0/W=
SS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf</div><div style=3D"color:rgb(=
51,51,255)">
=A0 =A0From: <a href=3D"mailto:sips%3Aalice@atlanta.com">sips:alice@atlanta=
.com</a>;tag=3D65bnmj.34asd</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0To: <a href=3D"mailto:sips%3Aalice@atlanta.com">sips:alice@atlanta.com</=
a></div><div style=3D"color:rgb(51,51,255)">
=A0 =A0Call-ID: aiuy7k9njasd</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0CSeq: 1 REGISTER</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Max-For=
wards: 70</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Supported: path, =
outbound, gruu</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0Contact: &lt;sips:alice@df7jal23=
ls0d.invalid;transport=3Dws&gt;</div><div style=3D"color:rgb(51,51,255)">=
=A0 =A0 =A0;reg-id=3D1</div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0=
;+sip.instance=3D&quot;&lt;urn:uuid:f81-7dec-14a06cf1&gt;&quot;</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)"><br></div><div style=3D"color:rgb(51,51,255)">=A0 =A0changes:</div><=
div style=3D"color:rgb(51,51,255)">=A0 =A0sip --&gt; sips in RURI, From, TO=
, contact headers</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0</div><div style=3D"color:rgb(51=
,51,255)">-----------------------------------------------------</div><div s=
tyle=3D"color:rgb(51,51,255)">Section : 8.2. =A0INVITE dialog through a pro=
xy</div>
<div style=3D"color:rgb(51,51,255)">CurrentText:</div><div style=3D"color:r=
gb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,255)">=A0 =A0F1 INVI=
TE =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com">proxy.atlanta.com</a=
> (transport WSS)</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0INVITE <a href=3D"mailto:sip%3Abob@atlanta.com">sip:bob@atlan=
ta.com</a> SIP/2.0</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Via: SIP=
/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bK56sdasks</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0From: <a href=3D"mailto:sip%3Aal=
ice@atlanta.com">sip:alice@atlanta.com</a>;tag=3Dasdyka899</div><div style=
=3D"color:rgb(51,51,255)">=A0 =A0To: <a href=3D"mailto:sip%3Abob@atlanta.co=
m">sip:bob@atlanta.com</a></div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0Call-ID: asidkj3ss</div><div sty=
le=3D"color:rgb(51,51,255)">=A0 =A0CSeq: 1 INVITE</div><div style=3D"color:=
rgb(51,51,255)">=A0 =A0Max-Forwards: 70</div><div style=3D"color:rgb(51,51,=
255)">=A0 =A0Supported: path, outbound, gruu</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0Route: &lt;sip:proxy.atlanta.com=
:443;transport=3Dws;lr&gt;</div><div style=3D"color:rgb(51,51,255)">=A0 =A0=
Contact: &lt;<a href=3D"mailto:sip%3Aalice@atlanta.com">sip:alice@atlanta.c=
om</a></div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0 ;gr=3Durn:uuid:f81-7dec-14a06cf=
1;ob&gt;&quot;</div><div style=3D"color:rgb(51,51,255)">=A0 =A0Content-Type=
: application/sdp</div><div style=3D"color:rgb(51,51,255)"><br></div><div s=
tyle=3D"color:rgb(51,51,255)">
<br></div><div style=3D"color:rgb(51,51,255)">ReplacementText / Comment:</d=
iv><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51=
,51,255)">=A0 =A0Extra closing double quote in contact header.</div><div st=
yle=3D"color:rgb(51,51,255)">
<br></div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 Contact: &lt;<a h=
ref=3D"mailto:sip%3Aalice@atlanta.com">sip:alice@atlanta.com</a></div><div =
style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 =A0 =A0 ;gr=3Durn:uuid:f81-7dec-=
14a06cf1;ob&gt;</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0Route: &lt;sip:proxy.atlanta.com:80;transport=3Dws;lr&gt; =A0=
=A0</div><div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:=
rgb(51,51,255)">
=A0 =A0change :=A0</div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 =A0=
 443--&gt;80 =A0 =A0 =A0 =A0 =A0 =A0 =A0assuming=A0HTTP=A0&lt;--=A0WS &lt;-=
- SIP URI</div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 =A0 This com=
ment is applicable for messages F1, F3, F4, F5, F6, F8, F9, F10,=A0</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">=A0 =A0Changes</div><div style=3D"color:rgb(51,51,255)"><span class=
=3D"Apple-tab-span" style=3D"white-space:pre">	</span>WSS --&gt; WS =A0 =A0=
 =A0 =A0 =A0 =A0assuming=A0HTTP=A0&lt;--=A0WS &lt;-- SIP URI</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 =A0 This comment is applica=
ble for messages F1, F2, F3, F4, F5, F6, F7,=A0</div><div style=3D"color:rg=
b(51,51,255)"><br></div><div style=3D"color:rgb(51,51,255)"><br></div><div =
style=3D"color:rgb(51,51,255)">
-----------------------------------------------------</div><div style=3D"co=
lor:rgb(51,51,255)">Section : 1. =A0Introduction</div><div style=3D"color:r=
gb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,255)">CurrentText:</=
div>
<div style=3D"color:rgb(51,51,255)">Media transport is out of the scope of =
this document</div><div style=3D"color:rgb(51,51,255)"><br></div><div style=
=3D"color:rgb(51,51,255)">ReplacementText / Comment:</div><div style=3D"col=
or:rgb(51,51,255)">
=A0 =A0Any other draft / rfc which addresses this issue. it would improve t=
he=A0</div><div style=3D"color:rgb(51,51,255)">=A0 =A0readability of the do=
cument if we can is suggest reference for completeness.</div><div style=3D"=
color:rgb(51,51,255)">
<br></div><div style=3D"color:rgb(51,51,255)">-----------------------------=
------------------------</div><div style=3D"color:rgb(51,51,255)"><br></div=
><div style=3D"color:rgb(51,51,255)">Section : 5.1. =A0General</div><div st=
yle=3D"color:rgb(51,51,255)">
CurrentText:</div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 (typicall=
y web-based applications with an strict and simple API</div><div style=3D"c=
olor:rgb(51,51,255)">=A0 =A0 =A0 for receiving WebSocket messages). =A0Ther=
e is no need to establish</div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)"><br></div><div style=3D"color:rgb(51,51,255)">ReplacementText / Comm=
ent:</div><div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 (typically web-ba=
sed applications with **a** strict and simple API</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0 =A0 for receiving WebSocket mes=
sages). =A0There is no need to establish</div><div style=3D"color:rgb(51,51=
,255)"><br></div><div style=3D"color:rgb(51,51,255)">----------------------=
-------------------------------</div>
<div style=3D"color:rgb(51,51,255)">Section :=A0</div><div style=3D"color:r=
gb(51,51,255)">CurrentText:</div><div style=3D"color:rgb(51,51,255)">=A0 =
=A0In the absence of an explicit port and DNS SRV resource records, the</di=
v><div style=3D"color:rgb(51,51,255)">
=A0 =A0default port for a SIP URI with &quot;ws&quot; transport parameter i=
s 80 in</div><div style=3D"color:rgb(51,51,255)">=A0 =A0case of SIP scheme =
and 443 in case of SIPS scheme.</div><div style=3D"color:rgb(51,51,255)"><b=
r></div>
<div style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51=
,255)">ReplacementText / Comment:</div><div style=3D"color:rgb(51,51,255)">=
=A0 =A0In the absence of an explicit port and DNS SRV resource records, the=
</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0default port for a SIP URI with =
&quot;ws&quot; transport parameter is 80 in</div><div style=3D"color:rgb(51=
,51,255)">=A0 =A0case of SIP scheme and 443 in case of SIPS scheme **, whic=
h is same as=A0</div>
<div style=3D"color:rgb(51,51,255)">=A0 =A0that used for http and https pro=
tocols respectively**</div><div style=3D"color:rgb(51,51,255)"><br></div><d=
iv style=3D"color:rgb(51,51,255)">These ports have been used for the sake o=
f backward compatibility with servers=A0</div>
<div style=3D"color:rgb(51,51,255)">which does not support WS/WSS.</div><di=
v style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,25=
5)">-----------------------------------------------------</div><div style=
=3D"color:rgb(51,51,255)">
Section :=A0</div><div style=3D"color:rgb(51,51,255)">CurrentText:</div><di=
v style=3D"color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,25=
5)">=A0_This section is non-normative._</div><div style=3D"color:rgb(51,51,=
255)"><br>
</div><div style=3D"color:rgb(51,51,255)">ReplacementText / Comment:</div><=
div style=3D"color:rgb(51,51,255)">=A0Some sections have been explicitly me=
ntioned with this text. Does that mean</div><div style=3D"color:rgb(51,51,2=
55)">
=A0other sections are normative text. Not clear though.</div><div style=3D"=
color:rgb(51,51,255)"><br></div><div style=3D"color:rgb(51,51,255)">-------=
----------------------------------------------</div><div style=3D"color:rgb=
(51,51,255)">
<br></div></font><font face=3D"&#39;courier new&#39;, monospace" color=3D"#=
3333ff"><font size=3D"1">Thanks,</font></font><div><font face=3D"&#39;couri=
er new&#39;, monospace" size=3D"1" color=3D"#3333ff">Nataraju A.B.</font></=
div><div>
<font face=3D"&#39;courier new&#39;, monospace" size=3D"1" color=3D"#3333ff=
"><a href=3D"mailto:nataraju.ab@gmail.com">nataraju.ab@gmail.com</a> / <a h=
ref=3D"mailto:nataraju.sip@gmail.com">nataraju.sip@gmail.com</a></font></di=
v><div>
<font face=3D"&#39;courier new&#39;, monospace" size=3D"1" color=3D"#3333ff=
">+91-98455-95744</font></div>

--bcaec554d9aae3688704bde2adcc--

From dworley@avaya.com  Tue Apr 17 09:57:00 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 17 Apr 2012 12:56:30 -0400
Thread-Topic: draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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Subject: [sipcore] draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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Here are some things that I think could be improved in sections 1 to 5
of the 07 draft.  They are listed in textual order.

Abstract

    In addition, this specification defines a value for the Privacy
    header field specific to the History-Info header field.

As in other places in the draft, mention is made that there *exists* a
related Privacy value without giving any hint of the *purpose* of the
value.  This seems to me to be needlessly obscure.  A better
description would be "This specification defines a value for the
Privacy header field that directs the anonymization of values in the
History-Info header field."

1. Introduction

    the receiving application to determine hints about how and why the

should be

    the receiving application to obtain information about how and why the

2. Conventions and Terminology

   process of a SIP entity changing a Uniform Resource Identifier (URI)
   in a request based on the rules for determining request targets as

This sentence does not specify *which* URI in the request is being
changed.  A better wording would be

   process of a SIP entity changing the request-URI [RFC3261, section 7.1]
   in a request based on the rules for determining request targets as

   The terms "target user" is used in this specification as the human
   user associated with particular AoR or AoRs (in case the human user
   has multiple alias).

The use of "alias" here is not particularly good.  A better phrase
would be "in case the human user has multiple AoRs".

3. Background

   Current network applications provide the ability for elements

should have added "Current network applications >>>for other
protocols<<< provide".

   2.  Email forwarding whereby the forwarded-to user obtains a
       "history" of who sent the email to whom and at what time

More apropos is the fact that most e-mail systems add "Received"
lines, giving a detailed trace of the path of the message from sender
to receiver, and if the Received headers include the "for" clause, the
translation of the original destination address to the ultimate
recipient's address can be tracked.  E.g.,

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   o  Capturing aliases and Globally Routable User Agent URIs (GRUUs)
      [RFC5627], which can be overwritten by a REGISTRAR or a "home

should be "registrar".

4.  Overview

   request to be changed by the same proxy/SIP Intermediary multiple

should be "intermediary".

   header field parameters, along with examples is provided in

reads better with a comma after "examples".

   In addition, this specification defines a priv-value for the Privacy
   header, "history", that applies to all the History-info header field
   entries in a Request or to a specific History-info header field hi-
   entry as described above.  Further detail is provided in
   Section 10.1.

Again, this description does not state that the priv-value requires
that History-Info values be anonymized.  Better would be

   In addition, this specification defines a priv-value for the
   Privacy header, "history", that >>>requires anonymization of<<< all
   the >>>History-Info<<< header field entries in a Request or to a
   specific History-info header field hi-entry as described above.
   Further detail is provided in Section 10.1.

And "History-info" needs to be capitalized as "History-Info".

5.  History-Info Header Field Protocol Structure

   these requests (ISSUER-req, see Appendix A).

It looks like this phrase is intended to be a reference to the
requirements listed in Appendix A.  In that case, it should use
brackets [...] rather than parentheses (...).  But since there are
only two such references (the other one is in section 13), it appears
that the references were incompletely removed.  Otherwise there would
be many more of them.

      parameter is a string of digits, separated by dots to indicate the

There is a long history of using the term "string of digits" in these
drafts, despite that it's not a correct way to look at the syntax.
The syntax is a sequence of positive integers separated by dots and
the text should say that:

      parameter is a sequence of positive integers, separated by dots
      to indicate the

      This results in a tree
      representation of the history of the request, with the lowest-
      level index reflecting a branch of the tree.

I'm not sure what is meant here.  It is true that the lowest-level
index reflects a branch of the tree, but so do all the higher-level
indexes.  I believe that what is intended is "leaf".

      This
      parameter contains either an "rc", "mp" or "np" header field
      parameter, which is interpreted as follows:

"contains" should be "is".

         "np": The hi-targeted-to-URI represents that there was no
         change in Request-URI.  This would apply for example when a
         proxy merely forwards a request to a next hop proxy and loose
         routing is used.

Unlike the previous two cases, this case does not describe the
function of the value of the np parameter.

   o  Privacy: An optional parameter for History-Info, reflected in the
      History-Info header field values by including the Privacy Header
      [RFC3323] included in the hi- targeted-to-uri or by adding the
      Privacy header field to the request.  The latter case indicates
      that the History-Info entries for the domain MUST be anonymized
      prior to forwarding, whereas the use of the Privacy header field
      included in the hi-targeted-to-uri means that a specific hi-entry
      MUST be anonymized.

The phrase "the domain" is completely undefined here.  I suspect you
mean something like "all History-Info entries whose hi-targeted-to-uri
has the same domain as the domain for which the SIP entity processing
the message is responsible".

   Note that the backslash and CRLF between the fields in
   the examples below are for readability purposes only.

Was "lines" meant instead of "fields"?

   Note that the backslash, CRLF and whitespace between the lines in
   the examples below are inserted for readability purposes only.
   (But History-Info can be broken into multiple lines due to the SWS
   that is part of HCOLON, COMMA and SEMI, and there can be multiple
   History-Info header fields due to the rule of section 7.3
   [RFC3261].)

   History-Info: <sip:UserA@ims.example.com>;index=3D1;foo=3Dbar

   History-Info: <sip:UserA@ims.example.com?Reason=3DSIP%3B\
                 cause%3D302>;index=3D1.1,\
                 <sip:UserB@example.com?Privacy=3Dhistory&Reason=3DSIP%3B\
                 cause%3D486>;index=3D1.2;mp=3D1.1,\
                 <sip:45432@192.168.0.3>;index=3D1.3;rc=3D1.2

5.1.  History-Info Header Field Example Scenario

   One important thing illustrated by this call flow is that without
   History-Info, Bob would "lose" the target information, including any
   parameters in the request URI.

It seems to me that "the target information" is ambiguous -- the
target information of which request would be lost when?  You want to
say something like "the initial request-URI" or "the original target
information".

   Note that an hi-
   entry is not included for the fork to sip:bob@192.0.2.7 since there
   was no response at the time the 200 OK is sent to Alice.

This would be a lot clearer if it was more detailed:

   Note that in the 200 response to Alice, an hi-entry is not included
   for the fork to sip:bob@192.0.2.7 (index 1.1.1) since
   biloxi.example.com had not received a response from that fork at
   the time it sent the 200 OK that ultimately reached Alice.

   The formatting in this scenario is for visual purposes; thus,
   backslash and CRLF are used between the fields for readability.

Although in fact there are no backslash-line breaks in the example.
If there were, it might be needed to mention the included whitespace
as in section 5.

   |                |                | Proxy Cancels INVITE      |
   |                |                |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|

Might be better illustrated by showing it as a CANCEL request:

   |                |                | CANCEL         |          |
   |                |                |-------------------------->|

Dale

From ibc@aliax.net  Tue Apr 17 10:25:42 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 17 Apr 2012 19:25:17 +0200
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To: "Nataraju A.B" <nataraju.sip@gmail.com>
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Cc: vpascual@acmepacket.com, jmillan@aliax.net, sipcore@ietf.org
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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2012/4/17 Nataraju A.B <nataraju.sip@gmail.com>:
> Hi Authros,
>
> Please find some comments on the draft
> "draft-ibc-sipcore-sip-websocket-02.txt"


Hi Nataraju, thanks for your comments. We will address them. Some
replies inline:



> -----------------------------------------------------
> Section : =C2=A08.1. =C2=A0Registration
> CurrentText:
> =C2=A0 =C2=A0F1 REGISTER =C2=A0Alice -> proxy.atlanta.com (transport WSS)
>
> =C2=A0 =C2=A0REGISTER sip:proxy.atlanta.com SIP/2.0
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
>
> =C2=A0 =C2=A0F2 200 OK =C2=A0proxy.atlanta.com -> Alice (transport WSS)
>
> =C2=A0 =C2=A0SIP/2.0 200 OK
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
>
> ReplacementText / Comment:
>
> If I understand right,
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 Websocket connection is mapped on to S=
IP URI =C2=A0--> HTTP
> =C2=A0 =C2=A0Secure Websocket connection is mapped on to SIPS URI --> HTT=
PS
>
> HTTP --> WS --> SIP URI --> port 80
>
> HTTPS --> WSS --> SIPS URI --> port 443
>
> Hence in the following messages "WSS" shall be replaced with "WS"
>
> =C2=A0 =C2=A0F1 REGISTER =C2=A0Alice -> proxy.atlanta.com (transport WSS)
>
> =C2=A0 =C2=A0REGISTER sip:proxy.atlanta.com SIP/2.0
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
>
> WSS --> WS
>
> =C2=A0 =C2=A0F2 200 OK =C2=A0proxy.atlanta.com -> Alice (transport WSS)
>
> =C2=A0 =C2=A0SIP/2.0 200 OK
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
>
> WSS --> WS


Usage of a secure transport (like TLS-TCP or WSS) does not mandate
usage of SIPS schema. This is clearly explained in:

  http://tools.ietf.org/html/rfc5630#section-3.1.3



> For example, REGISTER request looks like this (with WSS)...
>
> =C2=A0 =C2=A0REGISTER sips:proxy.atlanta.com SIP/2.0
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> =C2=A0 =C2=A0From: sips:alice@atlanta.com;tag=3D65bnmj.34asd
> =C2=A0 =C2=A0To: sips:alice@atlanta.com
> =C2=A0 =C2=A0Call-ID: aiuy7k9njasd
> =C2=A0 =C2=A0CSeq: 1 REGISTER
> =C2=A0 =C2=A0Max-Forwards: 70
> =C2=A0 =C2=A0Supported: path, outbound, gruu
> =C2=A0 =C2=A0Contact: <sips:alice@df7jal23ls0d.invalid;transport=3Dws>
> =C2=A0 =C2=A0 =C2=A0;reg-id=3D1
> =C2=A0 =C2=A0 =C2=A0;+sip.instance=3D"<urn:uuid:f81-7dec-14a06cf1>"
>
>
> =C2=A0 =C2=A0changes:
> =C2=A0 =C2=A0sip --> sips in RURI, From, TO, contact headers

As stated above this is not needed. WSS (or TLS-TCP) can be used
without using SIPS schema in SIP requests. That is a "best-effort TLS
/ WSS". Also using Outbound RFC 5626 helps here since the Outbound
proxy (or the server the UA is connected to via TLS / WSS) can reuse
the existing connection for incoming requests. This is also explained
in http://tools.ietf.org/html/rfc5630#section-3.1.3.

In summary: using a secure connection does not require using SIPS
schema in the requests sent over it.




> -----------------------------------------------------
> Section : 8.2. =C2=A0INVITE dialog through a proxy
> CurrentText:
>
> =C2=A0 =C2=A0F1 INVITE =C2=A0Alice -> proxy.atlanta.com (transport WSS)
>
> =C2=A0 =C2=A0INVITE sip:bob@atlanta.com SIP/2.0
> =C2=A0 =C2=A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bK56sdas=
ks
> =C2=A0 =C2=A0From: sip:alice@atlanta.com;tag=3Dasdyka899
> =C2=A0 =C2=A0To: sip:bob@atlanta.com
> =C2=A0 =C2=A0Call-ID: asidkj3ss
> =C2=A0 =C2=A0CSeq: 1 INVITE
> =C2=A0 =C2=A0Max-Forwards: 70
> =C2=A0 =C2=A0Supported: path, outbound, gruu
> =C2=A0 =C2=A0Route: <sip:proxy.atlanta.com:443;transport=3Dws;lr>
> =C2=A0 =C2=A0Contact: <sip:alice@atlanta.com
> =C2=A0 =C2=A0 ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob>"
> =C2=A0 =C2=A0Content-Type: application/sdp
>
>
> ReplacementText / Comment:
>
> =C2=A0 =C2=A0Extra closing double quote in contact header.

Typo, thanks :)


> =C2=A0 =C2=A0 =C2=A0 Contact: <sip:alice@atlanta.com
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob>
>
> =C2=A0 =C2=A0Route: <sip:proxy.atlanta.com:80;transport=3Dws;lr>
>
> =C2=A0 =C2=A0change :
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 443-->80 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =
=C2=A0 =C2=A0assuming=C2=A0HTTP=C2=A0<--=C2=A0WS <-- SIP URI
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 This comment is applicable for messages F1, F=
3, F4, F5, F6, F8, F9,
> F10,

Same comments as above :)

The example uses a secure WS connection (on top of TLS) but does not
use SIPS schema.


> =C2=A0 =C2=A0Changes
> WSS --> WS =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0assuming=C2=A0HTTP=C2=
=A0<--=C2=A0WS <-- SIP URI
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 This comment is applicable for messages F1, F=
2, F3, F4, F5, F6, F7,

Idem.




> -----------------------------------------------------
> Section : 1. =C2=A0Introduction
>
> CurrentText:
> Media transport is out of the scope of this document
>
> ReplacementText / Comment:
> =C2=A0 =C2=A0Any other draft / rfc which addresses this issue. it would i=
mprove the
> =C2=A0 =C2=A0readability of the document if we can is suggest reference f=
or
> completeness.
>
> -----------------------------------------------------

We would not like to cover it. In case of web browsers, WebRTC
specification is defining RTP/SRTP/ICE/TURN for web applications. But
a SIP client using a WebSocket stack running in, i.e, a smartphone,
could use usual RTP for media transport.

We don't want to focus the draft on "web browsers" use case.



> Section : 5.1. =C2=A0General
> CurrentText:
> =C2=A0 =C2=A0 =C2=A0 (typically web-based applications with an strict and=
 simple API
> =C2=A0 =C2=A0 =C2=A0 for receiving WebSocket messages). =C2=A0There is no=
 need to establish
>
>
> ReplacementText / Comment:
> =C2=A0 =C2=A0 =C2=A0 (typically web-based applications with **a** strict =
and simple API
> =C2=A0 =C2=A0 =C2=A0 for receiving WebSocket messages). =C2=A0There is no=
 need to establish

Typo :)




> Section :
> CurrentText:
> =C2=A0 =C2=A0In the absence of an explicit port and DNS SRV resource reco=
rds, the
> =C2=A0 =C2=A0default port for a SIP URI with "ws" transport parameter is =
80 in
> =C2=A0 =C2=A0case of SIP scheme and 443 in case of SIPS scheme.
>
>
> ReplacementText / Comment:
> =C2=A0 =C2=A0In the absence of an explicit port and DNS SRV resource reco=
rds, the
> =C2=A0 =C2=A0default port for a SIP URI with "ws" transport parameter is =
80 in
> =C2=A0 =C2=A0case of SIP scheme and 443 in case of SIPS scheme **, which =
is same as
> =C2=A0 =C2=A0that used for http and https protocols respectively**
>
> These ports have been used for the sake of backward compatibility with
> servers
> which does not support WS/WSS.
>
> -----------------------------------------------------
> Section :
> CurrentText:
>
> =C2=A0_This section is non-normative._
>
> ReplacementText / Comment:
> =C2=A0Some sections have been explicitly mentioned with this text. Does t=
hat mean
> =C2=A0other sections are normative text. Not clear though.

Basically yes. The style is the same as in RFC 6455
(http://tools.ietf.org/html/rfc6455) in which non normative sections
include the above line. Any suggestion to make it better is of course
welcome.




Thanks a lot for your review, we will address all the topics covered in it.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From nataraju.sip@gmail.com  Tue Apr 17 10:26:11 2012
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
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Content-Type: multipart/alternative; boundary=14dae9d67bd2cfc59104bde33b33
Subject: [sipcore] Call flow for Secure Web Socket creation --- draft-ibc-sipcore-sip-websocket-02.txt
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--14dae9d67bd2cfc59104bde33b33
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Hi All,

I have written a call flow for the secure web socket usage with TCP.

Please go through the same and share if any changes required.

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

8.3.  INVITE dialog through a proxy (Secure Web Socket, WSS)


   Alice    (SIP WSS)    proxy.atlanta.com    (SIP TLS)       Bob
   |                             |                             |
   |INVITE F1                    |                             |
   |---------------------------->|                             |
   |100 Trying F2                |                             |
   |<----------------------------|                             |
   |                             |INVITE F3                    |
   |                             |---------------------------->|
   |                             |200 OK F4                    |
   |                             |<----------------------------|
   |200 OK F5                    |                             |
   |<----------------------------|                             |
   |                             |                             |
   |ACK F6                       |                             |
   |---------------------------->|                             |
   |                             |ACK F7                       |
   |                             |---------------------------->|
   |                             |                             |
   |                    Both Way RTP Media                     |
   |<=========================================================>|
   |                             |                             |
   |                             |BYE F8                       |
   |                             |<----------------------------|
   |BYE F9                       |                             |
   |<----------------------------|                             |
   |200 OK F10                   |                             |
   |---------------------------->|                             |
   |                             |200 OK F11                   |
   |                             |---------------------------->|
   |                             |                             |

   In the same scenario Alice places a call to Bob's AoR.  The WebSocket
   SIP server at proxy.atlanta.com acts as a SIP proxy routing the
   INVITE to the UDP location of Bob over TLS, who answers the call and
   terminates it later.

   Message details (authentication and SDP bodies are omitted for
   simplicity):


   F1 INVITE  Alice -> proxy.atlanta.com (transport WSS)

   INVITE sips:bob@atlanta.com SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 70
   Supported: path, outbound, gruu
   Route: <sips:proxy.atlanta.com:443;transport=ws;lr>
   Contact: <sips:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob>
   Content-Type: application/sdp


   F2 100 Trying  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 100 Trying
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE


   F3 INVITE  proxy.atlanta.com -> Bob (transport TLS)

   INVITE sips:bob@203.0.113.22:5060 SIP/2.0
   Via: SIP/2.0/TLS proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sips:proxy.atlanta.com;transport=TCP;lr>,
     <sips:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 69
   Supported: path, outbound, gruu
   Contact: <sips:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob>
   Content-Type: application/sdp


   F4 200 OK  Bob -> proxy.atlanta.com (transport TLS)

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS proxy.atlanta.com;branch=z9hG4bKhjhjqw32c
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sips:proxy.atlanta.com;transport=TCP;lr>,
     <sips:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 69
   Contact: <sips:bob@203.0.113.22:5060;transport=TCP>
   Content-Type: application/sdp


   F5 200 OK  proxy.atlanta.com -> Alice (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK56sdasks
   Record-Route: <sips:proxy.atlanta.com;transport=TCP;lr>,
     <sips:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sip:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 INVITE
   Max-Forwards: 69
   Contact: <sips:bob@203.0.113.22:5060;transport=TCP>
   Content-Type: application/sdp


   F6 ACK  Alice -> proxy.atlanta.com (transport WSS)

   ACK sips:bob@203.0.113.22:5060;transport=TCP SIP/2.0
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
   Route: <sips:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>,
     <sips:proxy.atlanta.com;transport=TCP;lr>,
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 ACK
   Max-Forwards: 70


   F7 ACK  proxy.atlanta.com -> Bob (transport TLS)

   ACK sips:bob@203.0.113.22:5060;transport=TCP SIP/2.0
   Via: SIP/2.0/TLS proxy.atlanta.com;branch=z9hG4bKhwpoc80zzx
   Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKhgqqp090
   From: sips:alice@atlanta.com;tag=asdyka899
   To: sips:bob@atlanta.com;tag=bmqkjhsd
   Call-ID: asidkj3ss
   CSeq: 1 ACK
   Max-Forwards: 69


   F8 BYE  Bob -> proxy.atlanta.com (transport TLS)

   BYE sips:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
   Via: SIP/2.0/TLS 203.0.113.22;branch=z9hG4bKbiuiansd001
   Route: <sips:proxy.atlanta.com;transport=TCP;lr>,
     <sips:h7kjh12s@proxy.atlanta.com:443;transport=ws;lr>
   From: sips:bob@atlanta.com;tag=bmqkjhsd
   To: sips:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE
   Max-Forwards: 70


   F9 BYE  proxy.atlanta.com -> Alice (transport WSS)

   BYE sips:alice@atlanta.com;gr=urn:uuid:f81-7dec-14a06cf1;ob SIP/2.0
   Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
   Via: SIP/2.0/TLS 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sips:bob@atlanta.com;tag=bmqkjhsd
   To: sips:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE
   Max-Forwards: 69


   F10 200 OK  Alice -> proxy.atlanta.com (transport WSS)

   SIP/2.0 200 OK
   Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=z9hG4bKmma01m3r5
   Via: SIP/2.0/TLS 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sips:bob@atlanta.com;tag=bmqkjhsd
   To: sips:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE


   F11 200 OK  proxy.atlanta.com -> Bob (transport TLS)

   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 203.0.113.22;branch=z9hG4bKbiuiansd001
   From: sips:bob@atlanta.com;tag=bmqkjhsd
   To: sips:alice@atlanta.com;tag=asdyka899
   Call-ID: asidkj3ss
   CSeq: 1201 BYE


Thanks,
Nataraju A.B.
nataraju.ab@gamil.com / nataraju.sip@gmail.com
+91-98455-95744

--14dae9d67bd2cfc59104bde33b33
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">Hi All,</fo=
nt></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><b=
r></font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D=
"4">I have written a call flow for the secure web socket usage with TCP.=A0=
</font></div>
<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">Pleas=
e go through the same and share if any changes required.</font></div><div><=
font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>
</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">---------------------------------------------------------------------</fo=
nt></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><b=
r></font></div>
<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">8.3. =A0INV=
ITE dialog through a proxy (Secure Web Socket, WSS)</font></div><div><font =
face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font></div><div>=
<font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0Alice =A0 =A0(SIP WSS) =A0 =A0<a href=3D"http://proxy.atlanta.com"=
 target=3D"_blank">proxy.atlanta.com</a> =A0 =A0(SIP TLS) =A0 =A0 =A0 Bob</=
font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=
=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|INV=
ITE F1 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier =
new&#39;, monospace" size=3D"4">=A0 =A0|----------------------------&gt;| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|100=
 Trying F2 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier new&=
#39;, monospace" size=3D"4">=A0 =A0|&lt;----------------------------| =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |INVITE F3 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|</font></div><div><font face=3D"&#39;courie=
r new&#39;, monospace" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 |----------------------------&gt;|</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |200 OK F4 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0|</font></div><div><font face=3D"&#39;courie=
r new&#39;, monospace" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 |&lt;----------------------------|</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|200=
 OK F5 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier =
new&#39;, monospace" size=3D"4">=A0 =A0|&lt;----------------------------| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#=
39;courier new&#39;, monospace" size=3D"4">=A0 =A0|ACK F6 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|---=
-------------------------&gt;| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier new&#39;, monospa=
ce" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |ACK F7 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |------------------=
----------&gt;|</font></div><div><font face=3D"&#39;courier new&#39;, monos=
pace" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 | =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font><=
/div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Both Way RTP Media =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier new&#=
39;, monospace" size=3D"4">=A0 =A0|&lt;=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D&gt;|</font></d=
iv>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div><div><font face=3D"&#=
39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |BYE F8 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |&lt;--------------=
--------------|</font></div><div><font face=3D"&#39;courier new&#39;, monos=
pace" size=3D"4">=A0 =A0|BYE F9 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 | =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|&lt=
;----------------------------| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier new&#39;, monospa=
ce" size=3D"4">=A0 =A0|200 OK F10 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 | =A0=
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0|---=
-------------------------&gt;| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 |</font></div><div><font face=3D"&#39;courier new&#39;, monospa=
ce" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0=
 =A0 |200 OK F11 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0| =
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |------------------=
----------&gt;|</font></div><div><font face=3D"&#39;courier new&#39;, monos=
pace" size=3D"4">=A0 =A0| =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 | =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 |</font><=
/div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0In the same scenario Alice places a call to Bob&#39;s AoR. =A0The WebSoc=
ket</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0SIP =
server at <a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy.atla=
nta.com</a> acts as a SIP proxy routing the</font></div><div><font face=3D"=
&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0INVITE to the UDP locat=
ion of Bob over TLS, who answers the call and</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0term=
inates it later.</font></div><div><font face=3D"&#39;courier new&#39;, mono=
space" size=3D"4"><br></font></div><div><font face=3D"&#39;courier new&#39;=
, monospace" size=3D"4">=A0 =A0Message details (authentication and SDP bodi=
es are omitted for</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0simp=
licity):</font></div><div><font face=3D"&#39;courier new&#39;, monospace" s=
ize=3D"4"><br></font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0F1 INVITE =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com" targ=
et=3D"_blank">proxy.atlanta.com</a> (transport WSS)</font></div><div><font =
face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0INVITE <a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank"=
>sips:bob@atlanta.com</a> SIP/2.0</font></div><div><font face=3D"&#39;couri=
er new&#39;, monospace" size=3D"4">=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.inv=
alid;branch=3Dz9hG4bK56sdasks</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@=
atlanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Abob@at=
lanta.com" target=3D"_blank">sips:bob@atlanta.com</a></font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1 INVITE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4">=A0 =A0Max-Forwards: 70</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Supp=
orted: path, outbound, gruu</font></div><div><font face=3D"&#39;courier new=
&#39;, monospace" size=3D"4">=A0 =A0Route: &lt;sips:proxy.atlanta.com:443;t=
ransport=3Dws;lr&gt;</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Cont=
act: &lt;<a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips=
:alice@atlanta.com</a>;gr=3Durn:uuid:f81-7dec-14a06cf1;ob&gt;</font></div><=
div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Conte=
nt-Type: application/sdp</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br><=
/font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"=
>=A0 =A0F2 100 Trying =A0<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> -&gt; Alice (transport WSS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0SIP/2.0 100 Trying</font></div><div><font face=3D"&#39;courier new&#39;,=
 monospace" size=3D"4">=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=
=3Dz9hG4bK56sdasks</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@=
atlanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Abob@at=
lanta.com" target=3D"_blank">sips:bob@atlanta.com</a></font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1 INVITE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
"><br></font></div><div><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"4">=A0 =A0F3 INVITE =A0<a href=3D"http://proxy.atlanta.com" target=3D"=
_blank">proxy.atlanta.com</a> -&gt; Bob (transport TLS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0INVITE <a href=3D"http://sips:bob@203.0.113.22:5060" target=3D"_blank">s=
ips:bob@203.0.113.22:5060</a> SIP/2.0</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/TLS <a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy.a=
tlanta.com</a>;branch=3Dz9hG4bKhjhjqw32c</font></div><div><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"4">=A0 =A0Via: SIP/2.0/WSS df7jal23l=
s0d.invalid;branch=3Dz9hG4bK56sdasks</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Reco=
rd-Route: &lt;sips:<a href=3D"http://proxy.atlanta.com" target=3D"_blank">p=
roxy.atlanta.com</a>;transport=3DTCP;lr&gt;,</font></div><div><font face=3D=
"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0 =A0&lt;sips:h7kjh12s@=
proxy.atlanta.com:443;transport=3Dws;lr&gt;</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@=
atlanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Abob@at=
lanta.com" target=3D"_blank">sips:bob@atlanta.com</a></font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1 INVITE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4">=A0 =A0Max-Forwards: 69</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Supp=
orted: path, outbound, gruu</font></div><div><font face=3D"&#39;courier new=
&#39;, monospace" size=3D"4">=A0 =A0Contact: &lt;<a href=3D"mailto:sips%3Aa=
lice@atlanta.com" target=3D"_blank">sips:alice@atlanta.com</a>;gr=3Durn:uui=
d:f81-7dec-14a06cf1;ob&gt;</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Cont=
ent-Type: application/sdp</font></div><div><font face=3D"&#39;courier new&#=
39;, monospace" size=3D"4"><br></font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0F4 200 OK =A0Bob -&gt; <a href=3D"http://proxy.atlanta.com" target=
=3D"_blank">proxy.atlanta.com</a> (transport TLS)</font></div><div><font fa=
ce=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0SIP/2.0 200 OK</font></div><div><font face=3D"&#39;courier new&#39=
;, monospace" size=3D"4">=A0 =A0Via: SIP/2.0/TLS <a href=3D"http://proxy.at=
lanta.com" target=3D"_blank">proxy.atlanta.com</a>;branch=3Dz9hG4bKhjhjqw32=
c</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bK56sdasks</font></div><div=
><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Record-R=
oute: &lt;sips:<a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy=
.atlanta.com</a>;transport=3DTCP;lr&gt;,</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0 =A0=
&lt;sips:h7kjh12s@proxy.atlanta.com:443;transport=3Dws;lr&gt;</font></div><=
div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From:=
 <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@a=
tlanta.com</a>;tag=3Dasdyka899</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0To: =
<a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atlant=
a.com</a>;tag=3Dbmqkjhsd</font></div><div><font face=3D"&#39;courier new&#3=
9;, monospace" size=3D"4">=A0 =A0Call-ID: asidkj3ss</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0CSeq=
: 1 INVITE</font></div><div><font face=3D"&#39;courier new&#39;, monospace"=
 size=3D"4">=A0 =A0Max-Forwards: 69</font></div><div><font face=3D"&#39;cou=
rier new&#39;, monospace" size=3D"4">=A0 =A0Contact: &lt;sips:bob@203.0.113=
.22:5060;transport=3DTCP&gt;</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Cont=
ent-Type: application/sdp</font></div><div><font face=3D"&#39;courier new&#=
39;, monospace" size=3D"4"><br></font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0F5 200 OK =A0<a href=3D"http://proxy.atlanta.com" target=3D"_blank=
">proxy.atlanta.com</a> -&gt; Alice (transport WSS)</font></div><div><font =
face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0SIP/2.0 200 OK</font></div><div><font face=3D"&#39;courier new&#39=
;, monospace" size=3D"4">=A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branc=
h=3Dz9hG4bK56sdasks</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Reco=
rd-Route: &lt;sips:<a href=3D"http://proxy.atlanta.com" target=3D"_blank">p=
roxy.atlanta.com</a>;transport=3DTCP;lr&gt;,</font></div><div><font face=3D=
"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0 =A0&lt;sips:h7kjh12s@=
proxy.atlanta.com:443;transport=3Dws;lr&gt;</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@=
atlanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sip%3Abob@atl=
anta.com" target=3D"_blank">sip:bob@atlanta.com</a>;tag=3Dbmqkjhsd</font></=
div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1 INVITE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4">=A0 =A0Max-Forwards: 69</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Cont=
act: &lt;sips:bob@203.0.113.22:5060;transport=3DTCP&gt;</font></div><div><f=
ont face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Content-Typ=
e: application/sdp</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br><=
/font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"=
>=A0 =A0F6 ACK =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com" target=
=3D"_blank">proxy.atlanta.com</a> (transport WSS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0ACK sips:bob@203.0.113.22:5060;transport=3DTCP SIP/2.0</font></div><div>=
<font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via: SIP/=
2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKhgqqp090</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Rout=
e: &lt;sips:h7kjh12s@proxy.atlanta.com:443;transport=3Dws;lr&gt;,</font></d=
iv><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0 =
=A0&lt;sips:<a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy.at=
lanta.com</a>;transport=3DTCP;lr&gt;,</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@=
atlanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier=
 new&#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Abob@at=
lanta.com" target=3D"_blank">sips:bob@atlanta.com</a>;tag=3Dbmqkjhsd</font>=
</div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1 ACK</font></div><div><font face=3D"&#39;cour=
ier new&#39;, monospace" size=3D"4">=A0 =A0Max-Forwards: 70</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br><=
/font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"=
>=A0 =A0F7 ACK =A0<a href=3D"http://proxy.atlanta.com" target=3D"_blank">pr=
oxy.atlanta.com</a> -&gt; Bob (transport TLS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0ACK sips:bob@203.0.113.22:5060;transport=3DTCP SIP/2.0</font></div><div>=
<font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via: SIP/=
2.0/TLS <a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy.atlant=
a.com</a>;branch=3Dz9hG4bKhwpoc80zzx</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKhgqqp090</font></div><div=
><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From: <a=
 href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@atla=
nta.com</a>;tag=3Dasdyka899</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0To: =
<a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atlant=
a.com</a>;tag=3Dbmqkjhsd</font></div><div><font face=3D"&#39;courier new&#3=
9;, monospace" size=3D"4">=A0 =A0Call-ID: asidkj3ss</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0CSeq=
: 1 ACK</font></div><div><font face=3D"&#39;courier new&#39;, monospace" si=
ze=3D"4">=A0 =A0Max-Forwards: 69</font></div><div><font face=3D"&#39;courie=
r new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
"><br></font></div><div><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"4">=A0 =A0F8 BYE =A0Bob -&gt; <a href=3D"http://proxy.atlanta.com" tar=
get=3D"_blank">proxy.atlanta.com</a> (transport TLS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0BYE <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:a=
lice@atlanta.com</a>;gr=3Durn:uuid:f81-7dec-14a06cf1;ob SIP/2.0</font></div=
>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/TLS 203.0.113.22;branch=3Dz9hG4bKbiuiansd001</font></div><div><fon=
t face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Route: &lt;si=
ps:<a href=3D"http://proxy.atlanta.com" target=3D"_blank">proxy.atlanta.com=
</a>;transport=3DTCP;lr&gt;,</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0 =A0=
&lt;sips:h7kjh12s@proxy.atlanta.com:443;transport=3Dws;lr&gt;</font></div><=
div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From:=
 <a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atlan=
ta.com</a>;tag=3Dbmqkjhsd</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0To: =
<a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@at=
lanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"4">=A0 =A0Call-ID: asidkj3ss</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0CSeq=
: 1201 BYE</font></div><div><font face=3D"&#39;courier new&#39;, monospace"=
 size=3D"4">=A0 =A0Max-Forwards: 70</font></div><div><font face=3D"&#39;cou=
rier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
"><br></font></div><div><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"4">=A0 =A0F9 BYE =A0<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> -&gt; Alice (transport WSS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0BYE <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:a=
lice@atlanta.com</a>;gr=3Durn:uuid:f81-7dec-14a06cf1;ob SIP/2.0</font></div=
>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/WSS proxy.atlanta.com:443;branch=3Dz9hG4bKmma01m3r5</font></div><d=
iv><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via: S=
IP/2.0/TLS 203.0.113.22;branch=3Dz9hG4bKbiuiansd001</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atla=
nta.com</a>;tag=3Dbmqkjhsd</font></div><div><font face=3D"&#39;courier new&=
#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Aalice@atlan=
ta.com" target=3D"_blank">sips:alice@atlanta.com</a>;tag=3Dasdyka899</font>=
</div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1201 BYE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4">=A0 =A0Max-Forwards: 69</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br><=
/font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"=
>=A0 =A0F10 200 OK =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com" targ=
et=3D"_blank">proxy.atlanta.com</a> (transport WSS)</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4"><br></font>=
</div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =
=A0SIP/2.0 200 OK</font></div><div><font face=3D"&#39;courier new&#39;, mon=
ospace" size=3D"4">=A0 =A0Via: SIP/2.0/WSS proxy.atlanta.com:443;branch=3Dz=
9hG4bKmma01m3r5</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Via:=
 SIP/2.0/TLS 203.0.113.22;branch=3Dz9hG4bKbiuiansd001</font></div><div><fon=
t face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From: <a href=
=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atlanta.com</=
a>;tag=3Dbmqkjhsd</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0To: =
<a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blank">sips:alice@at=
lanta.com</a>;tag=3Dasdyka899</font></div><div><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"4">=A0 =A0Call-ID: asidkj3ss</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0CSeq=
: 1201 BYE</font></div><div><font face=3D"&#39;courier new&#39;, monospace"=
 size=3D"4"><br></font></div><div><font face=3D"&#39;courier new&#39;, mono=
space" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0F11 200 OK =A0<a href=3D"http://proxy.atlanta.com" target=3D"_blan=
k">proxy.atlanta.com</a> -&gt; Bob (transport TLS)</font></div><div><font f=
ace=3D"&#39;courier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
">=A0 =A0SIP/2.0 200 OK</font></div><div><font face=3D"&#39;courier new&#39=
;, monospace" size=3D"4">=A0 =A0Via: SIP/2.0/TLS 203.0.113.22;branch=3Dz9hG=
4bKbiuiansd001</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0From=
: <a href=3D"mailto:sips%3Abob@atlanta.com" target=3D"_blank">sips:bob@atla=
nta.com</a>;tag=3Dbmqkjhsd</font></div><div><font face=3D"&#39;courier new&=
#39;, monospace" size=3D"4">=A0 =A0To: <a href=3D"mailto:sips%3Aalice@atlan=
ta.com" target=3D"_blank">sips:alice@atlanta.com</a>;tag=3Dasdyka899</font>=
</div>


<div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">=A0 =A0Call=
-ID: asidkj3ss</font></div><div><font face=3D"&#39;courier new&#39;, monosp=
ace" size=3D"4">=A0 =A0CSeq: 1201 BYE</font></div><div><font face=3D"&#39;c=
ourier new&#39;, monospace" size=3D"4"><br>


</font></div><div><font face=3D"&#39;courier new&#39;, monospace" size=3D"4=
"><br></font></div><font face=3D"&#39;courier new&#39;, monospace" size=3D"=
4"><font color=3D"#000099">Thanks,</font></font><div><font color=3D"#000099=
"><font face=3D"&#39;courier new&#39;, monospace" size=3D"4">Nataraju A.B.<=
/font></font></div>
<div><font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace=
" size=3D"4"><a href=3D"mailto:nataraju.ab@gamil.com">nataraju.ab@gamil.com=
</a> / <a href=3D"mailto:nataraju.sip@gmail.com">nataraju.sip@gmail.com</a>=
</font></font></div>
<div><font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace=
" size=3D"4">+91-98455-95744</font></font></div>

<br>

--14dae9d67bd2cfc59104bde33b33--

From shida@ntt-at.com  Tue Apr 17 17:08:06 2012
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From: Shida Schubert <shida@ntt-at.com>
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Date: Wed, 18 Apr 2012 09:07:58 +0900
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To: "Worley, Dale R (Dale)" <dworley@avaya.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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Dale;

 Many thanks for your comments, my response inline.

On Apr 18, 2012, at 1:56 AM, Worley, Dale R (Dale) wrote:

> Here are some things that I think could be improved in sections 1 to 5
> of the 07 draft.  They are listed in textual order.
>=20
> Abstract
>=20
>    In addition, this specification defines a value for the Privacy
>    header field specific to the History-Info header field.
>=20
> As in other places in the draft, mention is made that there *exists* a
> related Privacy value without giving any hint of the *purpose* of the
> value.  This seems to me to be needlessly obscure.  A better
> description would be "This specification defines a value for the
> Privacy header field that directs the anonymization of values in the
> History-Info header field."

 Ok

>=20
> 1. Introduction
>=20
>    the receiving application to determine hints about how and why the
>=20
> should be
>=20
>    the receiving application to obtain information about how and why =
the

DONE

>=20
> 2. Conventions and Terminology
>=20
>   process of a SIP entity changing a Uniform Resource Identifier (URI)
>   in a request based on the rules for determining request targets as
>=20
> This sentence does not specify *which* URI in the request is being
> changed.  A better wording would be
>=20
>   process of a SIP entity changing the request-URI [RFC3261, section =
7.1]
>   in a request based on the rules for determining request targets as
>=20
>   The terms "target user" is used in this specification as the human
>   user associated with particular AoR or AoRs (in case the human user
>   has multiple alias).
>=20
> The use of "alias" here is not particularly good.  A better phrase
> would be "in case the human user has multiple AoRs".

DONE

>=20
> 3. Background
>=20
>   Current network applications provide the ability for elements
>=20
> should have added "Current network applications >>>for other
> protocols<<< provide".

 DONE

>=20
>   2.  Email forwarding whereby the forwarded-to user obtains a
>       "history" of who sent the email to whom and at what time
>=20
> More apropos is the fact that most e-mail systems add "Received"
> lines, giving a detailed trace of the path of the message from sender
> to receiver, and if the Received headers include the "for" clause, the
> translation of the original destination address to the ultimate
> recipient's address can be tracked.  E.g.,
>=20
>     Received: from eye.ariadne.com (eye.ariadne.com [127.0.0.1])
> 	     by eye.ariadne.com (8.14.4/8.14.4) with ESMTP id =
q3H8e3vI026332
> 	     for <worley@localhost>; Tue, 17 Apr 2012 04:40:03 -0400
>     Received: from mail.comcast.net [76.96.58.11]
> 	     by eye.ariadne.com with POP3 (fetchmail-6.3.20)
> 	     for <worley@localhost> (single-drop); Tue, 17 Apr 2012 =
04:40:03 -0400 (EDT)
>     Received: from imta34.emeryville.ca.mail.comcast.net (LHLO
>      imta34.emeryville.ca.mail.comcast.net) (76.96.28.168) by
>      sz0045.wc.mail.comcast.net with LMTP; Tue, 17 Apr 2012 08:26:05 =
+0000 (UTC)
>     Received: from VIMPORTPART.COM ([78.46.253.246])
> 	     by imta34.emeryville.ca.mail.comcast.net with comcast
> 	     id ywS21i00o5KkC6e0awS2oh; Tue, 17 Apr 2012 08:26:03 +0000
>     Received: by VIMPORTPART.COM (Postfix, from userid 0)
> 	     id 21D99831D7; Tue, 17 Apr 2012 10:26:02 +0200 (CEST)

 Tried to incorporate it.=20

>=20
>   o  Capturing aliases and Globally Routable User Agent URIs (GRUUs)
>      [RFC5627], which can be overwritten by a REGISTRAR or a "home
>=20
> should be "registrar".

DONE

>=20
> 4.  Overview
>=20
>   request to be changed by the same proxy/SIP Intermediary multiple
>=20
> should be "intermediary".

DONE

>=20
>   header field parameters, along with examples is provided in
>=20
> reads better with a comma after "examples".

 DONE

>=20
>   In addition, this specification defines a priv-value for the Privacy
>   header, "history", that applies to all the History-info header field
>   entries in a Request or to a specific History-info header field hi-
>   entry as described above.  Further detail is provided in
>   Section 10.1.
>=20
> Again, this description does not state that the priv-value requires
> that History-Info values be anonymized.  Better would be
>=20
>   In addition, this specification defines a priv-value for the
>   Privacy header, "history", that >>>requires anonymization of<<< all
>   the >>>History-Info<<< header field entries in a Request or to a
>   specific History-info header field hi-entry as described above.
>   Further detail is provided in Section 10.1.

DONE

>=20
> And "History-info" needs to be capitalized as "History-Info".

DONE

>=20
> 5.  History-Info Header Field Protocol Structure
>=20
>   these requests (ISSUER-req, see Appendix A).
>=20
> It looks like this phrase is intended to be a reference to the
> requirements listed in Appendix A.  In that case, it should use
> brackets [...] rather than parentheses (...).  But since there are
> only two such references (the other one is in section 13), it appears
> that the references were incompletely removed.  Otherwise there would
> be many more of them.

REMOVED.

>=20
>      parameter is a string of digits, separated by dots to indicate =
the
>=20
> There is a long history of using the term "string of digits" in these
> drafts, despite that it's not a correct way to look at the syntax.
> The syntax is a sequence of positive integers separated by dots and
> the text should say that:
>=20
>      parameter is a sequence of positive integers, separated by dots
>      to indicate the
>=20
>      This results in a tree
>      representation of the history of the request, with the lowest-
>      level index reflecting a branch of the tree.
>=20
> I'm not sure what is meant here.  It is true that the lowest-level
> index reflects a branch of the tree, but so do all the higher-level
> indexes.  I believe that what is intended is "leaf".

 FIXED

>=20
>      This
>      parameter contains either an "rc", "mp" or "np" header field
>      parameter, which is interpreted as follows:
>=20
> "contains" should be "is".

 FIXED

>=20
>         "np": The hi-targeted-to-URI represents that there was no
>         change in Request-URI.  This would apply for example when a
>         proxy merely forwards a request to a next hop proxy and loose
>         routing is used.
>=20
> Unlike the previous two cases, this case does not describe the
> function of the value of the np parameter.

 Any suggestion for rewording?

>=20
>   o  Privacy: An optional parameter for History-Info, reflected in the
>      History-Info header field values by including the Privacy Header
>      [RFC3323] included in the hi- targeted-to-uri or by adding the
>      Privacy header field to the request.  The latter case indicates
>      that the History-Info entries for the domain MUST be anonymized
>      prior to forwarding, whereas the use of the Privacy header field
>      included in the hi-targeted-to-uri means that a specific hi-entry
>      MUST be anonymized.
>=20
> The phrase "the domain" is completely undefined here.  I suspect you
> mean something like "all History-Info entries whose hi-targeted-to-uri
> has the same domain as the domain for which the SIP entity processing
> the message is responsible".

 Sounds good.=20

>=20
>   Note that the backslash and CRLF between the fields in
>   the examples below are for readability purposes only.
>=20
> Was "lines" meant instead of "fields"?
>=20
>   Note that the backslash, CRLF and whitespace between the lines in
>   the examples below are inserted for readability purposes only.
>   (But History-Info can be broken into multiple lines due to the SWS
>   that is part of HCOLON, COMMA and SEMI, and there can be multiple
>   History-Info header fields due to the rule of section 7.3
>   [RFC3261].)
>=20
>   History-Info: <sip:UserA@ims.example.com>;index=3D1;foo=3Dbar
>=20
>   History-Info: <sip:UserA@ims.example.com?Reason=3DSIP%3B\
>                 cause%3D302>;index=3D1.1,\
>                 <sip:UserB@example.com?Privacy=3Dhistory&Reason=3DSIP%3B=
\
>                 cause%3D486>;index=3D1.2;mp=3D1.1,\
>                 <sip:45432@192.168.0.3>;index=3D1.3;rc=3D1.2

 Will use this instead.

>=20
> 5.1.  History-Info Header Field Example Scenario
>=20
>   One important thing illustrated by this call flow is that without
>   History-Info, Bob would "lose" the target information, including any
>   parameters in the request URI.
>=20
> It seems to me that "the target information" is ambiguous -- the
> target information of which request would be lost when?  You want to
> say something like "the initial request-URI" or "the original target
> information".
>=20
>   Note that an hi-
>   entry is not included for the fork to sip:bob@192.0.2.7 since there
>   was no response at the time the 200 OK is sent to Alice.
>=20
> This would be a lot clearer if it was more detailed:
>=20
>   Note that in the 200 response to Alice, an hi-entry is not included
>   for the fork to sip:bob@192.0.2.7 (index 1.1.1) since
>   biloxi.example.com had not received a response from that fork at
>   the time it sent the 200 OK that ultimately reached Alice.
>=20
>   The formatting in this scenario is for visual purposes; thus,
>   backslash and CRLF are used between the fields for readability.
>=20
> Although in fact there are no backslash-line breaks in the example.
> If there were, it might be needed to mention the included whitespace
> as in section 5.

 Removed.

 Many Thanks & Regards
  Shida

>=20
>   |                |                | Proxy Cancels INVITE      |
>   |                |                |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D>|
>=20
> Might be better illustrated by showing it as a CANCEL request:
>=20
>   |                |                | CANCEL         |          |
>   |                |                |-------------------------->|
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Thread-Topic: Stateless or Transaction stateful proxy possible in Websocket transport? [RE: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Partha,

I also feel the same. The scenario is very similar to proxy interfacing
between UDP and TCP on different sides, where dialog stateful proxy
functionality is required. stateless / transaction stateful proxies would
not work properly when requests are sent over different protocols.

Thanks,
Nataraju A B

On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

> As per my current reading, The current draft assumes about dialog-statefu=
l
> proxy wherein all record-route header will be added whenever Websocket to
> other transport routing happens in the proxy and proxy will be involved i=
n
> all the transaction within the dialog. So, it is not possible to have
> stateless or transaction stateful proxy. It will be good in case this
> limitation is explicitly mentioned in the draft.
>
> Thanks
> Partha
>
> >-----Original Message-----
> >From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> >Behalf Of Adam Roach - SIPCORE Chair
> >Sent: Monday, April 16, 2012 11:33 PM
> >To: SIPCORE (Session Initiation Protocol Core) WG
> >Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
> >
> >[as chair]
> >
> >SIPCORE Working Group:
> >
> >This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
> >document to satisfy our recently added milestone to define a WebSockets
> >transport for the SIP protocol. Interested parties should voice their
> >support and/or concerns on the SIPCORE mailing list. The chairs plan to
> >make a determination of consensus on Friday, April 27th.
> >
> >Thank you. The draft announcement follows.
> >
> >/a
> >
> >-------- Original Message --------
> >Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-
> >websocket-02.txt
> >Date: Mon, 16 Apr 2012 18:49:38 +0200
> >From: I=F1aki Baz Castillo <ibc@aliax.net>
> >To: SIPCORE WG <sipcore@ietf.org>
> >
> >A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
> >successfully submitted by I=F1aki Baz Castillo and posted to the IETF
> >repository.
> >
> >Filename:        draft-ibc-sipcore-sip-websocket
> >Revision:        02
> >Title:           The WebSocket Protocol as a Transport for the Session
> >Initiation Protocol (SIP)
> >Creation date:   2012-04-16
> >WG ID:           Individual Submission
> >Number of pages: 20
> >
> >Abstract:
> >   The WebSocket protocol enables two-way realtime communication between
> >   clients and servers.  This document specifies a new WebSocket sub-
> >   protocol as a reliable transport mechanism between SIP (Session
> >   Initiation Protocol) entities and enables usage of the SIP protocol
> >   in new scenarios.
> >
> >
> >- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
> >- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
> >
> >
> >This new version incorporates improvements and changes based on the
> >comments and reviews on the previous version, along with the consensus
> >got in IETF 83 Paris.
> >
> >
> >As usual, comments are welcome. Thanks a lot.
> >
> >
> >--
> >I=F1aki Baz Castillo
> ><ibc@aliax.net>
> >_______________________________________________
> >sipcore mailing list
> >sipcore@ietf.org
> >https://www.ietf.org/mailman/listinfo/sipcore
> >_______________________________________________
> >sipcore mailing list
> >sipcore@ietf.org
> >https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



--=20
Thanks,
Nataraju A.B.

--f46d040122a324f02704bdee0608
Content-Type: text/html; charset=ISO-8859-1
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Partha,=A0<div><br></div><div>I also feel the same. The scenario is very si=
milar to proxy interfacing between UDP and TCP on different sides, where di=
alog stateful proxy functionality is required. stateless / transaction stat=
eful proxies would not work properly when requests are sent over different =
protocols.<div>
<br></div><div>Thanks,</div><div>Nataraju A B<br><br><div class=3D"gmail_qu=
ote">On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusnet.com">pravindran@sonusnet=
.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">As per my current reading, The current draft=
 assumes about dialog-stateful proxy wherein all record-route header will b=
e added whenever Websocket to other transport routing happens in the proxy =
and proxy will be involved in all the transaction within the dialog. So, it=
 is not possible to have stateless or transaction stateful proxy. It will b=
e good in case this limitation is explicitly mentioned in the draft.<br>

<br>
Thanks<br>
Partha<br>
<br>
&gt;-----Original Message-----<br>
&gt;From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.=
org</a> [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces=
@ietf.org</a>] On<br>
&gt;Behalf Of Adam Roach - SIPCORE Chair<br>
&gt;Sent: Monday, April 16, 2012 11:33 PM<br>
&gt;To: SIPCORE (Session Initiation Protocol Core) WG<br>
&gt;Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-0=
2<br>
&gt;<br>
&gt;[as chair]<br>
&gt;<br>
&gt;SIPCORE Working Group:<br>
&gt;<br>
&gt;This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02<b=
r>
&gt;document to satisfy our recently added milestone to define a WebSockets=
<br>
&gt;transport for the SIP protocol. Interested parties should voice their<b=
r>
&gt;support and/or concerns on the SIPCORE mailing list. The chairs plan to=
<br>
&gt;make a determination of consensus on Friday, April 27th.<br>
&gt;<br>
&gt;Thank you. The draft announcement follows.<br>
&gt;<br>
&gt;/a<br>
&gt;<br>
&gt;-------- Original Message --------<br>
&gt;Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02.txt<br>
&gt;Date: Mon, 16 Apr 2012 18:49:38 +0200<br>
&gt;From: I=F1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net">ibc@ali=
ax.net</a>&gt;<br>
&gt;To: SIPCORE WG &lt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org=
</a>&gt;<br>
&gt;<br>
&gt;A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been<b=
r>
&gt;successfully submitted by I=F1aki Baz Castillo and posted to the IETF<b=
r>
&gt;repository.<br>
&gt;<br>
&gt;Filename: =A0 =A0 =A0 =A0draft-ibc-sipcore-sip-websocket<br>
&gt;Revision: =A0 =A0 =A0 =A002<br>
&gt;Title: =A0 =A0 =A0 =A0 =A0 The WebSocket Protocol as a Transport for th=
e Session<br>
&gt;Initiation Protocol (SIP)<br>
&gt;Creation date: =A0 2012-04-16<br>
&gt;WG ID: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
&gt;Number of pages: 20<br>
&gt;<br>
&gt;Abstract:<br>
&gt; =A0 The WebSocket protocol enables two-way realtime communication betw=
een<br>
&gt; =A0 clients and servers. =A0This document specifies a new WebSocket su=
b-<br>
&gt; =A0 protocol as a reliable transport mechanism between SIP (Session<br=
>
&gt; =A0 Initiation Protocol) entities and enables usage of the SIP protoco=
l<br>
&gt; =A0 in new scenarios.<br>
&gt;<br>
&gt;<br>
&gt;- <a href=3D"http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket=
-02" target=3D"_blank">http://tools.ietf.org/html/draft-ibc-sipcore-sip-web=
socket-02</a><br>
&gt;- <a href=3D"http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-0=
2.txt" target=3D"_blank">http://tools.ietf.org/id/draft-ibc-sipcore-sip-web=
socket-02.txt</a><br>
&gt;<br>
&gt;<br>
&gt;This new version incorporates improvements and changes based on the<br>
&gt;comments and reviews on the previous version, along with the consensus<=
br>
&gt;got in IETF 83 Paris.<br>
&gt;<br>
&gt;<br>
&gt;As usual, comments are welcome. Thanks a lot.<br>
&gt;<br>
&gt;<br>
&gt;--<br>
&gt;I=F1aki Baz Castillo<br>
&gt;&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font color=
=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>
</div></div>

--f46d040122a324f02704bdee0608--

From nataraju.sip@gmail.com  Tue Apr 17 23:46:05 2012
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
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Cc: vpascual@acmepacket.com, jmillan@aliax.net, sipcore@ietf.org
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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Hi Inaki,

Please find some more comments inline...


Please let me know, the reasoning behind single transport parameter and two
sent-protocol values.
sent-protocol : WS and WSS
transport =3D ws (only one new value)

This might be confusing in future...


To make the draft more readable, it may be a good idea to add transport
parameter for WSS as well.

transport         =3D  "UDP" / "TCP" / "TLS" / "SCTP" / "WS" */ "WSS"*
                     / other-transport



On Tue, Apr 17, 2012 at 10:55 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:

> 2012/4/17 Nataraju A.B <nataraju.sip@gmail.com>:
> > Hi Authros,
> >
> > Please find some comments on the draft
> > "draft-ibc-sipcore-sip-websocket-02.txt"
>
>
> Hi Nataraju, thanks for your comments. We will address them. Some
> replies inline:
>
>
>
> > -----------------------------------------------------
> > Section :  8.1.  Registration
> > CurrentText:
> >    F1 REGISTER  Alice -> proxy.atlanta.com (transport WSS)
> >
> >    REGISTER sip:proxy.atlanta.com SIP/2.0
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> >
> >    F2 200 OK  proxy.atlanta.com -> Alice (transport WSS)
> >
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> >
> > ReplacementText / Comment:
> >
> > If I understand right,
> >           Websocket connection is mapped on to SIP URI  --> HTTP
> >    Secure Websocket connection is mapped on to SIPS URI --> HTTPS
> >
> > HTTP --> WS --> SIP URI --> port 80
> >
> > HTTPS --> WSS --> SIPS URI --> port 443
> >
> > Hence in the following messages "WSS" shall be replaced with "WS"
> >
> >    F1 REGISTER  Alice -> proxy.atlanta.com (transport WSS)
> >
> >    REGISTER sip:proxy.atlanta.com SIP/2.0
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> >
> > WSS --> WS
> >
> >    F2 200 OK  proxy.atlanta.com -> Alice (transport WSS)
> >
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> >
> > WSS --> WS
>
>
> Usage of a secure transport (like TLS-TCP or WSS) does not mandate
> usage of SIPS schema. This is clearly explained in:
>
>  http://tools.ietf.org/html/rfc5630#section-3.1.3
>
> <rfc5630> sec 3.1.3 <http://tools.ietf.org/html/rfc5630#section-3.1.3>.

   If one wants to use "best-effort TLS" for SIP, one just needs to use
   a SIP URI, and send the request over TLS.

</rfc5630>

if we want to use SIP URI with any other secure communication mechanism
like WSS, might require
update to RFC5630. like
<rfc5630> sec 3.1.3 <http://tools.ietf.org/html/rfc5630#section-3.1.3>.

   If one wants to use "best-effort TLS/WSS" for SIP, one just needs to use
   a SIP URI, and send the request over TLS ** or any other secure
transport mechanism like WSS**

</rfc5630>

for these changes, you can add a new section "5.4 Updates to RFC 5630"

Also add a normative reference for rfc5630 in sec "12.2. Informative
References"

   [RFC5630]  Audet, F., "The Use of the SIPS URI Scheme in the Session
              Initiation Protocol (SIP)", RFC 5630, October 2009.



>
> > For example, REGISTER request looks like this (with WSS)...
> >
> >    REGISTER sips:proxy.atlanta.com SIP/2.0
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf
> >    From: sips:alice@atlanta.com;tag=3D65bnmj.34asd
> >    To: sips:alice@atlanta.com
> >    Call-ID: aiuy7k9njasd
> >    CSeq: 1 REGISTER
> >    Max-Forwards: 70
> >    Supported: path, outbound, gruu
> >    Contact: <sips:alice@df7jal23ls0d.invalid;transport=3Dws>
> >      ;reg-id=3D1
> >      ;+sip.instance=3D"<urn:uuid:f81-7dec-14a06cf1>"
> >
> >
> >    changes:
> >    sip --> sips in RURI, From, TO, contact headers
>
> As stated above this is not needed. WSS (or TLS-TCP) can be used
> without using SIPS schema in SIP requests. That is a "best-effort TLS
> / WSS". Also using Outbound RFC 5626 helps here since the Outbound
> proxy (or the server the UA is connected to via TLS / WSS) can reuse
> the existing connection for incoming requests. This is also explained
> in http://tools.ietf.org/html/rfc5630#section-3.1.3.
>
> In summary: using a secure connection does not require using SIPS
> schema in the requests sent over it.
>
>
>
>
> > -----------------------------------------------------
> > Section : 8.2.  INVITE dialog through a proxy
> > CurrentText:
> >
> >    F1 INVITE  Alice -> proxy.atlanta.com (transport WSS)
> >
> >    INVITE sip:bob@atlanta.com SIP/2.0
> >    Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bK56sdasks
> >    From: sip:alice@atlanta.com;tag=3Dasdyka899
> >    To: sip:bob@atlanta.com
> >    Call-ID: asidkj3ss
> >    CSeq: 1 INVITE
> >    Max-Forwards: 70
> >    Supported: path, outbound, gruu
> >    Route: <sip:proxy.atlanta.com:443;transport=3Dws;lr>
> >    Contact: <sip:alice@atlanta.com
> >     ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob>"
> >    Content-Type: application/sdp
> >
> >
> > ReplacementText / Comment:
> >
> >    Extra closing double quote in contact header.
>
> Typo, thanks :)
>
> [ABN] similar typo mistake in message F3 as well.


>
> >       Contact: <sip:alice@atlanta.com
> >           ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob>
> >
> >    Route: <sip:proxy.atlanta.com:80;transport=3Dws;lr>
> >
> >    change :
> >         443-->80              assuming HTTP <-- WS <-- SIP URI
> >         This comment is applicable for messages F1, F3, F4, F5, F6, F8,
> F9,
> > F10,
>
> Same comments as above :)
>
> The example uses a secure WS connection (on top of TLS) but does not
> use SIPS schema.
>
>
[ABN] Probably for the sake of clarity, we can give call flow for Secure
Web Socket (WSS)
and non-secure Web Socket (WS). this content is already covered in
different sections (this doc and rfc6455 as well). But putting it together
with a call flow would make the concept more clearer.

for all the messages WS, port 80 shall be used and port 443 for WSS. This
is to make WS to be backward compatible with default HTTP and HTTPS ports.


> >    Changes
> > WSS --> WS            assuming HTTP <-- WS <-- SIP URI
> >         This comment is applicable for messages F1, F2, F3, F4, F5, F6,
> F7,
>
> Idem.
>
>
>
>
> > -----------------------------------------------------
> > Section : 1.  Introduction
> >
> > CurrentText:
> > Media transport is out of the scope of this document
> >
> > ReplacementText / Comment:
> >    Any other draft / rfc which addresses this issue. it would improve t=
he
> >    readability of the document if we can is suggest reference for
> > completeness.
> >
> > -----------------------------------------------------
>
> We would not like to cover it. In case of web browsers, WebRTC
> specification is defining RTP/SRTP/ICE/TURN for web applications. But
> a SIP client using a WebSocket stack running in, i.e, a smartphone,
> could use usual RTP for media transport.
>
> We don't want to focus the draft on "web browsers" use case.
>
>
>
> > Section : 5.1.  General
> > CurrentText:
> >       (typically web-based applications with an strict and simple API
> >       for receiving WebSocket messages).  There is no need to establish
> >
> >
> > ReplacementText / Comment:
> >       (typically web-based applications with **a** strict and simple AP=
I
> >       for receiving WebSocket messages).  There is no need to establish
>
> Typo :)
>
>
>
>
> > Section :
> > CurrentText:
> >    In the absence of an explicit port and DNS SRV resource records, the
> >    default port for a SIP URI with "ws" transport parameter is 80 in
> >    case of SIP scheme and 443 in case of SIPS scheme.
> >
> >
> > ReplacementText / Comment:
> >    In the absence of an explicit port and DNS SRV resource records, the
> >    default port for a SIP URI with "ws" transport parameter is 80 in
> >    case of SIP scheme and 443 in case of SIPS scheme **, which is same =
as
> >    that used for http and https protocols respectively**
> >
> > These ports have been used for the sake of backward compatibility with
> > servers
> > which does not support WS/WSS.
> >
>
???

> > -----------------------------------------------------
> > Section :
> > CurrentText:
> >
> >  _This section is non-normative._
> >
> > ReplacementText / Comment:
> >  Some sections have been explicitly mentioned with this text. Does that
> mean
> >  other sections are normative text. Not clear though.
>
> Basically yes. The style is the same as in RFC 6455
> (http://tools.ietf.org/html/rfc6455) in which non normative sections
> include the above line. Any suggestion to make it better is of course
> welcome.
>
>
>
>
> Thanks a lot for your review, we will address all the topics covered in i=
t.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>



--=20
Thanks,
Nataraju A.B.

--f46d044268226b541804bdee683d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hi Inaki,=A0<div><br><div>Please find some more comments inline...<br><br><=
div><br class=3D"Apple-interchange-newline">Please let me know, the reasoni=
ng behind single transport parameter and two sent-protocol values.=A0</div>=
<div>
sent-protocol : WS and WSS</div><div>transport =3D ws (only one new value)<=
/div><div><br></div><div>This might be confusing in future...</div><div><br=
></div><div><br></div>To make the draft more readable, it=A0may=A0be a good=
 idea to add transport parameter for WSS as well.=A0</div>
<div><br></div><div><pre class=3D"newpage" style=3D"font-size:1em;margin-to=
p:0px;margin-bottom:0px">transport         =3D  &quot;UDP&quot; / &quot;TCP=
&quot; / &quot;TLS&quot; / &quot;SCTP&quot; / &quot;WS&quot; <b>/ &quot;WSS=
&quot;</b>
                     / other-transport
</pre><div><br></div></div><div><br></div><div><div class=3D"gmail_quote">O=
n Tue, Apr 17, 2012 at 10:55 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</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">2012/4/17 Nataraju A.B &lt;<a href=3D"mailto=
:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gmail.com</a>&gt;:<=
br>

<div>&gt; Hi Authros,<br>
&gt;<br>
&gt; Please find some comments on the draft<br>
&gt; &quot;draft-ibc-sipcore-sip-websocket-02.txt&quot;<br>
<br>
<br>
</div>Hi Nataraju, thanks for your comments. We will address them. Some<br>
replies inline:<br>
<div><br>
<br>
<br>
&gt; -----------------------------------------------------<br>
&gt; Section : =A08.1. =A0Registration<br>
&gt; CurrentText:<br>
&gt; =A0 =A0F1 REGISTER =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com"=
 target=3D"_blank">proxy.atlanta.com</a> (transport WSS)<br>
&gt;<br>
&gt; =A0 =A0REGISTER sip:<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> SIP/2.0<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf<br>
&gt;<br>
&gt; =A0 =A0F2 200 OK =A0<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> -&gt; Alice (transport WSS)<br>
&gt;<br>
&gt; =A0 =A0SIP/2.0 200 OK<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt;<br>
&gt; If I understand right,<br>
&gt; =A0 =A0 =A0 =A0 =A0 Websocket connection is mapped on to SIP URI =A0--=
&gt; HTTP<br>
&gt; =A0 =A0Secure Websocket connection is mapped on to SIPS URI --&gt; HTT=
PS<br>
&gt;<br>
&gt; HTTP --&gt; WS --&gt; SIP URI --&gt; port 80<br>
&gt;<br>
&gt; HTTPS --&gt; WSS --&gt; SIPS URI --&gt; port 443<br>
&gt;<br>
&gt; Hence in the following messages &quot;WSS&quot; shall be replaced with=
 &quot;WS&quot;<br>
&gt;<br>
&gt; =A0 =A0F1 REGISTER =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com"=
 target=3D"_blank">proxy.atlanta.com</a> (transport WSS)<br>
&gt;<br>
&gt; =A0 =A0REGISTER sip:<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> SIP/2.0<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf<br>
&gt;<br>
&gt; WSS --&gt; WS<br>
&gt;<br>
&gt; =A0 =A0F2 200 OK =A0<a href=3D"http://proxy.atlanta.com" target=3D"_bl=
ank">proxy.atlanta.com</a> -&gt; Alice (transport WSS)<br>
&gt;<br>
&gt; =A0 =A0SIP/2.0 200 OK<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf<br>
&gt;<br>
&gt; WSS --&gt; WS<br>
<br>
<br>
</div>Usage of a secure transport (like TLS-TCP or WSS) does not mandate<br=
>
usage of SIPS schema. This is clearly explained in:<br>
<br>
 =A0<a href=3D"http://tools.ietf.org/html/rfc5630#section-3.1.3" target=3D"=
_blank">http://tools.ietf.org/html/rfc5630#section-3.1.3</a><br>
<div><br></div></blockquote><div>&lt;rfc5630&gt; sec=A0<a name=3D"136c37dc9=
354426d_section-3.1.3" href=3D"http://tools.ietf.org/html/rfc5630#section-3=
.1.3" style=3D"font-size:1em;line-height:0pt;font-weight:bold;color:black;t=
ext-decoration:none" target=3D"_blank">3.1.3</a><span style=3D"font-size:1e=
m;line-height:0pt;font-weight:bold">.</span></div>

<div><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px">   If on=
e wants to use &quot;best-effort TLS&quot; for SIP, one just needs to use
   a SIP URI, and send the request over TLS.</pre></div><div><div>&lt;/rfc5=
630&gt;</div><br></div><div>if we want to use SIP URI with any other secure=
 communication mechanism like WSS, might require=A0</div><div>update to RFC=
5630. like</div>
<div><div>&lt;rfc5630&gt; sec=A0<a name=3D"136c37dc9354426d_section-3.1.3" =
href=3D"http://tools.ietf.org/html/rfc5630#section-3.1.3" target=3D"_blank"=
 style=3D"font-size:1em;line-height:0pt;font-weight:bold;color:black;text-d=
ecoration:none">3.1.3</a><span style=3D"font-size:1em;line-height:0pt;font-=
weight:bold">.</span></div>
<div><pre style=3D"font-size:1em;margin-top:0px;margin-bottom:0px">   If on=
e wants to use &quot;best-effort TLS/WSS&quot; for SIP, one just needs to u=
se
   a SIP URI, and send the request over TLS ** or any other secure transpor=
t mechanism like WSS**</pre></div><div>&lt;/rfc5630&gt;</div><div><br></div=
><div>for these changes, you can add a new section &quot;5.4=A0<span style=
=3D"white-space:pre-wrap">Updates to RFC </span>5630&quot;</div>
<div><br></div><div>Also add a normative reference for rfc5630 in sec &quot=
;<span style=3D"white-space:pre-wrap">12.2.  Informative References&quot;</=
span></div><div><br></div><pre style=3D"word-wrap:break-word;white-space:pr=
e-wrap">
   [RFC5630]  Audet, F., &quot;The Use of the SIPS URI Scheme in the Sessio=
n
              Initiation Protocol (SIP)&quot;, RFC 5630, October 2009.</pre=
><br class=3D"Apple-interchange-newline"></div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x">
<div><br>
<br>
&gt; For example, REGISTER request looks like this (with WSS)...<br>
&gt;<br>
&gt; =A0 =A0REGISTER sips:<a href=3D"http://proxy.atlanta.com" target=3D"_b=
lank">proxy.atlanta.com</a> SIP/2.0<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bKasudf<br>
&gt; =A0 =A0From: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_bl=
ank">sips:alice@atlanta.com</a>;tag=3D65bnmj.34asd<br>
&gt; =A0 =A0To: <a href=3D"mailto:sips%3Aalice@atlanta.com" target=3D"_blan=
k">sips:alice@atlanta.com</a><br>
&gt; =A0 =A0Call-ID: aiuy7k9njasd<br>
&gt; =A0 =A0CSeq: 1 REGISTER<br>
&gt; =A0 =A0Max-Forwards: 70<br>
&gt; =A0 =A0Supported: path, outbound, gruu<br>
&gt; =A0 =A0Contact: &lt;sips:alice@df7jal23ls0d.invalid;transport=3Dws&gt;=
<br>
&gt; =A0 =A0 =A0;reg-id=3D1<br>
&gt; =A0 =A0 =A0;+sip.instance=3D&quot;&lt;urn:uuid:f81-7dec-14a06cf1&gt;&q=
uot;<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0changes:<br>
&gt; =A0 =A0sip --&gt; sips in RURI, From, TO, contact headers<br>
<br>
</div>As stated above this is not needed. WSS (or TLS-TCP) can be used<br>
without using SIPS schema in SIP requests. That is a &quot;best-effort TLS<=
br>
/ WSS&quot;. Also using Outbound RFC 5626 helps here since the Outbound<br>
proxy (or the server the UA is connected to via TLS / WSS) can reuse<br>
the existing connection for incoming requests. This is also explained<br>
in <a href=3D"http://tools.ietf.org/html/rfc5630#section-3.1.3" target=3D"_=
blank">http://tools.ietf.org/html/rfc5630#section-3.1.3</a>.<br>
<br>
In summary: using a secure connection does not require using SIPS<br>
schema in the requests sent over it.<br>
<div><br>
<br>
<br>
<br>
&gt; -----------------------------------------------------<br>
&gt; Section : 8.2. =A0INVITE dialog through a proxy<br>
&gt; CurrentText:<br>
&gt;<br>
&gt; =A0 =A0F1 INVITE =A0Alice -&gt; <a href=3D"http://proxy.atlanta.com" t=
arget=3D"_blank">proxy.atlanta.com</a> (transport WSS)<br>
&gt;<br>
&gt; =A0 =A0INVITE <a href=3D"mailto:sip%3Abob@atlanta.com" target=3D"_blan=
k">sip:bob@atlanta.com</a> SIP/2.0<br>
&gt; =A0 =A0Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=3Dz9hG4bK56sdasks<=
br>
&gt; =A0 =A0From: <a href=3D"mailto:sip%3Aalice@atlanta.com" target=3D"_bla=
nk">sip:alice@atlanta.com</a>;tag=3Dasdyka899<br>
&gt; =A0 =A0To: <a href=3D"mailto:sip%3Abob@atlanta.com" target=3D"_blank">=
sip:bob@atlanta.com</a><br>
&gt; =A0 =A0Call-ID: asidkj3ss<br>
&gt; =A0 =A0CSeq: 1 INVITE<br>
&gt; =A0 =A0Max-Forwards: 70<br>
&gt; =A0 =A0Supported: path, outbound, gruu<br>
&gt; =A0 =A0Route: &lt;sip:proxy.atlanta.com:443;transport=3Dws;lr&gt;<br>
&gt; =A0 =A0Contact: &lt;<a href=3D"mailto:sip%3Aalice@atlanta.com" target=
=3D"_blank">sip:alice@atlanta.com</a><br>
&gt; =A0 =A0 ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob&gt;&quot;<br>
&gt; =A0 =A0Content-Type: application/sdp<br>
&gt;<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt;<br>
&gt; =A0 =A0Extra closing double quote in contact header.<br>
<br>
</div>Typo, thanks :)<br>
<div><br></div></blockquote><div>[ABN] similar typo mistake in message F3 a=
s well.</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>
<br>
&gt; =A0 =A0 =A0 Contact: &lt;<a href=3D"mailto:sip%3Aalice@atlanta.com" ta=
rget=3D"_blank">sip:alice@atlanta.com</a><br>
&gt; =A0 =A0 =A0 =A0 =A0 ;gr=3Durn:uuid:f81-7dec-14a06cf1;ob&gt;<br>
&gt;<br>
&gt; =A0 =A0Route: &lt;sip:proxy.atlanta.com:80;transport=3Dws;lr&gt;<br>
&gt;<br>
&gt; =A0 =A0change :<br>
&gt; =A0 =A0 =A0 =A0 443--&gt;80 =A0 =A0 =A0 =A0 =A0 =A0 =A0assuming=A0HTTP=
=A0&lt;--=A0WS &lt;-- SIP URI<br>
&gt; =A0 =A0 =A0 =A0 This comment is applicable for messages F1, F3, F4, F5=
, F6, F8, F9,<br>
&gt; F10,<br>
<br>
</div>Same comments as above :)<br>
<br>
The example uses a secure WS connection (on top of TLS) but does not<br>
use SIPS schema.<br>
<div><br></div></blockquote><div>=A0</div><div>[ABN] Probably for the sake =
of clarity, we can give call flow for Secure Web Socket (WSS)=A0</div><div>=
and non-secure Web Socket (WS). this content is already covered in differen=
t sections (this doc and rfc6455 as well). But putting it together with a c=
all flow would make the concept more clearer.</div>
<div><br></div><div>for all the messages WS, port 80 shall be used and port=
 443 for WSS. This is to make WS to be backward compatible with default HTT=
P and HTTPS ports.</div><div><br></div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<br>
&gt; =A0 =A0Changes<br>
&gt; WSS --&gt; WS =A0 =A0 =A0 =A0 =A0 =A0assuming=A0HTTP=A0&lt;--=A0WS &lt=
;-- SIP URI<br>
&gt; =A0 =A0 =A0 =A0 This comment is applicable for messages F1, F2, F3, F4=
, F5, F6, F7,<br>
<br>
</div>Idem.<br>
<div><br>
<br>
<br>
<br>
&gt; -----------------------------------------------------<br>
&gt; Section : 1. =A0Introduction<br>
&gt;<br>
&gt; CurrentText:<br>
&gt; Media transport is out of the scope of this document<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt; =A0 =A0Any other draft / rfc which addresses this issue. it would impr=
ove the<br>
&gt; =A0 =A0readability of the document if we can is suggest reference for<=
br>
&gt; completeness.<br>
&gt;<br>
&gt; -----------------------------------------------------<br>
<br>
</div>We would not like to cover it. In case of web browsers, WebRTC<br>
specification is defining RTP/SRTP/ICE/TURN for web applications. But<br>
a SIP client using a WebSocket stack running in, i.e, a smartphone,<br>
could use usual RTP for media transport.<br>
<br>
We don&#39;t want to focus the draft on &quot;web browsers&quot; use case.<=
br>
<div><br>
<br>
<br>
&gt; Section : 5.1. =A0General<br>
&gt; CurrentText:<br>
&gt; =A0 =A0 =A0 (typically web-based applications with an strict and simpl=
e API<br>
&gt; =A0 =A0 =A0 for receiving WebSocket messages). =A0There is no need to =
establish<br>
&gt;<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt; =A0 =A0 =A0 (typically web-based applications with **a** strict and si=
mple API<br>
&gt; =A0 =A0 =A0 for receiving WebSocket messages). =A0There is no need to =
establish<br>
<br>
</div>Typo :)<br>
<div><br>
<br>
<br>
<br>
&gt; Section :<br>
&gt; CurrentText:<br>
&gt; =A0 =A0In the absence of an explicit port and DNS SRV resource records=
, the<br>
&gt; =A0 =A0default port for a SIP URI with &quot;ws&quot; transport parame=
ter is 80 in<br>
&gt; =A0 =A0case of SIP scheme and 443 in case of SIPS scheme.<br>
&gt;<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt; =A0 =A0In the absence of an explicit port and DNS SRV resource records=
, the<br>
&gt; =A0 =A0default port for a SIP URI with &quot;ws&quot; transport parame=
ter is 80 in<br>
&gt; =A0 =A0case of SIP scheme and 443 in case of SIPS scheme **, which is =
same as<br>
&gt; =A0 =A0that used for http and https protocols respectively**<br>
&gt;<br>
&gt; These ports have been used for the sake of backward compatibility with=
<br>
&gt; servers<br>
&gt; which does not support WS/WSS.<br>
&gt;<br></div></blockquote><div>???=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">=
<div>
&gt; -----------------------------------------------------<br>
&gt; Section :<br>
&gt; CurrentText:<br>
&gt;<br>
&gt; =A0_This section is non-normative._<br>
&gt;<br>
&gt; ReplacementText / Comment:<br>
&gt; =A0Some sections have been explicitly mentioned with this text. Does t=
hat mean<br>
&gt; =A0other sections are normative text. Not clear though.<br>
<br>
</div>Basically yes. The style is the same as in RFC 6455<br>
(<a href=3D"http://tools.ietf.org/html/rfc6455" target=3D"_blank">http://to=
ols.ietf.org/html/rfc6455</a>) in which non normative sections<br>
include the above line. Any suggestion to make it better is of course<br>
welcome.<br>
<br>
<br>
<br>
<br>
Thanks a lot for your review, we will address all the topics covered in it.=
<br>
<font color=3D"#888888"><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
</font></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font=
 color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"=
1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;cou=
rier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>

<br>
</div></div>

--f46d044268226b541804bdee683d--

From pravindran@sonusnet.com  Wed Apr 18 00:02:15 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Nataraju A.B <nataraju.sip@gmail.com>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Date: Wed, 18 Apr 2012 07:02:01 +0000
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Nataraju,

As per Sec 18 of RFC 3261 in , "All SIP elements MUST implement UDP and TCP=
.", so it is possible to have stateless or transaction stateful proxy routi=
ng with change in transport mechanism from UDP to TCP and vice versa. The s=
imple scenario like INVITE transaction over TCP whereas Re-INVITE over UDP =
based on contact SIP URI with UDP as a transport should work as per RFC 326=
1.

In case of draft-ibc-sipcore-sip-websocket-02 draft, SIP UA over websocket =
(like browser) will not implement either UDP or TCP transport which is a de=
viation from RFC 3261. This deviation leads to this limitation of not havin=
g stateless or Transaction stateful proxy in websocket transport.

Thanks
Partha

From: Nataraju A.B [mailto:nataraju.sip@gmail.com]
Sent: Wednesday, April 18, 2012 11:48 AM
To: Ravindran, Parthasarathi
Cc: SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Partha,

I also feel the same. The scenario is very similar to proxy interfacing bet=
ween UDP and TCP on different sides, where dialog stateful proxy functional=
ity is required. stateless / transaction stateful proxies would not work pr=
operly when requests are sent over different protocols.

Thanks,
Nataraju A B
On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi <pravindran@sonu=
snet.com<mailto:pravindran@sonusnet.com>> wrote:
As per my current reading, The current draft assumes about dialog-stateful =
proxy wherein all record-route header will be added whenever Websocket to o=
ther transport routing happens in the proxy and proxy will be involved in a=
ll the transaction within the dialog. So, it is not possible to have statel=
ess or transaction stateful proxy. It will be good in case this limitation =
is explicitly mentioned in the draft.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:si=
pcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>] On
>Behalf Of Adam Roach - SIPCORE Chair
>Sent: Monday, April 16, 2012 11:33 PM
>To: SIPCORE (Session Initiation Protocol Core) WG
>Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
>
>[as chair]
>
>SIPCORE Working Group:
>
>This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
>document to satisfy our recently added milestone to define a WebSockets
>transport for the SIP protocol. Interested parties should voice their
>support and/or concerns on the SIPCORE mailing list. The chairs plan to
>make a determination of consensus on Friday, April 27th.
>
>Thank you. The draft announcement follows.
>
>/a
>
>-------- Original Message --------
>Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-
>websocket-02.txt
>Date: Mon, 16 Apr 2012 18:49:38 +0200
>From: I=F1aki Baz Castillo <ibc@aliax.net<mailto:ibc@aliax.net>>
>To: SIPCORE WG <sipcore@ietf.org<mailto:sipcore@ietf.org>>
>
>A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
>successfully submitted by I=F1aki Baz Castillo and posted to the IETF
>repository.
>
>Filename:        draft-ibc-sipcore-sip-websocket
>Revision:        02
>Title:           The WebSocket Protocol as a Transport for the Session
>Initiation Protocol (SIP)
>Creation date:   2012-04-16
>WG ID:           Individual Submission
>Number of pages: 20
>
>Abstract:
>   The WebSocket protocol enables two-way realtime communication between
>   clients and servers.  This document specifies a new WebSocket sub-
>   protocol as a reliable transport mechanism between SIP (Session
>   Initiation Protocol) entities and enables usage of the SIP protocol
>   in new scenarios.
>
>
>- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
>- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
>
>
>This new version incorporates improvements and changes based on the
>comments and reviews on the previous version, along with the consensus
>got in IETF 83 Paris.
>
>
>As usual, comments are welcome. Thanks a lot.
>
>
>--
>I=F1aki Baz Castillo
><ibc@aliax.net<mailto:ibc@aliax.net>>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org<mailto:sipcore@ietf.org>
>https://www.ietf.org/mailman/listinfo/sipcore
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org<mailto:sipcore@ietf.org>
>https://www.ietf.org/mailman/listinfo/sipcore
_______________________________________________
sipcore mailing list
sipcore@ietf.org<mailto:sipcore@ietf.org>
https://www.ietf.org/mailman/listinfo/sipcore



--
Thanks,
Nataraju A.B.


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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Nataraju,<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">As per Sec 18 of RFC 3261=
 in , &#8220;</span>All SIP elements MUST implement UDP and TCP.&#8221;,
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D">so it is possible to have stateless or transacti=
on stateful proxy routing with change in transport mechanism from UDP to TC=
P and vice versa. The simple scenario like INVITE transaction
 over TCP whereas Re-INVITE over UDP based on contact SIP URI with UDP as a=
 transport should work as per RFC 3261. &nbsp;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In case of draft-ibc-sipc=
ore-sip-websocket-02 draft, SIP UA over websocket (like browser) will not i=
mplement either UDP or TCP transport which is a deviation
 from RFC 3261. This deviation leads to this limitation of not having state=
less or Transaction stateful proxy in websocket transport.<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Partha<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju=
 A.B [mailto:nataraju.sip@gmail.com]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 11:48 AM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG<br=
>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Partha,&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I also feel the same. The scenario is very similar t=
o proxy interfacing between UDP and TCP on different sides, where dialog st=
ateful proxy functionality is required. stateless / transaction stateful pr=
oxies would not work properly when
 requests are sent over different protocols.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Nataraju A B<o:p></o:=
p></p>
<div>
<p class=3D"MsoNormal">On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthas=
arathi &lt;<a href=3D"mailto:pravindran@sonusnet.com">pravindran@sonusnet.c=
om</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">As per my current reading, The current draft assumes=
 about dialog-stateful proxy wherein all record-route header will be added =
whenever Websocket to other transport routing happens in the proxy and prox=
y will be involved in all the transaction
 within the dialog. So, it is not possible to have stateless or transaction=
 stateful proxy. It will be good in case this limitation is explicitly ment=
ioned in the draft.<br>
<br>
Thanks<br>
Partha<br>
<br>
&gt;-----Original Message-----<br>
&gt;From: <a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.=
org</a> [mailto:<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces=
@ietf.org</a>] On<br>
&gt;Behalf Of Adam Roach - SIPCORE Chair<br>
&gt;Sent: Monday, April 16, 2012 11:33 PM<br>
&gt;To: SIPCORE (Session Initiation Protocol Core) WG<br>
&gt;Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-0=
2<br>
&gt;<br>
&gt;[as chair]<br>
&gt;<br>
&gt;SIPCORE Working Group:<br>
&gt;<br>
&gt;This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02<b=
r>
&gt;document to satisfy our recently added milestone to define a WebSockets=
<br>
&gt;transport for the SIP protocol. Interested parties should voice their<b=
r>
&gt;support and/or concerns on the SIPCORE mailing list. The chairs plan to=
<br>
&gt;make a determination of consensus on Friday, April 27th.<br>
&gt;<br>
&gt;Thank you. The draft announcement follows.<br>
&gt;<br>
&gt;/a<br>
&gt;<br>
&gt;-------- Original Message --------<br>
&gt;Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02.txt<br>
&gt;Date: Mon, 16 Apr 2012 18:49:38 &#43;0200<br>
&gt;From: I=F1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net">ibc@ali=
ax.net</a>&gt;<br>
&gt;To: SIPCORE WG &lt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org=
</a>&gt;<br>
&gt;<br>
&gt;A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been<b=
r>
&gt;successfully submitted by I=F1aki Baz Castillo and posted to the IETF<b=
r>
&gt;repository.<br>
&gt;<br>
&gt;Filename: &nbsp; &nbsp; &nbsp; &nbsp;draft-ibc-sipcore-sip-websocket<br=
>
&gt;Revision: &nbsp; &nbsp; &nbsp; &nbsp;02<br>
&gt;Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; The WebSocket Protocol as a T=
ransport for the Session<br>
&gt;Initiation Protocol (SIP)<br>
&gt;Creation date: &nbsp; 2012-04-16<br>
&gt;WG ID: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
&gt;Number of pages: 20<br>
&gt;<br>
&gt;Abstract:<br>
&gt; &nbsp; The WebSocket protocol enables two-way realtime communication b=
etween<br>
&gt; &nbsp; clients and servers. &nbsp;This document specifies a new WebSoc=
ket sub-<br>
&gt; &nbsp; protocol as a reliable transport mechanism between SIP (Session=
<br>
&gt; &nbsp; Initiation Protocol) entities and enables usage of the SIP prot=
ocol<br>
&gt; &nbsp; in new scenarios.<br>
&gt;<br>
&gt;<br>
&gt;- <a href=3D"http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket=
-02" target=3D"_blank">
http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02</a><br>
&gt;- <a href=3D"http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-0=
2.txt" target=3D"_blank">
http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt</a><br>
&gt;<br>
&gt;<br>
&gt;This new version incorporates improvements and changes based on the<br>
&gt;comments and reviews on the previous version, along with the consensus<=
br>
&gt;got in IETF 83 Paris.<br>
&gt;<br>
&gt;<br>
&gt;As usual, comments are welcome. Thanks a lot.<br>
&gt;<br>
&gt;<br>
&gt;--<br>
&gt;I=F1aki Baz Castillo<br>
&gt;&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
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sipcore mailing list<br>
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<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
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From nataraju.sip@gmail.com  Wed Apr 18 00:28:47 2012
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com>
Date: Wed, 18 Apr 2012 12:58:41 +0530
Message-ID: <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com>
From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Content-Type: multipart/alternative; boundary=f46d040122a33014d804bdef01fa
Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--f46d040122a33014d804bdef01fa
Content-Type: text/plain; charset=windows-1252
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Partha,

I was referring to scenario,

UAC                  Proxy                   UAS
 -----sips/tcp---->     ------>sip/udp --->

UAC requested for a secure connection. But assume proxy at the edge and
can't guarantee secure connection towards UAS. hence It is sending a
request to UAS over simple udp *without security*.

In this case, proxy must behave like a dialog stateful proxy. stateless or
transaction stateful proxy can't inter work in this scenario.

This is my understanding. Others please share your opinion on the same....

Thanks,
Nataraju A B

On Wed, Apr 18, 2012 at 12:32 PM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

>  Nataraju,**
>
> ** **
>
> As per Sec 18 of RFC 3261 in , =93All SIP elements MUST implement UDP and
> TCP.=94, so it is possible to have stateless or transaction stateful prox=
y
> routing with change in transport mechanism from UDP to TCP and vice versa=
.
> The simple scenario like INVITE transaction over TCP whereas Re-INVITE ov=
er
> UDP based on contact SIP URI with UDP as a transport should work as per R=
FC
> 3261.  ****
>
> ** **
>
> In case of draft-ibc-sipcore-sip-websocket-02 draft, SIP UA over websocke=
t
> (like browser) will not implement either UDP or TCP transport which is a
> deviation from RFC 3261. This deviation leads to this limitation of not
> having stateless or Transaction stateful proxy in websocket transport.***=
*
>
> ** **
>
> Thanks****
>
> Partha****
>
> ** **
>
> *From:* Nataraju A.B [mailto:nataraju.sip@gmail.com]
> *Sent:* Wednesday, April 18, 2012 11:48 AM
> *To:* Ravindran, Parthasarathi
> *Cc:* SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG
> *Subject:* Re: [sipcore] Stateless or Transaction stateful proxy possible
> in Websocket transport? [RE: Call for Adoption:
> draft-ibc-sipcore-sip-websocket-02]****
>
> ** **
>
> Partha, ****
>
> ** **
>
> I also feel the same. The scenario is very similar to proxy interfacing
> between UDP and TCP on different sides, where dialog stateful proxy
> functionality is required. stateless / transaction stateful proxies would
> not work properly when requests are sent over different protocols.****
>
> ** **
>
> Thanks,****
>
> Nataraju A B****
>
> On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi <
> pravindran@sonusnet.com> wrote:****
>
> As per my current reading, The current draft assumes about dialog-statefu=
l
> proxy wherein all record-route header will be added whenever Websocket to
> other transport routing happens in the proxy and proxy will be involved i=
n
> all the transaction within the dialog. So, it is not possible to have
> stateless or transaction stateful proxy. It will be good in case this
> limitation is explicitly mentioned in the draft.
>
> Thanks
> Partha
>
> >-----Original Message-----
> >From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> >Behalf Of Adam Roach - SIPCORE Chair
> >Sent: Monday, April 16, 2012 11:33 PM
> >To: SIPCORE (Session Initiation Protocol Core) WG
> >Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
> >
> >[as chair]
> >
> >SIPCORE Working Group:
> >
> >This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
> >document to satisfy our recently added milestone to define a WebSockets
> >transport for the SIP protocol. Interested parties should voice their
> >support and/or concerns on the SIPCORE mailing list. The chairs plan to
> >make a determination of consensus on Friday, April 27th.
> >
> >Thank you. The draft announcement follows.
> >
> >/a
> >
> >-------- Original Message --------
> >Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-
> >websocket-02.txt
> >Date: Mon, 16 Apr 2012 18:49:38 +0200
> >From: I=F1aki Baz Castillo <ibc@aliax.net>
> >To: SIPCORE WG <sipcore@ietf.org>
> >
> >A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
> >successfully submitted by I=F1aki Baz Castillo and posted to the IETF
> >repository.
> >
> >Filename:        draft-ibc-sipcore-sip-websocket
> >Revision:        02
> >Title:           The WebSocket Protocol as a Transport for the Session
> >Initiation Protocol (SIP)
> >Creation date:   2012-04-16
> >WG ID:           Individual Submission
> >Number of pages: 20
> >
> >Abstract:
> >   The WebSocket protocol enables two-way realtime communication between
> >   clients and servers.  This document specifies a new WebSocket sub-
> >   protocol as a reliable transport mechanism between SIP (Session
> >   Initiation Protocol) entities and enables usage of the SIP protocol
> >   in new scenarios.
> >
> >
> >- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
> >- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
> >
> >
> >This new version incorporates improvements and changes based on the
> >comments and reviews on the previous version, along with the consensus
> >got in IETF 83 Paris.
> >
> >
> >As usual, comments are welcome. Thanks a lot.
> >
> >
> >--
> >I=F1aki Baz Castillo
> ><ibc@aliax.net>
> >_______________________________________________
> >sipcore mailing list
> >sipcore@ietf.org
> >https://www.ietf.org/mailman/listinfo/sipcore
> >_______________________________________________
> >sipcore mailing list
> >sipcore@ietf.org
> >https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore****
>
>
>
> ****
>
> ** **
>
> --
> Thanks,****
>
> Nataraju A.B.****
>
> ** **
>



--=20
Thanks,
Nataraju A.B.

--f46d040122a33014d804bdef01fa
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

<font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace">Partha,=
=A0</font><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monos=
pace"><br></font></div><div><font color=3D"#000099" face=3D"&#39;courier ne=
w&#39;, monospace">I was referring to scenario,=A0</font></div>
<div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br>=
</font></div><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, mo=
nospace">UAC =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Proxy =A0 =A0 =A0 =A0 =A0 =
=A0 =A0 =A0 =A0 UAS</font></div><div><font color=3D"#000099" face=3D"&#39;c=
ourier new&#39;, monospace">=A0-----sips/tcp----&gt; =A0 =A0 ------&gt;sip/=
udp ---&gt;</font></div>
<div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br>=
</font></div><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, mo=
nospace">UAC requested for a secure connection. But assume proxy at the edg=
e and can&#39;t guarantee secure connection towards UAS. hence It is sendin=
g a request to UAS over simple udp <b>without security</b>.=A0</font></div>
<div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br>=
</font></div><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, mo=
nospace">In this case, proxy must=A0behave=A0like a dialog stateful proxy. =
stateless or transaction stateful proxy can&#39;t=A0inter work=A0in this sc=
enario.=A0</font></div>
<div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br>=
</font></div><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, mo=
nospace">This is my understanding. Others please share your opinion on the =
same....<br>
<br>Thanks,</font></div><div><font color=3D"#000099" face=3D"&#39;courier n=
ew&#39;, monospace">Nataraju A B</font></div><div><br><div class=3D"gmail_q=
uote">On Wed, Apr 18, 2012 at 12:32 PM, Ravindran, Parthasarathi <span dir=
=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusnet.com">pravindran@sonusnet=
.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">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Nataraju,<u></u></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">As per Sec 18 of RFC 3261=
 in , =93</span>All SIP elements MUST implement UDP and TCP.=94,
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1f497d">so it is possible to have stateless or transacti=
on stateful proxy routing with change in transport mechanism from UDP to TC=
P and vice versa. The simple scenario like INVITE transaction
 over TCP whereas Re-INVITE over UDP based on contact SIP URI with UDP as a=
 transport should work as per RFC 3261. =A0<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">In case of draft-ibc-sipc=
ore-sip-websocket-02 draft, SIP UA over websocket (like browser) will not i=
mplement either UDP or TCP transport which is a deviation
 from RFC 3261. This deviation leads to this limitation of not having state=
less or Transaction stateful proxy in websocket transport.<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Thanks<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Partha<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju=
 A.B [mailto:<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">na=
taraju.sip@gmail.com</a>]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 11:48 AM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG<br=
>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]<u></u><u></u></span></p>
</div>
</div><div><div></div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">Partha,=A0<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I also feel the same. The scenario is very similar t=
o proxy interfacing between UDP and TCP on different sides, where dialog st=
ateful proxy functionality is required. stateless / transaction stateful pr=
oxies would not work properly when
 requests are sent over different protocols.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Thanks,<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">Nataraju A B<u></u><u=
></u></p>
<div>
<p class=3D"MsoNormal">On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthas=
arathi &lt;<a href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pra=
vindran@sonusnet.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">As per my current reading, The current draft assumes=
 about dialog-stateful proxy wherein all record-route header will be added =
whenever Websocket to other transport routing happens in the proxy and prox=
y will be involved in all the transaction
 within the dialog. So, it is not possible to have stateless or transaction=
 stateful proxy. It will be good in case this limitation is explicitly ment=
ioned in the draft.<br>
<br>
Thanks<br>
Partha<br>
<br>
&gt;-----Original Message-----<br>
&gt;From: <a href=3D"mailto:sipcore-bounces@ietf.org" target=3D"_blank">sip=
core-bounces@ietf.org</a> [mailto:<a href=3D"mailto:sipcore-bounces@ietf.or=
g" target=3D"_blank">sipcore-bounces@ietf.org</a>] On<br>
&gt;Behalf Of Adam Roach - SIPCORE Chair<br>
&gt;Sent: Monday, April 16, 2012 11:33 PM<br>
&gt;To: SIPCORE (Session Initiation Protocol Core) WG<br>
&gt;Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-0=
2<br>
&gt;<br>
&gt;[as chair]<br>
&gt;<br>
&gt;SIPCORE Working Group:<br>
&gt;<br>
&gt;This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02<b=
r>
&gt;document to satisfy our recently added milestone to define a WebSockets=
<br>
&gt;transport for the SIP protocol. Interested parties should voice their<b=
r>
&gt;support and/or concerns on the SIPCORE mailing list. The chairs plan to=
<br>
&gt;make a determination of consensus on Friday, April 27th.<br>
&gt;<br>
&gt;Thank you. The draft announcement follows.<br>
&gt;<br>
&gt;/a<br>
&gt;<br>
&gt;-------- Original Message --------<br>
&gt;Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02.txt<br>
&gt;Date: Mon, 16 Apr 2012 18:49:38 +0200<br>
&gt;From: I=F1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;<br>
&gt;To: SIPCORE WG &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank=
">sipcore@ietf.org</a>&gt;<br>
&gt;<br>
&gt;A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been<b=
r>
&gt;successfully submitted by I=F1aki Baz Castillo and posted to the IETF<b=
r>
&gt;repository.<br>
&gt;<br>
&gt;Filename: =A0 =A0 =A0 =A0draft-ibc-sipcore-sip-websocket<br>
&gt;Revision: =A0 =A0 =A0 =A002<br>
&gt;Title: =A0 =A0 =A0 =A0 =A0 The WebSocket Protocol as a Transport for th=
e Session<br>
&gt;Initiation Protocol (SIP)<br>
&gt;Creation date: =A0 2012-04-16<br>
&gt;WG ID: =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
&gt;Number of pages: 20<br>
&gt;<br>
&gt;Abstract:<br>
&gt; =A0 The WebSocket protocol enables two-way realtime communication betw=
een<br>
&gt; =A0 clients and servers. =A0This document specifies a new WebSocket su=
b-<br>
&gt; =A0 protocol as a reliable transport mechanism between SIP (Session<br=
>
&gt; =A0 Initiation Protocol) entities and enables usage of the SIP protoco=
l<br>
&gt; =A0 in new scenarios.<br>
&gt;<br>
&gt;<br>
&gt;- <a href=3D"http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket=
-02" target=3D"_blank">
http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02</a><br>
&gt;- <a href=3D"http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-0=
2.txt" target=3D"_blank">
http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt</a><br>
&gt;<br>
&gt;<br>
&gt;This new version incorporates improvements and changes based on the<br>
&gt;comments and reviews on the previous version, along with the consensus<=
br>
&gt;got in IETF 83 Paris.<br>
&gt;<br>
&gt;<br>
&gt;As usual, comments are welcome. Thanks a lot.<br>
&gt;<br>
&gt;<br>
&gt;--<br>
&gt;I=F1aki Baz Castillo<br>
&gt;&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a=
>&gt;<br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
/a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
/a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
</div>
</div></div></div>
</div>
</div>

</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font color=
=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>
</div>

--f46d040122a33014d804bdef01fa--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 10:16:50 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> As per my current reading, The current draft assumes about dialog-statefu=
l proxy wherein all record-route header will be added whenever Websocket to=
 other transport routing happens in the proxy and proxy will be involved in=
 all the transaction within the dialog.

Hi, the draft does not assume that. Perhaps you talk about the example
scenario in the non-normative Appendix A?.

Anyhow what you mean "dialog-stateful proxy" is incorrect (see below please=
).



> So, it is not possible to have stateless or transaction stateful proxy. I=
t will be good in case this limitation is explicitly mentioned in the draft=
.

AFAIR a proxy can be *transaction stateless* regardless it bridges
different SIP transports. A stateless proxy is supposed to route
responses based on Via header. In case of reliable transports, the
first choice must be using the existing connection. If the *stateless*
proxy adds ;received and ;rport to the Via in the incoming request,
then it could later route responses to the UAC by inspecting those
parameters in the responses and getting the associated connection. The
proxy could also mantain a relationship between branch values and
associated connections, but then it would be not so "stateless".


Also, please don't mix "loose-routing" and "dialog-stateful". A pure
SIP proxy performing loose-routing (in-dialog requests go through the
proxy) can be a transaction stateful proxy but know nothing about
"dialogs".


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Nataraju A.B <nataraju.sip@gmail.com>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--_000_387F9047F55E8C42850AD6B3A7A03C6C0E227F98inbamail01sonus_
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Hi Nataraju,

As per Sec 20.42 of RFC 3261  SIPS URI indicates that the transport is TLS =
(and not TCP)
and the snippet is "When a request is sent to a SIPS URI, the protocol stil=
l indicates "SIP",
and the transport protocol is TLS."

As per Sec 26.2.1 Para 7 of RFC 3261, TLS transport is hop-by-hop and not e=
nd-to-end and RFC
Snippet is as follows:

"Note that
transport mechanisms are specified on a hop-by-hop basis in SIP, thus
a UA that sends requests over TLS to a proxy server has no assurance
that TLS will be used end-to-end."

Please note that SIP UA with TLS transport is not going to violate Sec 18 o=
f RFC 3261 "All SIP
elements MUST implement UDP and TCP.".  In the below mentioned callflow, pr=
oxy shall
acts as stateless or transaction stateful proxy wherein UAC shall sent INVI=
TE with SIPS URI
towards proxy, proxy forwards INVITE in UDP/TCP towards UAS and receive con=
tact with UDP/TCP
as a transport in 2xx response from UAS, 2xx is forwarded towards UAC. Here=
, UAC is expected to
send RE-INVITE/BYE towards UAS with transport as UDP/TCP. So, it is possibl=
e to interwork with
TLS to UDP/TCP as per RFC 3261.

In case you mention the same above logic of TLS applies to Websocket, anybo=
dy implement SIP
over websocket in browser using JavaScript violates RFC 3261 as it is not p=
ossible for browser to
send REINVITE/BYE using UDP/TCP transport towards UAS directly with statele=
ss/Transaction
stateful proxy as a first proxy hop.

Thanks
Partha


From: Nataraju A.B [mailto:nataraju.sip@gmail.com]
Sent: Wednesday, April 18, 2012 12:59 PM
To: Ravindran, Parthasarathi
Cc: SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Partha,

I was referring to scenario,

UAC                  Proxy                   UAS
 -----sips/tcp---->     ------>sip/udp --->

UAC requested for a secure connection. But assume proxy at the edge and can=
't guarantee secure connection towards UAS. hence It is sending a request t=
o UAS over simple udp without security.

In this case, proxy must behave like a dialog stateful proxy. stateless or =
transaction stateful proxy can't inter work in this scenario.

This is my understanding. Others please share your opinion on the same....

Thanks,
Nataraju A B

On Wed, Apr 18, 2012 at 12:32 PM, Ravindran, Parthasarathi <pravindran@sonu=
snet.com<mailto:pravindran@sonusnet.com>> wrote:
Nataraju,

As per Sec 18 of RFC 3261 in , "All SIP elements MUST implement UDP and TCP=
.", so it is possible to have stateless or transaction stateful proxy routi=
ng with change in transport mechanism from UDP to TCP and vice versa. The s=
imple scenario like INVITE transaction over TCP whereas Re-INVITE over UDP =
based on contact SIP URI with UDP as a transport should work as per RFC 326=
1.

In case of draft-ibc-sipcore-sip-websocket-02 draft, SIP UA over websocket =
(like browser) will not implement either UDP or TCP transport which is a de=
viation from RFC 3261. This deviation leads to this limitation of not havin=
g stateless or Transaction stateful proxy in websocket transport.

Thanks
Partha

From: Nataraju A.B [mailto:nataraju.sip@gmail.com<mailto:nataraju.sip@gmail=
.com>]
Sent: Wednesday, April 18, 2012 11:48 AM
To: Ravindran, Parthasarathi
Cc: SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Partha,

I also feel the same. The scenario is very similar to proxy interfacing bet=
ween UDP and TCP on different sides, where dialog stateful proxy functional=
ity is required. stateless / transaction stateful proxies would not work pr=
operly when requests are sent over different protocols.

Thanks,
Nataraju A B
On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi <pravindran@sonu=
snet.com<mailto:pravindran@sonusnet.com>> wrote:
As per my current reading, The current draft assumes about dialog-stateful =
proxy wherein all record-route header will be added whenever Websocket to o=
ther transport routing happens in the proxy and proxy will be involved in a=
ll the transaction within the dialog. So, it is not possible to have statel=
ess or transaction stateful proxy. It will be good in case this limitation =
is explicitly mentioned in the draft.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:si=
pcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org>] On
>Behalf Of Adam Roach - SIPCORE Chair
>Sent: Monday, April 16, 2012 11:33 PM
>To: SIPCORE (Session Initiation Protocol Core) WG
>Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-02
>
>[as chair]
>
>SIPCORE Working Group:
>
>This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02
>document to satisfy our recently added milestone to define a WebSockets
>transport for the SIP protocol. Interested parties should voice their
>support and/or concerns on the SIPCORE mailing list. The chairs plan to
>make a determination of consensus on Friday, April 27th.
>
>Thank you. The draft announcement follows.
>
>/a
>
>-------- Original Message --------
>Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-
>websocket-02.txt
>Date: Mon, 16 Apr 2012 18:49:38 +0200
>From: I=F1aki Baz Castillo <ibc@aliax.net<mailto:ibc@aliax.net>>
>To: SIPCORE WG <sipcore@ietf.org<mailto:sipcore@ietf.org>>
>
>A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been
>successfully submitted by I=F1aki Baz Castillo and posted to the IETF
>repository.
>
>Filename:        draft-ibc-sipcore-sip-websocket
>Revision:        02
>Title:           The WebSocket Protocol as a Transport for the Session
>Initiation Protocol (SIP)
>Creation date:   2012-04-16
>WG ID:           Individual Submission
>Number of pages: 20
>
>Abstract:
>   The WebSocket protocol enables two-way realtime communication between
>   clients and servers.  This document specifies a new WebSocket sub-
>   protocol as a reliable transport mechanism between SIP (Session
>   Initiation Protocol) entities and enables usage of the SIP protocol
>   in new scenarios.
>
>
>- http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02
>- http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt
>
>
>This new version incorporates improvements and changes based on the
>comments and reviews on the previous version, along with the consensus
>got in IETF 83 Paris.
>
>
>As usual, comments are welcome. Thanks a lot.
>
>
>--
>I=F1aki Baz Castillo
><ibc@aliax.net<mailto:ibc@aliax.net>>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org<mailto:sipcore@ietf.org>
>https://www.ietf.org/mailman/listinfo/sipcore
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org<mailto:sipcore@ietf.org>
>https://www.ietf.org/mailman/listinfo/sipcore
_______________________________________________
sipcore mailing list
sipcore@ietf.org<mailto:sipcore@ietf.org>
https://www.ietf.org/mailman/listinfo/sipcore



--
Thanks,
Nataraju A.B.




--
Thanks,
Nataraju A.B.


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Nataraju,<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">As per Sec 20.42 of RFC 3=
261 &nbsp;SIPS URI indicates that the transport is TLS (and not TCP)
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">and the snippet is &#8220=
;When a request is sent to a SIPS URI, the protocol still indicates &quot;S=
IP&quot;,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">and the transport protoco=
l is TLS.&#8221;<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">As per Sec 26.2.1 Para 7 =
of RFC 3261, TLS transport is hop-by-hop and not end-to-end and RFC
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Snippet is as follows:<o:=
p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&#8220;Note that
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">transport mechanisms are specified on a hop-by-hop basis i=
n SIP, thus<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">a UA that sends requests over TLS to a proxy server has no=
 assurance<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">that TLS will be used end-to-end.&#8221;<o:p></o:p></span>=
</p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Please note that SIP UA w=
ith TLS transport is not going to violate Sec 18 of RFC 3261 &#8220;</span>=
All SIP
<o:p></o:p></p>
<p class=3D"MsoNormal">elements MUST implement UDP and TCP.&#8221;. &nbsp;<=
span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-s=
erif&quot;;color:#1F497D">In the below mentioned callflow, proxy shall<o:p>=
</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">acts as stateless or tran=
saction stateful proxy wherein UAC shall sent INVITE with SIPS URI
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">towards proxy, proxy forw=
ards INVITE in UDP/TCP towards UAS and receive contact with UDP/TCP<o:p></o=
:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">as a transport in 2xx res=
ponse from UAS, 2xx is forwarded towards UAC. Here, UAC is expected to
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">send RE-INVITE/BYE toward=
s UAS with transport as UDP/TCP. So, it is possible to interwork with
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">TLS to UDP/TCP as per RFC=
 3261.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">In case you mention the s=
ame above logic of TLS applies to Websocket, anybody implement SIP
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">over websocket in browser=
 using JavaScript violates RFC 3261 as it is not possible for browser to<o:=
p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">send REINVITE/BYE using U=
DP/TCP transport towards UAS directly with stateless/Transaction<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">stateful proxy as a first=
 proxy hop.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Partha<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju=
 A.B [mailto:nataraju.sip@gmail.com]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 12:59 PM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG<br=
>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">Partha,&nbsp;</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">I was referring to scenario,&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">UAC &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &=
nbsp;Proxy &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; U=
AS</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">&nbsp;-----sips/tcp----&gt; &nbsp; &nbsp; ------&gt;sip/udp -=
--&gt;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">UAC requested for a secure connection. But assume proxy at th=
e edge and can't guarantee secure connection towards UAS. hence It is sendi=
ng a request to UAS over simple udp
<b>without security</b>.&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">In this case, proxy must&nbsp;behave&nbsp;like a dialog state=
ful proxy. stateless or transaction stateful proxy can't&nbsp;inter work&nb=
sp;in this scenario.&nbsp;</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">This is my understanding. Others please share your opinion on=
 the same....<br>
<br>
Thanks,</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Courier New&quot;;c=
olor:#000099">Nataraju A B</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Apr 18, 2012 at 12:32 PM, Ravindran, Parthas=
arathi &lt;<a href=3D"mailto:pravindran@sonusnet.com">pravindran@sonusnet.c=
om</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Nataraju,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">As per Sec 18 of RFC 3261 in , &#8220;<=
/span>All SIP elements MUST implement UDP and TCP.&#8221;,
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1F497D">so it is possible to have stateless or transacti=
on stateful proxy routing with change in transport mechanism from UDP to TC=
P and vice versa. The simple scenario like INVITE transaction
 over TCP whereas Re-INVITE over UDP based on contact SIP URI with UDP as a=
 transport should work as per RFC 3261. &nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">In case of draft-ibc-sipcore-sip-websoc=
ket-02 draft, SIP UA over websocket (like browser) will not
 implement either UDP or TCP transport which is a deviation from RFC 3261. =
This deviation leads to this limitation of not having stateless or Transact=
ion stateful proxy in websocket transport.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Thanks</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">Partha</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&q=
uot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></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=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,=
&quot;sans-serif&quot;">From:</span></b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju A.B [mailto:<=
a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gma=
il.com</a>]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 11:48 AM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> SIPCORE Chairs; SIPCORE (Session Initiation Protocol Core) WG<br=
>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Partha,&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">I also feel the same. The scenario is very similar to proxy interf=
acing between UDP and TCP on different sides, where dialog stateful proxy f=
unctionality is required. stateless
 / transaction stateful proxies would not work properly when requests are s=
ent over different protocols.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Thanks,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">Nataraju A B<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Wed, Apr 18, 2012 at 10:31 AM, Ravindran, Parthasarathi &lt;<a =
href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pravindran@sonusn=
et.com</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">As per my current reading, The current draft assumes about dialog-=
stateful proxy wherein all record-route header will be added whenever Webso=
cket to other transport routing happens
 in the proxy and proxy will be involved in all the transaction within the =
dialog. So, it is not possible to have stateless or transaction stateful pr=
oxy. It will be good in case this limitation is explicitly mentioned in the=
 draft.<br>
<br>
Thanks<br>
Partha<br>
<br>
&gt;-----Original Message-----<br>
&gt;From: <a href=3D"mailto:sipcore-bounces@ietf.org" target=3D"_blank">sip=
core-bounces@ietf.org</a> [mailto:<a href=3D"mailto:sipcore-bounces@ietf.or=
g" target=3D"_blank">sipcore-bounces@ietf.org</a>] On<br>
&gt;Behalf Of Adam Roach - SIPCORE Chair<br>
&gt;Sent: Monday, April 16, 2012 11:33 PM<br>
&gt;To: SIPCORE (Session Initiation Protocol Core) WG<br>
&gt;Subject: [sipcore] Call for Adoption: draft-ibc-sipcore-sip-websocket-0=
2<br>
&gt;<br>
&gt;[as chair]<br>
&gt;<br>
&gt;SIPCORE Working Group:<br>
&gt;<br>
&gt;This is a call for adoption of the draft-ibc-sipcore-sip-websocket-02<b=
r>
&gt;document to satisfy our recently added milestone to define a WebSockets=
<br>
&gt;transport for the SIP protocol. Interested parties should voice their<b=
r>
&gt;support and/or concerns on the SIPCORE mailing list. The chairs plan to=
<br>
&gt;make a determination of consensus on Friday, April 27th.<br>
&gt;<br>
&gt;Thank you. The draft announcement follows.<br>
&gt;<br>
&gt;/a<br>
&gt;<br>
&gt;-------- Original Message --------<br>
&gt;Subject: [sipcore] New Version Notification for draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02.txt<br>
&gt;Date: Mon, 16 Apr 2012 18:49:38 &#43;0200<br>
&gt;From: I=F1aki Baz Castillo &lt;<a href=3D"mailto:ibc@aliax.net" target=
=3D"_blank">ibc@aliax.net</a>&gt;<br>
&gt;To: SIPCORE WG &lt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank=
">sipcore@ietf.org</a>&gt;<br>
&gt;<br>
&gt;A new version of I-D, draft-ibc-sipcore-sip-websocket-02.txt has been<b=
r>
&gt;successfully submitted by I=F1aki Baz Castillo and posted to the IETF<b=
r>
&gt;repository.<br>
&gt;<br>
&gt;Filename: &nbsp; &nbsp; &nbsp; &nbsp;draft-ibc-sipcore-sip-websocket<br=
>
&gt;Revision: &nbsp; &nbsp; &nbsp; &nbsp;02<br>
&gt;Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; The WebSocket Protocol as a T=
ransport for the Session<br>
&gt;Initiation Protocol (SIP)<br>
&gt;Creation date: &nbsp; 2012-04-16<br>
&gt;WG ID: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Individual Submission<br>
&gt;Number of pages: 20<br>
&gt;<br>
&gt;Abstract:<br>
&gt; &nbsp; The WebSocket protocol enables two-way realtime communication b=
etween<br>
&gt; &nbsp; clients and servers. &nbsp;This document specifies a new WebSoc=
ket sub-<br>
&gt; &nbsp; protocol as a reliable transport mechanism between SIP (Session=
<br>
&gt; &nbsp; Initiation Protocol) entities and enables usage of the SIP prot=
ocol<br>
&gt; &nbsp; in new scenarios.<br>
&gt;<br>
&gt;<br>
&gt;- <a href=3D"http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket=
-02" target=3D"_blank">
http://tools.ietf.org/html/draft-ibc-sipcore-sip-websocket-02</a><br>
&gt;- <a href=3D"http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-0=
2.txt" target=3D"_blank">
http://tools.ietf.org/id/draft-ibc-sipcore-sip-websocket-02.txt</a><br>
&gt;<br>
&gt;<br>
&gt;This new version incorporates improvements and changes based on the<br>
&gt;comments and reviews on the previous version, along with the consensus<=
br>
&gt;got in IETF 83 Paris.<br>
&gt;<br>
&gt;<br>
&gt;As usual, comments are welcome. Thanks a lot.<br>
&gt;<br>
&gt;<br>
&gt;--<br>
&gt;I=F1aki Baz Castillo<br>
&gt;&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a=
>&gt;<br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
/a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
&gt;_______________________________________________<br>
&gt;sipcore mailing list<br>
&gt;<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org<=
/a><br>
&gt;<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/sipcore</a><br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">--
<br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;=
;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

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Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 10:44:50 +0200
Message-ID: <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>, "Nataraju A.B" <nataraju.sip@gmail.com>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> anybody implement SIP over websocket in browser using JavaScript violates=
 RFC 3261 as it is not
> possible for browser to send REINVITE/BYE using UDP/TCP transport towards=
 UAS directly with
> stateless/Transaction stateful proxy as a first proxy hop.

Then SIP UAs in Outbound (RFC 5626) scenarios also violate RFC 3261
since TCP UAs behind NAT cannot receive requests and responses via
UDP, neither they can receive inbound TCP connections.

But what is the exact problem? the draft is not for "web browsers",
but for any SIP entity running a WebSocket client and/or server (along
with other SIP transports).

Does a SIP stack in JavaScrtip running in a web browser and making
usage of WS transport violate RFC 3261 since it does not support
UDP/TCP transports? probably. Is that a real problem? not at all if
the SIP infrastucture applies other SIP standards (such as RFC 5626)
or "propietary mechanisms" for ensuring that the web browser sends and
receives all the SIP requests and responses through its opened
WebSocket connection.


Best regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 10:46:18 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> Inaki,
>
> Please provide SIP over Webexample example without record-route and 2xx r=
esponse from UAS with UDP/TCP transport in contact header, then it will be =
obvious why it is not possible to be transaction stateful proxy for Websock=
et to UDP/TCP transport conversion.

Ravindran, please read my previous mail in which I explain that your
concept of "transaction stateful proxy" is incorrect.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 10:59:00 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> One minor correction, I agree that dialog-stateful proxy is not mandatory=
 but transaction stateful proxy with record-route header is mandatory for w=
ebsocket transport

That is just true in case the UA just implements a WebSocket client
(so it's a SIP WebSocket Client according to "Definitions" section in
the draft, but not a SIP WebSocket Server). This is true in the case
of a web browser but the draft does NOT assume that clients are web
browsers.


> and stateless proxy is not possible.

In a previous mail I've explained that a staless proxy can bridge
different SIP transports. Please find a reference for your assertion
(I could be wrong).



> This has to be mentioned in the draft.

Please check RFC 4168 (SIP SCTP transport), and let's imagine this scenario=
:

  alice ----(SCTP)----- proxy -----(UDP)---- bob

If bob does not implement SCTP transport (and alice announces a URI
with ;transport=3Dstcp in her Contact), then the proxy needs to add
Record-Route (loose-routing) thinkgs to work, am I right?

Well, please look in RFC 4168 for keywords "proxy", "loose-routing",
"record-route", "stateless", "stateful" and so. No matches. So, why
should this draft solve all the problems of the world related to SIPS
or transport bridging? That is not the purpose of this draft which
just defines a new SIP transport. The issues you mean are not specific
for this new transport, but for any other transport which is not UDP
or TCP. There should be other drafts or places for solving or defining
procedures for them.



Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Wed Apr 18 02:08:35 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 11:08:14 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> In case the intention of the draft is to violate RFC 3261,
> IMO, we will need more analysis before adopting this draft.

Thanks a lot Partha.

Please let us know where the draft violates RFC 3261. If you assume
that a SIP UA implementing the WebSocket transport is a web browser
(so can only make outbound connections and does not speak SIP UDP/TCP)
then that is your assumption, but NOT the assumption of the draft.

Now imagine that I code a SIP softphone for my smartphone, and such a
softphone speaks SIP over UDP, TCP and WebSocket, and implements both
a WebSocket client and a WebSocket server (so becomes a SIP WebSocket
Client and a SIP WebSocket Server according to "Definitions" section
in the draft). Does it violate RFC 3261?

Best regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi,

On Apr 18, 2012, at 10:57 AM, Ravindran, Parthasarathi wrote:

> In case the intention of the draft is to violate RFC 3261,=20
> IMO, we will need more analysis before adopting this draft.
>=20

SIP clients implementing this draft may also implement UDP and TCP =
transports, this draft doesn't say they shouldn't.

Now lets assume we are using a JS SIP stack in a browser: UDP and TCP =
can't be used, so why implement them in the first place? Also, the =
client will register with WS transport and *use* transport=3Dws in the =
Contat header so trying to reach him on TCP or UDP would be incorrect.

Also, TCP and UDP are mandatory to implement as per RFC3261, not =
mandatory to use.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:

> As per your below statement, JavaScript with SIP and RTCWeb Browser =
MUST NOT use this draft as it violates RFC 3261.
>=20

It doesn't, see my previous email explaining why.

> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket =
will not listen for the incoming call without registration but in case =
of your draft is written with SIP softphone (smartphone) in mind , it is =
possible to listen in Port 80 or any other Websocket port which is not =
addressed in this draft till now. We need to discuss and sort it out.
>=20

The fact that a JS SIP client (running in a web browser) does not listen =
for incoming connections is a limitation imposed by the WebSocket API, =
not by this draft. This specification defines 2 entities: a SIP =
WebSocket Client and a SIP WebSocket Server, so nobody stops you from =
creating an entity that implements both and can accept incoming =
WebSocket connections. Of course, such an application would not run in a =
browser.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>, "SIPCORE (Session Initiation Protocol Core) WG" <sipcore@ietf.org>, "SIPCORE Chairs" <sipcore-chairs@tools.ietf.org>
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227FFD@inba-mail01.sonusnet.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Even if it is the limitations of websocket, SIP UA MUST listen in
UDP & TCP as per RFC 3261 and not adhere to this requirement which MUST
be considered as a violation of RFC 3261. In case you think that this draft
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not
clear enough, Please mention it explicitly in the draft that

"As per this specification, SIP UA MUST support UDP, TCP transport apart=20
from Websocket as a transport"

This information helps for the implementers who wish to use this draft for=
=20
RTCWeb client to SIP Interop in the standard way. I'm writing this mail as=
=20
most of the folks support this draft for interworking RTCWeb client to SIP=
=20
in the standard manner as per my discussion in IETF-83.=20

I'll write the separate mail thread for discussing incoming SIP over websoc=
ket
as it is one another major open item in this draft.

Thanks
Partha

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>Sent: Wednesday, April 18, 2012 3:23 PM
>To: Ravindran, Parthasarathi
>Cc: I=F1aki Baz Castillo; SIPCORE (Session Initiation Protocol Core) WG;
>SIPCORE Chairs; Nataraju A.B
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>
>On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:
>
>> As per your below statement, JavaScript with SIP and RTCWeb Browser
>MUST NOT use this draft as it violates RFC 3261.
>>
>
>It doesn't, see my previous email explaining why.
>
>> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket
>will not listen for the incoming call without registration but in case
>of your draft is written with SIP softphone (smartphone) in mind , it is
>possible to listen in Port 80 or any other Websocket port which is not
>addressed in this draft till now. We need to discuss and sort it out.
>>
>
>The fact that a JS SIP client (running in a web browser) does not listen
>for incoming connections is a limitation imposed by the WebSocket API,
>not by this draft. This specification defines 2 entities: a SIP
>WebSocket Client and a SIP WebSocket Server, so nobody stops you from
>creating an entity that implements both and can accept incoming
>WebSocket connections. Of course, such an application would not run in a
>browser.
>
>
>Regards,
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>


From saul@ag-projects.com  Wed Apr 18 03:14:29 2012
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227FFD@inba-mail01.sonusnet.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On Apr 18, 2012, at 12:07 PM, Ravindran, Parthasarathi wrote:

> Even if it is the limitations of websocket, SIP UA MUST listen in
> UDP & TCP as per RFC 3261 and not adhere to this requirement which =
MUST
> be considered as a violation of RFC 3261. In case you think that this =
draft
> does not violate any RFC 3261 for SIP UA, the Current text in the =
draft is not
> clear enough, Please mention it explicitly in the draft that
>=20

"MUST implement" doesn't imply "MUST use" AFAIK.

> "As per this specification, SIP UA MUST support UDP, TCP transport =
apart=20
> from Websocket as a transport"
>=20

It's not needed, 3261 sec 18 covers this: " All SIP elements MUST =
implement UDP and TCP.  SIP elements MAY implement other protocols."

> This information helps for the implementers who wish to use this draft =
for=20
> RTCWeb client to SIP Interop in the standard way. I'm writing this =
mail as=20
> most of the folks support this draft for interworking RTCWeb client to =
SIP=20
> in the standard manner as per my discussion in IETF-83.=20
>=20
> I'll write the separate mail thread for discussing incoming SIP over =
websocket
> as it is one another major open item in this draft.
>=20

I don't think it's a "major open item", the draft specifies it, but t =
can't be used in a RTCWeb context because browsers don't listen for =
incoming WebSocket connections. Other endpoints implementing this draft =
will be able to listen for incoming connections just fine.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From nataraju.sip@gmail.com  Wed Apr 18 03:20:03 2012
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Date: Wed, 18 Apr 2012 15:50:01 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--f46d04426822e5b69c04bdf16561
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On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

> Even if it is the limitations of websocket, SIP UA MUST listen in
> UDP & TCP as per RFC 3261 and not adhere to this requirement which MUST
> be considered as a violation of RFC 3261. In case you think that this dra=
ft
> does not violate any RFC 3261 for SIP UA, the Current text in the draft i=
s
> not
> clear enough, Please mention it explicitly in the draft that
>
> "As per this specification, SIP UA MUST support UDP, TCP transport apart
> from Websocket as a transport"
>
> This information helps for the implementers who wish to use this draft fo=
r
> RTCWeb client to SIP Interop in the standard way. I'm writing this mail a=
s
> most of the folks support this draft for interworking RTCWeb client to SI=
P
> in the standard manner as per my discussion in IETF-83.
>
> I'll write the separate mail thread for discussing incoming SIP over
> websocket
> as it is one another major open item in this draft.
>
[ABN] Partha,
          Can you name some scenario which need this incoming SIP requests
@ the WebSocket client ????

>
> Thanks
> Partha
>
> >-----Original Message-----
> >From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
> >Sent: Wednesday, April 18, 2012 3:23 PM
> >To: Ravindran, Parthasarathi
> >Cc: I=F1aki Baz Castillo; SIPCORE (Session Initiation Protocol Core) WG;
> >SIPCORE Chairs; Nataraju A.B
> >Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
> >in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
> >websocket-02]
> >
> >
> >On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:
> >
> >> As per your below statement, JavaScript with SIP and RTCWeb Browser
> >MUST NOT use this draft as it violates RFC 3261.
> >>
> >
> >It doesn't, see my previous email explaining why.
> >
> >> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket
> >will not listen for the incoming call without registration but in case
> >of your draft is written with SIP softphone (smartphone) in mind , it is
> >possible to listen in Port 80 or any other Websocket port which is not
> >addressed in this draft till now. We need to discuss and sort it out.
> >>
> >
> >The fact that a JS SIP client (running in a web browser) does not listen
> >for incoming connections is a limitation imposed by the WebSocket API,
> >not by this draft. This specification defines 2 entities: a SIP
> >WebSocket Client and a SIP WebSocket Server, so nobody stops you from
> >creating an entity that implements both and can accept incoming
> >WebSocket connections. Of course, such an application would not run in a
> >browser.
> >
> >
> >Regards,
> >
> >--
> >Sa=FAl Ibarra Corretg=E9
> >AG Projects
> >
> >
>
>


--=20
Thanks,
Nataraju A.B.

--f46d04426822e5b69c04bdf16561
Content-Type: text/html; charset=ISO-8859-1
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<br><br><div class=3D"gmail_quote">On Wed, Apr 18, 2012 at 3:37 PM, Ravindr=
an, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusn=
et.com">pravindran@sonusnet.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
Even if it is the limitations of websocket, SIP UA MUST listen in<br>
UDP &amp; TCP as per RFC 3261 and not adhere to this requirement which MUST=
<br>
be considered as a violation of RFC 3261. In case you think that this draft=
<br>
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not<br>
clear enough, Please mention it explicitly in the draft that<br>
<br>
&quot;As per this specification, SIP UA MUST support UDP, TCP transport apa=
rt<br>
from Websocket as a transport&quot;<br>
<br>
This information helps for the implementers who wish to use this draft for<=
br>
RTCWeb client to SIP Interop in the standard way. I&#39;m writing this mail=
 as<br>
most of the folks support this draft for interworking RTCWeb client to SIP<=
br>
in the standard manner as per my discussion in IETF-83.<br>
<br>
I&#39;ll write the separate mail thread for discussing incoming SIP over we=
bsocket<br>
as it is one another major open item in this draft.<br></blockquote><div>[A=
BN] Partha,=A0</div><div>=A0 =A0 =A0 =A0 =A0 Can you name some scenario whi=
ch need this incoming SIP requests @ the WebSocket client ????=A0</div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">

<br>
Thanks<br>
Partha<br>
<div class=3D"im"><br>
&gt;-----Original Message-----<br>
&gt;From: Sa=FAl Ibarra Corretg=E9 [mailto:<a href=3D"mailto:saul@ag-projec=
ts.com">saul@ag-projects.com</a>]<br>
&gt;Sent: Wednesday, April 18, 2012 3:23 PM<br>
&gt;To: Ravindran, Parthasarathi<br>
</div><div class=3D"im">&gt;Cc: I=F1aki Baz Castillo; SIPCORE (Session Init=
iation Protocol Core) WG;<br>
&gt;SIPCORE Chairs; Nataraju A.B<br>
&gt;Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=
<br>
&gt;in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02]<br>
&gt;<br>
&gt;<br>
</div><div><div></div><div class=3D"h5">&gt;On Apr 18, 2012, at 11:35 AM, R=
avindran, Parthasarathi wrote:<br>
&gt;<br>
&gt;&gt; As per your below statement, JavaScript with SIP and RTCWeb Browse=
r<br>
&gt;MUST NOT use this draft as it violates RFC 3261.<br>
&gt;&gt;<br>
&gt;<br>
&gt;It doesn&#39;t, see my previous email explaining why.<br>
&gt;<br>
&gt;&gt; Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websoc=
ket<br>
&gt;will not listen for the incoming call without registration but in case<=
br>
&gt;of your draft is written with SIP softphone (smartphone) in mind , it i=
s<br>
&gt;possible to listen in Port 80 or any other Websocket port which is not<=
br>
&gt;addressed in this draft till now. We need to discuss and sort it out.<b=
r>
&gt;&gt;<br>
&gt;<br>
&gt;The fact that a JS SIP client (running in a web browser) does not liste=
n<br>
&gt;for incoming connections is a limitation imposed by the WebSocket API,<=
br>
&gt;not by this draft. This specification defines 2 entities: a SIP<br>
&gt;WebSocket Client and a SIP WebSocket Server, so nobody stops you from<b=
r>
&gt;creating an entity that implements both and can accept incoming<br>
&gt;WebSocket connections. Of course, such an application would not run in =
a<br>
&gt;browser.<br>
&gt;<br>
&gt;<br>
&gt;Regards,<br>
&gt;<br>
&gt;--<br>
&gt;Sa=FAl Ibarra Corretg=E9<br>
&gt;AG Projects<br>
&gt;<br>
&gt;<br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>

--f46d04426822e5b69c04bdf16561--

From pravindran@sonusnet.com  Wed Apr 18 03:27:12 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Date: Wed, 18 Apr 2012 10:26:58 +0000
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227FFD@inba-mail01.sonusnet.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com> <FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Saul,

I guess that you wish to see the requirement like

"As per this specification, SIP UA MUST implement UDP, TCP transport
apart from Websocket as a transport"

Please note that my intention of the this mail thread is not to
create this requirement but to clarify the implementers about proxy usage.

Thanks
Partha=20

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>Sent: Wednesday, April 18, 2012 3:44 PM
>To: Ravindran, Parthasarathi
>Cc: SIPCORE (Session Initiation Protocol Core) WG; SIPCORE Chairs; I=F1aki
>Baz Castillo; Nataraju A.B
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>
>On Apr 18, 2012, at 12:07 PM, Ravindran, Parthasarathi wrote:
>
>> Even if it is the limitations of websocket, SIP UA MUST listen in UDP
>> & TCP as per RFC 3261 and not adhere to this requirement which MUST be
>> considered as a violation of RFC 3261. In case you think that this
>> draft does not violate any RFC 3261 for SIP UA, the Current text in
>> the draft is not clear enough, Please mention it explicitly in the
>> draft that
>>
>
>"MUST implement" doesn't imply "MUST use" AFAIK.
>
>> "As per this specification, SIP UA MUST support UDP, TCP transport
>> apart from Websocket as a transport"
>>
>
>It's not needed, 3261 sec 18 covers this: " All SIP elements MUST
>implement UDP and TCP.  SIP elements MAY implement other protocols."
>
>> This information helps for the implementers who wish to use this draft
>> for RTCWeb client to SIP Interop in the standard way. I'm writing this
>> mail as most of the folks support this draft for interworking RTCWeb
>> client to SIP in the standard manner as per my discussion in IETF-83.
>>
>> I'll write the separate mail thread for discussing incoming SIP over
>> websocket as it is one another major open item in this draft.
>>
>
>I don't think it's a "major open item", the draft specifies it, but t
>can't be used in a RTCWeb context because browsers don't listen for
>incoming WebSocket connections. Other endpoints implementing this draft
>will be able to listen for incoming connections just fine.
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>


From nataraju.sip@gmail.com  Wed Apr 18 03:29:31 2012
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Date: Wed, 18 Apr 2012 15:59:28 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Content-Type: multipart/alternative; boundary=f46d0408d3bfbad11204bdf187b4
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--f46d0408d3bfbad11204bdf187b4
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On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

> Even if it is the limitations of websocket, SIP UA MUST listen in
> UDP & TCP as per RFC 3261 and not adhere to this requirement which MUST
> be considered as a violation of RFC 3261. In case you think that this dra=
ft
> does not violate any RFC 3261 for SIP UA, the Current text in the draft i=
s
> not
> clear enough, Please mention it explicitly in the draft that
>
> [ABN] It may be overkill to ask Websocket client to support for TCP and
UDP and listen for incoming connections, just to be inline / compatible
with SIP.

          is it worth doing so ???


> "As per this specification, SIP UA MUST support UDP, TCP transport apart
> from Websocket as a transport"
>
> This information helps for the implementers who wish to use this draft fo=
r
> RTCWeb client to SIP Interop in the standard way. I'm writing this mail a=
s
> most of the folks support this draft for interworking RTCWeb client to SI=
P
> in the standard manner as per my discussion in IETF-83.
>
> I'll write the separate mail thread for discussing incoming SIP over
> websocket
> as it is one another major open item in this draft.
>
> Thanks
> Partha
>
> >-----Original Message-----
> >From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
> >Sent: Wednesday, April 18, 2012 3:23 PM
> >To: Ravindran, Parthasarathi
> >Cc: I=F1aki Baz Castillo; SIPCORE (Session Initiation Protocol Core) WG;
> >SIPCORE Chairs; Nataraju A.B
> >Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
> >in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
> >websocket-02]
> >
> >
> >On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:
> >
> >> As per your below statement, JavaScript with SIP and RTCWeb Browser
> >MUST NOT use this draft as it violates RFC 3261.
> >>
> >
> >It doesn't, see my previous email explaining why.
> >
> >> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket
> >will not listen for the incoming call without registration but in case
> >of your draft is written with SIP softphone (smartphone) in mind , it is
> >possible to listen in Port 80 or any other Websocket port which is not
> >addressed in this draft till now. We need to discuss and sort it out.
> >>
> >
> >The fact that a JS SIP client (running in a web browser) does not listen
> >for incoming connections is a limitation imposed by the WebSocket API,
> >not by this draft. This specification defines 2 entities: a SIP
> >WebSocket Client and a SIP WebSocket Server, so nobody stops you from
> >creating an entity that implements both and can accept incoming
> >WebSocket connections. Of course, such an application would not run in a
> >browser.
> >
> >
> >Regards,
> >
> >--
> >Sa=FAl Ibarra Corretg=E9
> >AG Projects
> >
> >
>
>


--=20
Thanks,
Nataraju A.B.

--f46d0408d3bfbad11204bdf187b4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_quote">On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, =
Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusnet.c=
om">pravindran@sonusnet.com</a>&gt;</span> wrote:<br><blockquote class=3D"g=
mail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-l=
eft:1ex">
Even if it is the limitations of websocket, SIP UA MUST listen in<br>
UDP &amp; TCP as per RFC 3261 and not adhere to this requirement which MUST=
<br>
be considered as a violation of RFC 3261. In case you think that this draft=
<br>
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not<br>
clear enough, Please mention it explicitly in the draft that<br>
<br></blockquote><div>[ABN] It may be overkill to ask Websocket client to s=
upport for TCP and UDP and listen for=A0incoming=A0connections, just to be =
inline / compatible with SIP.</div><div><br></div><div>=A0 =A0 =A0 =A0 =A0 =
is it worth doing so ???</div>
<div>=A0 =A0 =A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0=
 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
&quot;As per this specification, SIP UA MUST support UDP, TCP transport apa=
rt<br>
from Websocket as a transport&quot;<br>
<br>
This information helps for the implementers who wish to use this draft for<=
br>
RTCWeb client to SIP Interop in the standard way. I&#39;m writing this mail=
 as<br>
most of the folks support this draft for interworking RTCWeb client to SIP<=
br>
in the standard manner as per my discussion in IETF-83.<br>
<br>
I&#39;ll write the separate mail thread for discussing incoming SIP over we=
bsocket<br>
as it is one another major open item in this draft.<br>
<br>
Thanks<br>
Partha<br>
<div class=3D"im"><br>
&gt;-----Original Message-----<br>
&gt;From: Sa=FAl Ibarra Corretg=E9 [mailto:<a href=3D"mailto:saul@ag-projec=
ts.com">saul@ag-projects.com</a>]<br>
&gt;Sent: Wednesday, April 18, 2012 3:23 PM<br>
&gt;To: Ravindran, Parthasarathi<br>
</div><div class=3D"im">&gt;Cc: I=F1aki Baz Castillo; SIPCORE (Session Init=
iation Protocol Core) WG;<br>
&gt;SIPCORE Chairs; Nataraju A.B<br>
&gt;Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=
<br>
&gt;in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02]<br>
&gt;<br>
&gt;<br>
</div><div><div></div><div class=3D"h5">&gt;On Apr 18, 2012, at 11:35 AM, R=
avindran, Parthasarathi wrote:<br>
&gt;<br>
&gt;&gt; As per your below statement, JavaScript with SIP and RTCWeb Browse=
r<br>
&gt;MUST NOT use this draft as it violates RFC 3261.<br>
&gt;&gt;<br>
&gt;<br>
&gt;It doesn&#39;t, see my previous email explaining why.<br>
&gt;<br>
&gt;&gt; Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websoc=
ket<br>
&gt;will not listen for the incoming call without registration but in case<=
br>
&gt;of your draft is written with SIP softphone (smartphone) in mind , it i=
s<br>
&gt;possible to listen in Port 80 or any other Websocket port which is not<=
br>
&gt;addressed in this draft till now. We need to discuss and sort it out.<b=
r>
&gt;&gt;<br>
&gt;<br>
&gt;The fact that a JS SIP client (running in a web browser) does not liste=
n<br>
&gt;for incoming connections is a limitation imposed by the WebSocket API,<=
br>
&gt;not by this draft. This specification defines 2 entities: a SIP<br>
&gt;WebSocket Client and a SIP WebSocket Server, so nobody stops you from<b=
r>
&gt;creating an entity that implements both and can accept incoming<br>
&gt;WebSocket connections. Of course, such an application would not run in =
a<br>
&gt;browser.<br>
&gt;<br>
&gt;<br>
&gt;Regards,<br>
&gt;<br>
&gt;--<br>
&gt;Sa=FAl Ibarra Corretg=E9<br>
&gt;AG Projects<br>
&gt;<br>
&gt;<br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>

--f46d0408d3bfbad11204bdf187b4--

From saul@ag-projects.com  Wed Apr 18 03:35:30 2012
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To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>, SIPCORE Chairs <sipcore-chairs@tools.ietf.org>, "Nataraju A.B" <nataraju.sip@gmail.com>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Partha,

On Apr 18, 2012, at 12:26 PM, Ravindran, Parthasarathi wrote:

> Saul,
>=20
> I guess that you wish to see the requirement like
>=20
> "As per this specification, SIP UA MUST implement UDP, TCP transport
> apart from Websocket as a transport"
>=20
> Please note that my intention of the this mail thread is not to
> create this requirement but to clarify the implementers about proxy =
usage.
>=20

No, I don't wish to see any of that. Have a look at RFC4168, it defines =
SCTP and it doesn't say any of that, it's perfectly clear that that =
specification (and this one) define *new* transports for the SIP =
protocol. I don't see a need to remind people of what RFC3261 says.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 13:37:16 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> I guess that you wish to see the requirement like
>
> "As per this specification, SIP UA MUST implement UDP, TCP transport
> apart from Websocket as a transport"

Please check RFC 4168 (SIP SCTP transport) and find a similar statement.


> Please note that my intention of the this mail thread is not to
> create this requirement but to clarify the implementers about proxy usage=
.

Thanks for clarifying it. If not, someone might think that what you
claim is that a web browser can not be SIP compliant since it does not
speak SIP UDP/TCP.


Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 4/18/12 4:40 PM, Christer Holmberg wrote:
> Hi,
>
> I may have misunderstood something in the conversation, but my suggestion would be to consider *NOT* mandating SIP WebSocket clients to implement SIP UDP and TCP.
>
> Especially if I write a JavaScript SIP browser app, I very likely would want to support only WebSocket.
>
> Also, I don't think we can directly compare with SCTP, since the typical use-case for WebSocket is browser apps (eventhough a native app can of course also use WS).

I think that in general we can not compare SCTP to WebSocket at all
because the first is a pure Transport Protocol, the latter is an 
application protocol that is used as transport.

 From a pure WebSocket prospective, what you are going to define is SIP 
as a websocket sub-protocol that is transported on top of.

The important thing to standardize from a SIPCore prospective is the 
interaction/translations that happens between the WebSocket server and 
the SIP proxy
that are two different logical entities.

my two cents
Sal

-- 
Salvatore Loreto, PhD
www.sloreto.com


From saul@ag-projects.com  Wed Apr 18 07:24:11 2012
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To: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi,

On Apr 18, 2012, at 4:11 PM, Salvatore Loreto wrote:

> On 4/18/12 4:40 PM, Christer Holmberg wrote:
>> Hi,
>>=20
>> I may have misunderstood something in the conversation, but my =
suggestion would be to consider *NOT* mandating SIP WebSocket clients to =
implement SIP UDP and TCP.
>>=20
>> Especially if I write a JavaScript SIP browser app, I very likely =
would want to support only WebSocket.
>>=20
>> Also, I don't think we can directly compare with SCTP, since the =
typical use-case for WebSocket is browser apps (eventhough a native app =
can of course also use WS).
>=20
> I think that in general we can not compare SCTP to WebSocket at all
> because the first is a pure Transport Protocol, the latter is an =
application protocol that is used as transport.
>=20
> =46rom a pure WebSocket prospective, what you are going to define is =
SIP as a websocket sub-protocol that is transported on top of.
>=20
> The important thing to standardize from a SIPCore prospective is the =
interaction/translations that happens between the WebSocket server and =
the SIP proxy
> that are two different logical entities.
>=20

Sorry about the confusion, the only reason why I mentioned SCTP was to =
illustrate that the suggested additions weren't IMHO necessary as they =
weren't for SCTP. Just as an example.

As I mentioned earlier, I would not mention anything regarding UDP and =
TCP, 3261 applies. Even if browsers are not able to use TCP and UDP =
directly I don't see a strong reason to forbid them.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 16:31:59 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
> As I mentioned earlier, I would not mention anything regarding UDP and TC=
P, 3261 applies. Even if browsers are not able to use TCP and UDP directly =
I don't see a strong reason to forbid them.

Agreed. Let's imagine a SIP local network in which the administrator
just wants SIP over TLS-TCP for security/privacy reasons, so SIP
phones and the SIP proxy/PBX/server just talk TLS-TCP transport. Would
we say that those SIP phones are violating the RFC 3261 since they are
not listening into insecure UDP transport? I don't think so.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 16:24:13 +0200
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To: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Salvatore Loreto <salvatore.loreto@ericsson.com>:
> The important thing to standardize from a SIPCore prospective is the
> interaction/translations that happens between the WebSocket server and th=
e
> SIP proxy that are two different logical entities.

Hi Salvatore, I don't consider them two different logical entities.
The draft 02 states the following in "Definitions" section (which was
added after your review):

   SIP WebSocket Server:  A SIP entity capable of listening for inbound
         connections from WebSocket clients and speaking the WebSocket
         SIP Sub-Protocol as defined by this document.


So when the WebSocket message/arrives to the WebSocket listening
server, the application on top of it (a SIP proxy/UAS) should extract
the SIP message from the WS message, parse it and process it in the
same way that if the SIP message arrived via another SIP transport.
Just it. And for sending responses the SIP proxy/UAS should reuse the
existing WS connection (which implies creating a WS message and
writting the SIP response inside). Just it.


The same is true for a SIP WebSocket Client:

   SIP WebSocket Client:  A SIP entity capable of opening outbound
         connections with WebSocket servers and speaking the WebSocket
         SIP Sub-Protocol as defined by this document.


For example, if the SIP application is a JavaScript code running on a
web browser, the SIP stack (JS code) sends and receives SIP messages
to/from its transport layer (as in any other SIP transport), but in
this case the transport layer is a WebSocket client interface which
generates a WS message (using the JS WebSocket API) and sends it. Same
occurs when receiving an incoming SIP request/response which is
extracted by the JS app (the SIP stack) from the received WS message.


So honestly I see no "interaction/translations that happens between
the WebSocket server and the SIP proxy".


Best regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From pkyzivat@alum.mit.edu  Wed Apr 18 07:47:56 2012
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 4/18/12 4:23 AM, Ravindran, Parthasarathi wrote:

> Please note that SIP UA with TLS transport is not going to violate Sec
> 18 of RFC 3261 All SIP elements MUST implement UDP and TCP..
> In the below mentioned callflow, proxy shall
> acts as stateless or transaction stateful proxy wherein UAC shall sent
> INVITE with SIPS URI

I think we need to bite the bullet here and decide to revise/clarify 
that statement in 3261. There is a note following that statement that is 
useful context for understanding it:

    All SIP elements MUST implement UDP and TCP.  SIP elements MAY
    implement other protocols.

       Making TCP mandatory for the UA is a substantial change from RFC
       2543.  It has arisen out of the need to handle larger messages,
       which MUST use TCP, as discussed below.  Thus, even if an element
       never sends large messages, it may receive one and needs to be
       able to handle them.

Clearly there are some objectives that this normative statement is 
trying to solve:

- a basic level of protocol interoperability
- support for large messages

I think we can find a way to achieve those objectives while loosening 
the requirements to support the sorts of constrained implementation 
environments we are seeing.

Both the websocket transport and outbound rely upon a neighboring sip 
element to bridge the gap to interoperability. So rather than 
implementing UDP and TCP itself, the websocket client depends upon the 
websocket server to implement them on its behalf.

I think it will take a bit work to craft new language that provides the 
needed additional flexibility while not giving license to create 
implementations that lack basic interoperability.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Wed Apr 18 07:58:54 2012
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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There has already been a lot of discussion on this thread. But one 
fundamental point about the message below doesn't seem to have been 
mentioned. (Or did I miss it?) See inline.

On 4/18/12 3:28 AM, Nataraju A.B wrote:
> Partha,
>
> I was referring to scenario,
>
> UAC                  Proxy                   UAS
>   -----sips/tcp---->     ------>sip/udp --->
>
> UAC requested for a secure connection. But assume proxy at the edge and
> can't guarantee secure connection towards UAS. hence It is sending a
> request to UAS over simple udp *without security*.

This was allowed by 3261, but that behavior was repealed by rfc5630 
section 5.3.

	Thanks,
	Paul

> In this case, proxy must behave like a dialog stateful proxy. stateless
> or transaction stateful proxy can't inter work in this scenario.
>
> This is my understanding. Others please share your opinion on the same....
>
> Thanks,
> Nataraju A B


From pravindran@sonusnet.com  Wed Apr 18 08:10:56 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi Paul/Inaki,

We are in the same page that there is no need of UDP and TCP transport whil=
e websocket is used as a transport in SIP UA even though it is a deviation =
from RFC 3261. As Nataraj/Christer mentioned in the other mail threads, UDP=
 & TCP is an overkill for SIP UA while using SIP UA with websocket and non-=
starter for web-browsers with SIP based on Javascript.=20

Having said that, please indicate that Websocket (to any transport) Proxy M=
UST include record-route header in all SIP messages and it is a mandatory h=
eader for websocket proxy as per this specification. It may be preferable t=
o provide the reason for this mandating. IMO, This information helps websoc=
ket proxy implementers.  Also indicates that, stateless proxy is not applic=
able for this transport.

I like to hear in case any other way to make SIP over websocket works witho=
ut record-route header.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Paul Kyzivat
>Sent: Wednesday, April 18, 2012 8:18 PM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>On 4/18/12 4:23 AM, Ravindran, Parthasarathi wrote:
>
>> Please note that SIP UA with TLS transport is not going to violate Sec
>> 18 of RFC 3261 "All SIP elements MUST implement UDP and TCP.".
>> In the below mentioned callflow, proxy shall acts as stateless or
>> transaction stateful proxy wherein UAC shall sent INVITE with SIPS URI
>
>I think we need to bite the bullet here and decide to revise/clarify
>that statement in 3261. There is a note following that statement that is
>useful context for understanding it:
>
>    All SIP elements MUST implement UDP and TCP.  SIP elements MAY
>    implement other protocols.
>
>       Making TCP mandatory for the UA is a substantial change from RFC
>       2543.  It has arisen out of the need to handle larger messages,
>       which MUST use TCP, as discussed below.  Thus, even if an element
>       never sends large messages, it may receive one and needs to be
>       able to handle them.
>
>Clearly there are some objectives that this normative statement is
>trying to solve:
>
>- a basic level of protocol interoperability
>- support for large messages
>
>I think we can find a way to achieve those objectives while loosening
>the requirements to support the sorts of constrained implementation
>environments we are seeing.
>
>Both the websocket transport and outbound rely upon a neighboring sip
>element to bridge the gap to interoperability. So rather than
>implementing UDP and TCP itself, the websocket client depends upon the
>websocket server to implement them on its behalf.
>
>I think it will take a bit work to craft new language that provides the
>needed additional flexibility while not giving license to create
>implementations that lack basic interoperability.
>
>	Thanks,
>	Paul
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

From dworley@avaya.com  Wed Apr 18 08:48:34 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Shida Schubert <shida@ntt-at.com>
Date: Wed, 18 Apr 2012 11:48:28 -0400
Thread-Topic: [sipcore] draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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> From: Shida Schubert [shida@ntt-at.com]
>=20
> >         "np": The hi-targeted-to-URI represents that there was no
> >         change in Request-URI.  This would apply for example when a
> >         proxy merely forwards a request to a next hop proxy and loose
> >         routing is used.
> >
> > Unlike the previous two cases, this case does not describe the
> > function of the value of the np parameter.
>=20
> Any suggestion for rewording?

Assuming that the values of the three parameters are intended to carry
the same information (that is, the index of the hi-entry from which
this request-URI was derived), the value of "np" would be the index of
the parent hi-entry of this hi-entry.  As in the example in 5.1:

   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1

Largely copying the wording of the entry for "rc", we can add:

         "np": The hi-targeted-to-URI represents that there was no
         change in Request-URI.  This would apply for example when a
         proxy merely forwards a request to a next hop proxy and loose
         routing is used.
         >>>The "np"
         header field parameter contains the value of the hi-index in
         the hi-entry with an hi-targeted-to-uri that reflects the
         Request-URI that was copied unchanged into the request
         represented by this hi-entry.  That value will usually be the
         hi-index of the parent hi-entry of this hi-entry.<<<

While I was lookng at this, the entry for "mp" has some awkward
wording that could be improved by copying the wording from "rc".
Instead of

         The "mp" header field parameter contains the
         >>>value which represents the value<<< of the hi-index in the
         hi-entry ...

perhaps

         The "mp"
         header field parameter contains the >>>value<<< of the hi-index in
         the hi-entry ...

Dale

From ibc@aliax.net  Wed Apr 18 08:56:14 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 17:55:52 +0200
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:

> Having said that, please indicate that Websocket (to any transport) Proxy=
 MUST include record-route header in all SIP messages and it is a mandatory=
 header for websocket proxy as per this specification. It may be preferable=
 to provide the reason for this mandating. IMO, This information helps webs=
ocket proxy implementers.


Hi Ravindran. If a proxy behaves as an Outbound Edge Proxy (RFC 5626)
then it adds Record-Route always (since NATted clients can only
receive incoming requests from their Outbound proxy via TCP, UDP, or
WebSocket).

Implementing RFC 5626 "Outbound" was a MUST in previous draft version,
but such a requirement was removed in IETF #83 after consensus by
members of this WG, so the draft does NOT mandate RFC 5626 (which IMHO
is a good decision since it "just defines a new SIP transport" rather
than a complete SIP topology).

Anyhow, Record-Route (Loose-Routing mechanism as defined in RFC 3261)
is *NOT* required in *ALL* the cases in which a SIP WebSocket UA is
involved, for example:


- alice is a SIP UA implementing a SIP WS Client and a SIP WS Server
(so it can open and accept WS connections).

- The very same for bob, another SIP WS UA. But bob also speaks UDP
and registers to P1 via UDP.

- P1 is a proxy/registrar implementing a SIP WebSocket Server (so just
listens for inbound WS connections) and also UDP.

- P1 does NOT perform Loose-Routing (it does NOT add Record-Route headers).

- alice, bob and P1 are located in the same network (or have public IP...).


So:



      alice                     P1                    bob

F1    INVITE -----(WS)--------->
F2                                INVITE ----(UDP)------->

F3                                <---(UDP)----------- 200
F4     <-----(WS)----------- 200

F5    ACK ---------------------(WS)---------------------->

F6    <------------------------(WS)------------------- BYE
F7    200 ---------------------(WS)---------------------->


F1: INVITE from alice has a Contact URI with ;transport=3Dws.

F2. P1 does NOT add Record-Route header.

F3. 200 from bob contains a Contact URI with ;transport=3Dws (which is
valid regardless the INVITE arrived via UDP transport).

F5. alice opens a new WS connection to the Contact URI of bob (since
ACK must be send end to end when there is no Record-Route in 200) and
sends the ACK.

F7. bob opens a new WS connection to the Contact URI of alice and sends the=
 BYE.


This is perfectly valid according to RFC 3261 and there is NO need for
Loose-Routing mechanism, so the draft MUST NOT mandate Loose-Routing
mechanism.

Again, you assume that the SIP WebSocket UA is a web browser that can
only open outbound WS connections. The draft is not based on such an
assumption (regardless we all know that this will be the scenario in
99% of cases). We leave the SIP topology open (this was recently
discussed in this maillist before IETF #83).



> Also indicates that, stateless proxy is not applicable for this transport=
.

Please read my previous replies. Stateless proxy CAN be used (please
read my first reply in this thread in which I explain it).




> I like to hear in case any other way to make SIP over websocket works wit=
hout record-route header.

See example above.



Regards.




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From christer.holmberg@ericsson.com  Wed Apr 18 09:09:30 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Date: Wed, 18 Apr 2012 18:08:37 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi,

>> As I mentioned earlier, I would not mention anything regarding UDP and T=
CP, 3261 applies. Even if >>browsers are not able to use TCP and UDP direct=
ly I don't see a strong reason to forbid them.
>
> Agreed. Let's imagine a SIP local network in which the administrator
> just wants SIP over TLS-TCP for security/privacy reasons, so SIP
> phones and the SIP proxy/PBX/server just talk TLS-TCP transport. Would
> we say that those SIP phones are violating the RFC 3261 since they are
> not listening into insecure UDP transport? I don't think so.

I agree that we shouldn't forbid.

Regards,

Christer=

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 18:13:25 +0200
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> I think we can find a way to achieve those objectives while loosening the
> requirements to support the sorts of constrained implementation environme=
nts
> we are seeing.
>
> Both the websocket transport and outbound rely upon a neighboring sip
> element to bridge the gap to interoperability. So rather than implementin=
g
> UDP and TCP itself, the websocket client depends upon the websocket serve=
r
> to implement them on its behalf.
>
> I think it will take a bit work to craft new language that provides the
> needed additional flexibility while not giving license to create
> implementations that lack basic interoperability.

Hi Paul, the scenario is similar to the one in which a SIP UA behind
NAT registers using UDP through its Outbound Edge Proxy, so it can
only receive incoming requests (from outside its local network) via
its Outbound proxy. Neither it can receive inbound TCP connections.

So, perhaps we could extract some related text from RFC 5626? But note
that mandating RFC 5626 was removes in Paris due the got consensus.


However, this discussion assumes that the SIP UA is a SIP WebSocket
Client just capable of using WebSocket transport (so probably a web
browser), and that the SIP WebSocket Server is a SIP proxy which
cannot open outbound WS connections (and speaks other transports).
This assumption does not exist in the draft. If a UA or a proxy
implements both a SIP WebSocket Client and a SIP WebSocket Server then
all this discussion becomes "invalid" since topology constrains are
relaxed.

So:

- This draft does not assume network topology.
- This draft does not assume clients and proxies/servers nature.
- RFC 3261 mandates UDP and TCP.


JavaScripts SIP stacks running in browsers will not implement SIP UDP/TCP.
Are those SIP stacks "violating" RFC 3261? well, things change since
2002, and probably a SIP transport like WebSocket was not expected
when RFC 3261 was written. But anyhow, from the point of view of the
draft itself, IMHO the spec is not violating RFC 3261 since it does
not assume the "web browser" scenario.


Best regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From christer.holmberg@ericsson.com  Wed Apr 18 09:21:18 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Date: Wed, 18 Apr 2012 18:17:14 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi,

In general I agree that a transport mechanism specification shouldn't manda=
te entities to Record-Route.

But, as others have also indicated, in most cases it will be needed. The si=
gnalling path between the UA and the "outbound proxy" will be established a=
lready during registration (which means the proxy will also have to insert =
Path), and will then be used for UA originating and terminating sessions.

Regards,

Christer





________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of I=F1=
aki Baz Castillo [ibc@aliax.net]
Sent: Wednesday, April 18, 2012 6:55 PM
To: Ravindran, Parthasarathi
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:

> Having said that, please indicate that Websocket (to any transport) Proxy=
 MUST include record-route header in all SIP messages and it is a mandatory=
 header for websocket proxy as per this specification. It may be preferable=
 to provide the reason for this mandating. IMO, This information helps webs=
ocket proxy implementers.


Hi Ravindran. If a proxy behaves as an Outbound Edge Proxy (RFC 5626)
then it adds Record-Route always (since NATted clients can only
receive incoming requests from their Outbound proxy via TCP, UDP, or
WebSocket).

Implementing RFC 5626 "Outbound" was a MUST in previous draft version,
but such a requirement was removed in IETF #83 after consensus by
members of this WG, so the draft does NOT mandate RFC 5626 (which IMHO
is a good decision since it "just defines a new SIP transport" rather
than a complete SIP topology).

Anyhow, Record-Route (Loose-Routing mechanism as defined in RFC 3261)
is *NOT* required in *ALL* the cases in which a SIP WebSocket UA is
involved, for example:


- alice is a SIP UA implementing a SIP WS Client and a SIP WS Server
(so it can open and accept WS connections).

- The very same for bob, another SIP WS UA. But bob also speaks UDP
and registers to P1 via UDP.

- P1 is a proxy/registrar implementing a SIP WebSocket Server (so just
listens for inbound WS connections) and also UDP.

- P1 does NOT perform Loose-Routing (it does NOT add Record-Route headers).

- alice, bob and P1 are located in the same network (or have public IP...).


So:



      alice                     P1                    bob

F1    INVITE -----(WS)--------->
F2                                INVITE ----(UDP)------->

F3                                <---(UDP)----------- 200
F4     <-----(WS)----------- 200

F5    ACK ---------------------(WS)---------------------->

F6    <------------------------(WS)------------------- BYE
F7    200 ---------------------(WS)---------------------->


F1: INVITE from alice has a Contact URI with ;transport=3Dws.

F2. P1 does NOT add Record-Route header.

F3. 200 from bob contains a Contact URI with ;transport=3Dws (which is
valid regardless the INVITE arrived via UDP transport).

F5. alice opens a new WS connection to the Contact URI of bob (since
ACK must be send end to end when there is no Record-Route in 200) and
sends the ACK.

F7. bob opens a new WS connection to the Contact URI of alice and sends the=
 BYE.


This is perfectly valid according to RFC 3261 and there is NO need for
Loose-Routing mechanism, so the draft MUST NOT mandate Loose-Routing
mechanism.

Again, you assume that the SIP WebSocket UA is a web browser that can
only open outbound WS connections. The draft is not based on such an
assumption (regardless we all know that this will be the scenario in
99% of cases). We leave the SIP topology open (this was recently
discussed in this maillist before IETF #83).



> Also indicates that, stateless proxy is not applicable for this transport=
.

Please read my previous replies. Stateless proxy CAN be used (please
read my first reply in this thread in which I explain it).




> I like to hear in case any other way to make SIP over websocket works wit=
hout record-route header.

See example above.



Regards.




--
I=F1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From pkyzivat@alum.mit.edu  Wed Apr 18 09:58:52 2012
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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IĆ±aki,

I agree that there is nothing in this draft that requires someone 
implementing it to violate 3261.

However, as you mention below, a typical use case for this draft is 
javascript sip stacks running in a browser that would not be able to 
implement sip tcp and udp transports. AFAIK those would indeed be 
non-compliant to 3261. If we want to encourage such implementations, 
then we should do something so that they can be compliant. ISTM that 
this will require a revision to 3261. That could be done in a separate 
draft, or it could be done in this draft that is already revising 3261.

One disadvantage to doing it in this draft is that then compliance to 
the revised language would be advertised by advertising support for 
*this* document. If the relaxation is to apply to other implementations 
of sip that don't want to implement the websocket transport, then it 
should be done separately.

	Thanks,
	Paul

On 4/18/12 12:13 PM, IĆ±aki Baz Castillo wrote:
> 2012/4/18 Paul Kyzivat<pkyzivat@alum.mit.edu>:
>> I think we can find a way to achieve those objectives while loosening the
>> requirements to support the sorts of constrained implementation environments
>> we are seeing.
>>
>> Both the websocket transport and outbound rely upon a neighboring sip
>> element to bridge the gap to interoperability. So rather than implementing
>> UDP and TCP itself, the websocket client depends upon the websocket server
>> to implement them on its behalf.
>>
>> I think it will take a bit work to craft new language that provides the
>> needed additional flexibility while not giving license to create
>> implementations that lack basic interoperability.
>
> Hi Paul, the scenario is similar to the one in which a SIP UA behind
> NAT registers using UDP through its Outbound Edge Proxy, so it can
> only receive incoming requests (from outside its local network) via
> its Outbound proxy. Neither it can receive inbound TCP connections.
>
> So, perhaps we could extract some related text from RFC 5626? But note
> that mandating RFC 5626 was removes in Paris due the got consensus.
>
>
> However, this discussion assumes that the SIP UA is a SIP WebSocket
> Client just capable of using WebSocket transport (so probably a web
> browser), and that the SIP WebSocket Server is a SIP proxy which
> cannot open outbound WS connections (and speaks other transports).
> This assumption does not exist in the draft. If a UA or a proxy
> implements both a SIP WebSocket Client and a SIP WebSocket Server then
> all this discussion becomes "invalid" since topology constrains are
> relaxed.
>
> So:
>
> - This draft does not assume network topology.
> - This draft does not assume clients and proxies/servers nature.
> - RFC 3261 mandates UDP and TCP.
>
>
> JavaScripts SIP stacks running in browsers will not implement SIP UDP/TCP.
> Are those SIP stacks "violating" RFC 3261? well, things change since
> 2002, and probably a SIP transport like WebSocket was not expected
> when RFC 3261 was written. But anyhow, from the point of view of the
> draft itself, IMHO the spec is not violating RFC 3261 since it does
> not assume the "web browser" scenario.
>
>
> Best regards.
>
>


From ibc@aliax.net  Wed Apr 18 10:07:27 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 19:07:05 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/18 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> However, as you mention below, a typical use case for this draft is
> javascript sip stacks running in a browser that would not be able to
> implement sip tcp and udp transports. AFAIK those would indeed be
> non-compliant to 3261. If we want to encourage such implementations, then=
 we
> should do something so that they can be compliant. ISTM that this will
> require a revision to 3261. That could be done in a separate draft, or it
> could be done in this draft that is already revising 3261.
>
> One disadvantage to doing it in this draft is that then compliance to the
> revised language would be advertised by advertising support for *this*
> document. If the relaxation is to apply to other implementations of sip t=
hat
> don't want to implement the websocket transport, then it should be done
> separately.

I agree. A new draft should address this issue by covering the following to=
pics:

- Scenarios in which just *secure* SIP transport is required (so UDP
has no place). This could occur in local networks, providers networks
or whatever.

- Scenarios in which a single transport can be used (i.e. in Outbound
topology in which UA's are behind different NAT's and connect to the
same Edge Proxy using TCP or UDP, so they cannot receive requests via
a different transport).

- Special cases in which it's clear that ~100% of implementations are
not able to support SIP UDP/TCP transports (as in this draft).


Basically it would say "in certain cases a SIP entity MAY NOT
implement UDP and/or TCP transports".


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 18 Apr 2012 19:51:57 +0200
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To: "Nataraju A.B" <nataraju.sip@gmail.com>
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Cc: vpascual@acmepacket.com, jmillan@aliax.net, sipcore@ietf.org
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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2012/4/18 Nataraju A.B <nataraju.sip@gmail.com>:
> Hi Inaki,
>
> Please find some more comments inline...
>
>
> Please let me know, the reasoning behind single transport parameter and t=
wo
> sent-protocol values.
> sent-protocol : WS and WSS
> transport =3D ws (only one new value)
>
> This might be confusing in future...

Not really. In the same way that using SIP over TLS-TCP means
;transport=3Dtcp with sips schema, or sending SIP over TLS-SCTP means
;transport=3Dsctp with sips schema. So sending SIP over TLS-WebSocket
must involve ;transport=3Dws as sips schema.

Note that ;transport=3Dtls has never been defined (nor in RFC 2543) and
it seems to appear in RFC 3261 due to dubious reasons, but it is not
used in RFC 3261.

Please check http://tools.ietf.org/html/rfc5630#section-3.1.4.






> if we want to use SIP URI with any other secure communication mechanism l=
ike
> WSS, might require
> update to RFC5630. like
> <rfc5630> sec=C2=A03.1.3.
>
>    If one wants to use "best-effort TLS/WSS" for SIP, one just needs to u=
se
>    a SIP URI, and send the request over TLS ** or any other secure transp=
ort
> mechanism like WSS**
>
> </rfc5630>

TLS is a extra-layer on top of TCP and also on top of WebSocket. When
applied on top of TCP transport it is commonly named "TLS", but it
should be better to call it "TLS-TCP" (in the same way that TLS over
SCTP is named "TLS-SCTP" for Via transport value in RFC 4168.





>> > =C2=A0 =C2=A0Extra closing double quote in contact header.
>>
>> Typo, thanks :)
>>
> [ABN] similar typo mistake in message F3 as well.

Good catch.




>> The example uses a secure WS connection (on top of TLS) but does not
>> use SIPS schema.
>>
>
> [ABN] Probably for the sake of clarity, we can give call flow for Secure =
Web
> Socket (WSS)
> and non-secure Web Socket (WS). this content is already covered in differ=
ent
> sections (this doc and rfc6455 as well). But putting it together with a c=
all
> flow would make the concept more clearer.

The problem is that, trying to include a full secure call would lead
to multiple mail threads discussing SIPS stuf that is really difficult
to understand and implement. There is a RFC trying to improve/explain
it (RFC 5630). I don't think the mission of this draft is to clarify
SIPS usage.



Thanks a lot.




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
Thread-Index: AQHNHSsWQiKr/IZJokCfX43Mp6RbG5agJJrA//+unYCAAGacsIAAFBwAgABihBD//7B5AIAABfgAgAD3fzA=
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Inaki,

I agree with Christer for mandating path in case REGISTER is the first mess=
age
and it is not required to be mandated in case INVITE is the first message.

In the mentioned below callflow, P1 should be some prior knowledge about Bo=
b
that Bob supports websocket transport which is not always feasible through=
=20
standard mechanism. Also, the callflow is not generic enough to handle=20
SIP entity which supports UDP & TCP transport only.

Thanks
Partha

>-----Original Message-----
>From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>Sent: Wednesday, April 18, 2012 9:47 PM
>To: I=F1aki Baz Castillo; Ravindran, Parthasarathi
>Cc: sipcore@ietf.org
>Subject: RE: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>Hi,
>
>In general I agree that a transport mechanism specification shouldn't
>mandate entities to Record-Route.
>
>But, as others have also indicated, in most cases it will be needed. The
>signalling path between the UA and the "outbound proxy" will be
>established already during registration (which means the proxy will also
>have to insert Path), and will then be used for UA originating and
>terminating sessions.
>
>Regards,
>
>Christer
>
>
>
>
>
>________________________________________
>From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of
>I=F1aki Baz Castillo [ibc@aliax.net]
>Sent: Wednesday, April 18, 2012 6:55 PM
>To: Ravindran, Parthasarathi
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>2012/4/18 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
>
>> Having said that, please indicate that Websocket (to any transport)
>Proxy MUST include record-route header in all SIP messages and it is a
>mandatory header for websocket proxy as per this specification. It may
>be preferable to provide the reason for this mandating. IMO, This
>information helps websocket proxy implementers.
>
>
>Hi Ravindran. If a proxy behaves as an Outbound Edge Proxy (RFC 5626)
>then it adds Record-Route always (since NATted clients can only receive
>incoming requests from their Outbound proxy via TCP, UDP, or WebSocket).
>
>Implementing RFC 5626 "Outbound" was a MUST in previous draft version,
>but such a requirement was removed in IETF #83 after consensus by
>members of this WG, so the draft does NOT mandate RFC 5626 (which IMHO
>is a good decision since it "just defines a new SIP transport" rather
>than a complete SIP topology).
>
>Anyhow, Record-Route (Loose-Routing mechanism as defined in RFC 3261) is
>*NOT* required in *ALL* the cases in which a SIP WebSocket UA is
>involved, for example:
>
>
>- alice is a SIP UA implementing a SIP WS Client and a SIP WS Server (so
>it can open and accept WS connections).
>
>- The very same for bob, another SIP WS UA. But bob also speaks UDP and
>registers to P1 via UDP.
>
>- P1 is a proxy/registrar implementing a SIP WebSocket Server (so just
>listens for inbound WS connections) and also UDP.
>
>- P1 does NOT perform Loose-Routing (it does NOT add Record-Route
>headers).
>
>- alice, bob and P1 are located in the same network (or have public
>IP...).
>
>
>So:
>
>
>
>      alice                     P1                    bob
>
>F1    INVITE -----(WS)--------->
>F2                                INVITE ----(UDP)------->
>
>F3                                <---(UDP)----------- 200
>F4     <-----(WS)----------- 200
>
>F5    ACK ---------------------(WS)---------------------->
>
>F6    <------------------------(WS)------------------- BYE
>F7    200 ---------------------(WS)---------------------->
>
>
>F1: INVITE from alice has a Contact URI with ;transport=3Dws.
>
>F2. P1 does NOT add Record-Route header.
>
>F3. 200 from bob contains a Contact URI with ;transport=3Dws (which is
>valid regardless the INVITE arrived via UDP transport).
>
>F5. alice opens a new WS connection to the Contact URI of bob (since ACK
>must be send end to end when there is no Record-Route in 200) and sends
>the ACK.
>
>F7. bob opens a new WS connection to the Contact URI of alice and sends
>the BYE.
>
>
>This is perfectly valid according to RFC 3261 and there is NO need for
>Loose-Routing mechanism, so the draft MUST NOT mandate Loose-Routing
>mechanism.
>
>Again, you assume that the SIP WebSocket UA is a web browser that can
>only open outbound WS connections. The draft is not based on such an
>assumption (regardless we all know that this will be the scenario in 99%
>of cases). We leave the SIP topology open (this was recently discussed
>in this maillist before IETF #83).
>
>
>
>> Also indicates that, stateless proxy is not applicable for this
>transport.
>
>Please read my previous replies. Stateless proxy CAN be used (please
>read my first reply in this thread in which I explain it).
>
>
>
>
>> I like to hear in case any other way to make SIP over websocket works
>without record-route header.
>
>See example above.
>
>
>
>Regards.
>
>
>
>
>--
>I=F1aki Baz Castillo
><ibc@aliax.net>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Nataraju A.B <nataraju.sip@gmail.com>
Thread-Topic: Incoming call notification for websocket transport
Thread-Index: AQHNHdbJfJCzdI+KQE+2AIFAn6luHw==
Date: Thu, 19 Apr 2012 02:47:05 +0000
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: [sipcore] Incoming call notification for websocket transport
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Hi Nataraju,

[ABN] Partha,
          Can you name some scenario which need this incoming SIP requests =
@ the WebSocket client ????

There are two scenario which I could think of


1)      Skype sort of websocket application in endpoint

2)      B2BUA which listen on websocket to receive the calls without regist=
ration

Thanks
Partha

From: Nataraju A.B [mailto:nataraju.sip@gmail.com]
Sent: Wednesday, April 18, 2012 3:50 PM
To: Ravindran, Parthasarathi
Cc: Sa=FAl Ibarra Corretg=E9; SIPCORE (Session Initiation Protocol Core) WG=
; SIPCORE Chairs; I=F1aki Baz Castillo
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]


On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasarathi <pravindran@sonus=
net.com<mailto:pravindran@sonusnet.com>> wrote:
Even if it is the limitations of websocket, SIP UA MUST listen in
UDP & TCP as per RFC 3261 and not adhere to this requirement which MUST
be considered as a violation of RFC 3261. In case you think that this draft
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not
clear enough, Please mention it explicitly in the draft that

"As per this specification, SIP UA MUST support UDP, TCP transport apart
from Websocket as a transport"

This information helps for the implementers who wish to use this draft for
RTCWeb client to SIP Interop in the standard way. I'm writing this mail as
most of the folks support this draft for interworking RTCWeb client to SIP
in the standard manner as per my discussion in IETF-83.

I'll write the separate mail thread for discussing incoming SIP over websoc=
ket
as it is one another major open item in this draft.
[ABN] Partha,
          Can you name some scenario which need this incoming SIP requests =
@ the WebSocket client ????

Thanks
Partha

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com<mailto:saul@ag=
-projects.com>]
>Sent: Wednesday, April 18, 2012 3:23 PM
>To: Ravindran, Parthasarathi
>Cc: I=F1aki Baz Castillo; SIPCORE (Session Initiation Protocol Core) WG;
>SIPCORE Chairs; Nataraju A.B
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>
>On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:
>
>> As per your below statement, JavaScript with SIP and RTCWeb Browser
>MUST NOT use this draft as it violates RFC 3261.
>>
>
>It doesn't, see my previous email explaining why.
>
>> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket
>will not listen for the incoming call without registration but in case
>of your draft is written with SIP softphone (smartphone) in mind , it is
>possible to listen in Port 80 or any other Websocket port which is not
>addressed in this draft till now. We need to discuss and sort it out.
>>
>
>The fact that a JS SIP client (running in a web browser) does not listen
>for incoming connections is a limitation imposed by the WebSocket API,
>not by this draft. This specification defines 2 entities: a SIP
>WebSocket Client and a SIP WebSocket Server, so nobody stops you from
>creating an entity that implements both and can accept incoming
>WebSocket connections. Of course, such an application would not run in a
>browser.
>
>
>Regards,
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>



--
Thanks,
Nataraju A.B.


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Nataraju,<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">[ABN] Partha,&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Can you name some=
 scenario which need this incoming SIP requests @ the WebSocket client ????=
&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">There are two scenario wh=
ich I could think of<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l1 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">1)<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Skype sort of web=
socket application in endpoint
<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l1 level=
1 lfo2"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">2)<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">B2BUA which liste=
n on websocket to receive the calls without registration<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Partha<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju=
 A.B [mailto:nataraju.sip@gmail.com]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 3:50 PM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> Sa=FAl Ibarra Corretg=E9; SIPCORE (Session Initiation Protocol C=
ore) WG; SIPCORE Chairs; I=F1aki Baz Castillo<br>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasa=
rathi &lt;<a href=3D"mailto:pravindran@sonusnet.com">pravindran@sonusnet.co=
m</a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Even if it is the limitations of websocket, SIP UA M=
UST listen in<br>
UDP &amp; TCP as per RFC 3261 and not adhere to this requirement which MUST=
<br>
be considered as a violation of RFC 3261. In case you think that this draft=
<br>
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not<br>
clear enough, Please mention it explicitly in the draft that<br>
<br>
&quot;As per this specification, SIP UA MUST support UDP, TCP transport apa=
rt<br>
from Websocket as a transport&quot;<br>
<br>
This information helps for the implementers who wish to use this draft for<=
br>
RTCWeb client to SIP Interop in the standard way. I'm writing this mail as<=
br>
most of the folks support this draft for interworking RTCWeb client to SIP<=
br>
in the standard manner as per my discussion in IETF-83.<br>
<br>
I'll write the separate mail thread for discussing incoming SIP over websoc=
ket<br>
as it is one another major open item in this draft.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal">[ABN] Partha,&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Can you name some=
 scenario which need this incoming SIP requests @ the WebSocket client ????=
&nbsp;<o:p></o:p></p>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal"><br>
Thanks<br>
Partha<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><br>
&gt;-----Original Message-----<br>
&gt;From: Sa=FAl Ibarra Corretg=E9 [mailto:<a href=3D"mailto:saul@ag-projec=
ts.com">saul@ag-projects.com</a>]<br>
&gt;Sent: Wednesday, April 18, 2012 3:23 PM<br>
&gt;To: Ravindran, Parthasarathi<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;Cc: I=F1aki Baz Castillo; SIPCORE (Session Initi=
ation Protocol Core) WG;<br>
&gt;SIPCORE Chairs; Nataraju A.B<br>
&gt;Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=
<br>
&gt;in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02]<br>
&gt;<br>
&gt;<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&gt;On Apr 18, 2012, =
at 11:35 AM, Ravindran, Parthasarathi wrote:<br>
&gt;<br>
&gt;&gt; As per your below statement, JavaScript with SIP and RTCWeb Browse=
r<br>
&gt;MUST NOT use this draft as it violates RFC 3261.<br>
&gt;&gt;<br>
&gt;<br>
&gt;It doesn't, see my previous email explaining why.<br>
&gt;<br>
&gt;&gt; Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websoc=
ket<br>
&gt;will not listen for the incoming call without registration but in case<=
br>
&gt;of your draft is written with SIP softphone (smartphone) in mind , it i=
s<br>
&gt;possible to listen in Port 80 or any other Websocket port which is not<=
br>
&gt;addressed in this draft till now. We need to discuss and sort it out.<b=
r>
&gt;&gt;<br>
&gt;<br>
&gt;The fact that a JS SIP client (running in a web browser) does not liste=
n<br>
&gt;for incoming connections is a limitation imposed by the WebSocket API,<=
br>
&gt;not by this draft. This specification defines 2 entities: a SIP<br>
&gt;WebSocket Client and a SIP WebSocket Server, so nobody stops you from<b=
r>
&gt;creating an entity that implements both and can accept incoming<br>
&gt;WebSocket connections. Of course, such an application would not run in =
a<br>
&gt;browser.<br>
&gt;<br>
&gt;<br>
&gt;Regards,<br>
&gt;<br>
&gt;--<br>
&gt;Sa=FAl Ibarra Corretg=E9<br>
&gt;AG Projects<br>
&gt;<br>
&gt;<o:p></o:p></p>
</div>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

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Subject: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07.txt, sections 6-9, 10.2, 10.3
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NOTE:  In my comments on sections 1-5, I was unaware of the (somewhat
new) use of "0" to indicate a gap, so I incorrectly referred to
"positive integer" instead of "nonnegative integer".

NOTE:  Should we add somewhere prominent toward the beginning:

   The hi-entries in the History-Info header field are always given in
   the preorder of their logical tree structure, which is the
   lexicographic ordering of their hi-indexes.

This makes it clear that the ordering is always the same and avoids
specifying the ordering in multiple places (as if it was somehow a
special feature of certain steps of processing).


5.  History-Info Header Field Protocol Structure

   index-val =3D  1*DIGIT *("." 1*DIGIT)

Have we ever settled whether we allow leading zeros?  Perhaps we
should forbid leading zeros (except for "0"):

   index-val =3D  number *("." number)

   number =3D  [ %31-39 *DIGIT ] DIGIT

And there probably should be a reference to

    5234 Augmented BNF for Syntax Specifications: ABNF. D. Crocker, Ed.,
	 P. Overell. January 2008. (Format: TXT=3D26359 bytes) (Obsoletes
	 RFC4234) (Also STD0068) (Status: STANDARD)

6.1.  User Agent Client (UAC) Behavior

   In the case of an initial request, except
   where the UAC is part of a B2BUA, there is no cache of hi- entries
   with which to populate the History-info header field and the hi-index
   is set to 1 per Section 10.3.

I don't know if it needs to be stated or not, but there are a couple
of cases where a UAC can generate requests with hi-entries whose
hi-indexes start with 2 or a higher number:

- the UAC acts on a 3xx response it has received, thus generating a
  request that is a sibling of the initial request

- the UAC for some reason is generating a set of requests that are
  intended to be "forks" of an implicit original request=20
  (This happens in draft-ietf-bliss-call-completion.)
  (An alternative would be for the UAC to use internal retargeting and
  label the forks 1.1, 1.2, etc.  But that is more verbose.)

I think that the first point is implicit in the description of 3xx
response processing, but that the second point is probably not
implied.

This could be handled by adding the sentence:

   If the UAC generates further forks of the initial request (either
   due to acting on a 3xx response or internally-directed forking to
   multiple destinations), the successive requests will add hi-entries
   with hi-indexes of 2, 3, etc.

6.2.  User Agent Server (UAS) Behavior

   When receiving a request, a UAS MUST follow the procedures defined in
   Section 9.2.

Do you mean "Section 9.1", which is the one titled "Receiving a
Request"?

   UAS MUST follows the procedures as defined in Section 9.4.  When

should be

   UAS MUST >>>follow<<< the >>>procedures in<<< Section 9.4.  When

7.  Proxy/Intermediary Handling of History-Info Header Fields

   Figure 1 provides an example of internal retargeting.

I think this statement is incorrect -- the only figure 1 is in section
5.1, and does not show internal retargeting.  I suspect this statement
is left over from an earlier version.

9.2.  Sending a Request with History-Info

   In addition, the SIP entity MUST add a new
   hi-entry to the outgoing request, but the SIP entity MUST NOT add the
   hi-entry to the outgoing request(but not the cache) populating the
   new hi-entry to the cache at this time.

This sentence clearly has been damaged during editing.  I think it
should be

   In addition, the SIP entity MUST add a new hi-entry to the outgoing
   request, but the SIP entity MUST NOT add the hi-entry to the cache
   at this time.

At this point, we need to specify the ordering of the hi-entries in
the outgoing History-Info, since the ordering of the hi-entries in the
cache isn't specified:

   The hi-entries in the outgoing request's History-Info header field
   is the preorder of the tree of hi-entries, that is, by the
   lexicographic ordering of the hi-indexes.

This needs to be updated to add "np":

   rc/mp>>>/np<<<:  The SIP entity MUST include an "rc", "mp" or "np" heade=
r
      field parameter in the hi-entry, if applicable, per the procedures
      in Section 10.4.

9.4.  Sending History-Info in Responses

   When sending a response other than a 100, a SIP entity MUST include
   all the cached hi-entries in the response, subject to the privacy
   consideration in Section 10.1.2 with the following exception: If the
   received request contained no hi-entries and there is no "histinfo"
   option tag in the Supported header field, the SIP entity MUST NOT
   include hi-entries in the response.

I think this can be clarified by

   When sending a response other than a 100, a SIP entity MUST include
   all the cached hi-entries in the response, subject to the privacy
   consideration in Section 10.1.2>>>, and<<< with the following
   exception: If the
   received request contained no hi-entries and there is no "histinfo"
   option tag in the Supported header field, the SIP entity MUST NOT
   include >>>History-Info<<< in the response.

10.2.  Reason in the History-info Header Field

   If a request has timed out (instead of being explicitly rejected),
   the SIP entity MUST include a Reason header field, containing a SIP
   error response code of 408 "Request Timeout".

More exact would be to say

   If a request has timed out (instead of being explicitly rejected),
   the SIP entity MUST update the cache as if the request received a=20
   SIP error response code of 408 "Request Timeout".

That allows tel: URIs to be handled correctly, i.e., not receiving a
Reason header.

There needs to be some discussion of the fact that a request can
receive multiple "non-100 non-2xx" responses, and that their Reason
headers can have duplicate or non-duplicate reason-values.  That could
be done by adding at this point in 10.2 this paragraph:

   A request can receive multiple non-100 non-2xx responses which
   carry or imply (for responses without Reason headers, and for
   timeouts) multiple, possibly duplicated, reason-values to be
   applied to a non-tel hi-targeted-to-uri.  In these situations, the
   SIP entity must accumulate without duplications all the distinct
   reason-values in the received or implied Reason header fields.

... or whatever rule we want to adopt.

10.3.  Indexing in the History-Info Header Field

   Per the syntax in Section 5, the hi-index consists of a series of
   digits separated by dots (e.g., 1.1.2).

As far as I can tell, "digit" should be replaced with "nonnegative
integer" (or perhaps just "integer", since the BNF enforces
nonnegativeness) everywhere in the document.

   The highest digit at each
   hop reflects the number of entities to which the request has been
   retargeted at the specific hop (i.e., the number of branches).

This should be expanded by adding "at the time that the request
represented by this hi-entry was generated", since the "number of
entities to which..." changes over time, but of course, the integer in
the sent request cannot.

   Thus,
   the indexing results in a logical tree representation for the history
   of the request.

I think it would be useful to add "and the hi-entries are given in the
preorder of the tree".

   In the case that a SIP entity (intermediary or UAS) adds an hi-entry
   on behalf of the previous hop, the hi-index MUST be set to 1.

I'm not certain of the meaning of this, but I think it's incorrect as
written.  At the least, if the intermediary adds an additional
hi-entry to an existing History-Info, then it has to follow the "0.1"
rule.

If a first-hop proxy adds History-Info on behalf of a UAC, then it can
use index 1 for the initial hi-entry.

If a non-first-hop proxy adds History-Info on behalf of the upstream
entities, the situation is messier.  In principle, it should probably
apply the "0.1" rule, so the first hi-entry does not have index "1".

However, in that case, the intermediary probably won't be sending
History-Info upstream in the response, so there is no way that the
upstream entities can get confused by receiving multiple replies with
different History-Info trees that use the same root index (1).

(Actually, a UAC that acts on 3xx or initially forks can cause the
first-hop proxy to be in the same situation as a non-first-hop proxy.
But the same analysis shows that always using an initial index of 1
won't confuse the UAC.)

So I think the statement should be

   In the case that a SIP entity (intermediary or UAS) adds >>>a
   first<<< hi-entry on behalf of the previous hop, the hi-index MUST
   be set to 1.

   For
   each forward hop (i.e., each new level of indexing), the hi-index
   MUST start at 1.  An increment of 1 MUST be used for advancing to a
   new branch.

Actually, that should be "the last integers of the hi-indexes of the
new requests MUST be generated starting at 1 and incrementing by 1 for
each additional request".  The hi-indexes *themselves* don't start at
1.

       Note, that in the case of parallel forking,
       only the hi-entry corresponding to the fork is included in the
       request.

It might be useful to add "because no response can yet have been
received for any of the parallel forked requests" -- that may not be
obvious to the first-time reader.

   6.  Missing entity: If the request clearly has a gap in the hi-entry
       (e.g. by evaluating via header to the existing request history to
       see if it traversed a domain which doesn't exist in History-Info
       header field.), the entity adding an hi-entry MUST add an index a
       digit of "0" to the current index prior to adding appropriate
       index for the action to be taken.  If the index of the last hi-
       entry in the request received was 1.1.2 and there was a missing
       hi-entry and the request was being forwarded to the next hop, the
       resulting index will be 1.1.2.0.1.

This rule doesn't really belong here.  Properly, it describes an
additional condition that triggers the adding of an hi-entry described
in section 9.1.

It's also not clear to me that this trigger condition can be specified
sufficiently accurately to allow this to become a MUST.  If we are
relaxed about this, the revision would be to delete this item 6 and
update section 9.1:

   The entity MUST attempt, in the following manner, to determine if the
   previous SIP entity that sent the request did not add a hi-entry to
   the History-Info header field:

   1. It MUST detect if the Request-URI of the incoming request does
   not match the hi-targeted-to-uri in the last hi-entry.

   2. It MAY detect the condition via other means, for example,=20
   if the topmost Via header shows traversing a domain which is
   different from the last hi-entry, or if the request was received
   from an address to which the last hi-targeted-to-uri could not have
   addressed the request.

(I expect that border devices may often be able to apply the latter conditi=
on.)

   When this condition is detected, the SIP entity MUST add a hi-entry
   to end of the cache, on behalf of the previous SIP entity, as
   follows:

Dale

From shida@ntt-at.com  Wed Apr 18 23:28:18 2012
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Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-07, sections 1 to 5
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Dale;

 Thanks for the suggestion and the catch. 

 Regards
  Shida

On Apr 19, 2012, at 12:48 AM, Worley, Dale R (Dale) wrote:

>> From: Shida Schubert [shida@ntt-at.com]
>> 
>>>        "np": The hi-targeted-to-URI represents that there was no
>>>        change in Request-URI.  This would apply for example when a
>>>        proxy merely forwards a request to a next hop proxy and loose
>>>        routing is used.
>>> 
>>> Unlike the previous two cases, this case does not describe the
>>> function of the value of the np parameter.
>> 
>> Any suggestion for rewording?
> 
> Assuming that the values of the three parameters are intended to carry
> the same information (that is, the index of the hi-entry from which
> this request-URI was derived), the value of "np" would be the index of
> the parent hi-entry of this hi-entry.  As in the example in 5.1:
> 
>   | History-Info: <sip:bob@biloxi.example.com;p=x>;index=1
>   | History-Info: <sip:bob@biloxi.example.com;p=x>;np=1;index=1.1
> 
> Largely copying the wording of the entry for "rc", we can add:
> 
>         "np": The hi-targeted-to-URI represents that there was no
>         change in Request-URI.  This would apply for example when a
>         proxy merely forwards a request to a next hop proxy and loose
>         routing is used.
>>>> The "np"
>         header field parameter contains the value of the hi-index in
>         the hi-entry with an hi-targeted-to-uri that reflects the
>         Request-URI that was copied unchanged into the request
>         represented by this hi-entry.  That value will usually be the
>         hi-index of the parent hi-entry of this hi-entry.<<<
> 
> While I was lookng at this, the entry for "mp" has some awkward
> wording that could be improved by copying the wording from "rc".
> Instead of
> 
>         The "mp" header field parameter contains the
>>>> value which represents the value<<< of the hi-index in the
>         hi-entry ...
> 
> perhaps
> 
>         The "mp"
>         header field parameter contains the >>>value<<< of the hi-index in
>         the hi-entry ...
> 
> Dale


From shida@ntt-at.com  Thu Apr 19 03:44:29 2012
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Subject: Re: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07.txt, sections 6-9, 10.2, 10.3
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Hi Dale;

 Thanks again for your extensive review.

 My comments inline.

On Apr 19, 2012, at 12:43 PM, Worley, Dale R (Dale) wrote:

> NOTE:  In my comments on sections 1-5, I was unaware of the (somewhat
> new) use of "0" to indicate a gap, so I incorrectly referred to
> "positive integer" instead of "nonnegative integer".
>=20
> NOTE:  Should we add somewhere prominent toward the beginning:
>=20
>   The hi-entries in the History-Info header field are always given in
>   the preorder of their logical tree structure, which is the
>   lexicographic ordering of their hi-indexes.
>=20
> This makes it clear that the ordering is always the same and avoids
> specifying the ordering in multiple places (as if it was somehow a
> special feature of certain steps of processing).

 I thought the description of hi-index was sufficient but if you don't=20=

think so, we can add this. Where do you recommend we add this?=20
In the Introduction??=20

>=20
>=20
> 5.  History-Info Header Field Protocol Structure
>=20
>   index-val =3D  1*DIGIT *("." 1*DIGIT)
>=20
> Have we ever settled whether we allow leading zeros?  Perhaps we
> should forbid leading zeros (except for "0"):
>=20
>   index-val =3D  number *("." number)
>=20
>   number =3D  [ %31-39 *DIGIT ] DIGIT
>=20
> And there probably should be a reference to
>=20
>    5234 Augmented BNF for Syntax Specifications: ABNF. D. Crocker, =
Ed.,
> 	 P. Overell. January 2008. (Format: TXT=3D26359 bytes) =
(Obsoletes
> 	 RFC4234) (Also STD0068) (Status: STANDARD)

 DONE.

>=20
> 6.1.  User Agent Client (UAC) Behavior
>=20
>   In the case of an initial request, except
>   where the UAC is part of a B2BUA, there is no cache of hi- entries
>   with which to populate the History-info header field and the =
hi-index
>   is set to 1 per Section 10.3.
>=20
> I don't know if it needs to be stated or not, but there are a couple
> of cases where a UAC can generate requests with hi-entries whose
> hi-indexes start with 2 or a higher number:
>=20
> - the UAC acts on a 3xx response it has received, thus generating a
>  request that is a sibling of the initial request
>=20
> - the UAC for some reason is generating a set of requests that are
>  intended to be "forks" of an implicit original request=20
>  (This happens in draft-ietf-bliss-call-completion.)
>  (An alternative would be for the UAC to use internal retargeting and
>  label the forks 1.1, 1.2, etc.  But that is more verbose.)
>=20
> I think that the first point is implicit in the description of 3xx
> response processing, but that the second point is probably not
> implied.
>=20
> This could be handled by adding the sentence:
>=20
>   If the UAC generates further forks of the initial request (either
>   due to acting on a 3xx response or internally-directed forking to
>   multiple destinations), the successive requests will add hi-entries
>   with hi-indexes of 2, 3, etc.

DONE.

>=20
> 6.2.  User Agent Server (UAS) Behavior
>=20
>   When receiving a request, a UAS MUST follow the procedures defined =
in
>   Section 9.2.
>=20
> Do you mean "Section 9.1", which is the one titled "Receiving a
> Request"?

That is the intent.

>=20
>   UAS MUST follows the procedures as defined in Section 9.4.  When
>=20
> should be
>=20
>   UAS MUST >>>follow<<< the >>>procedures in<<< Section 9.4.  When

DONE.

>=20
> 7.  Proxy/Intermediary Handling of History-Info Header Fields
>=20
>   Figure 1 provides an example of internal retargeting.
>=20
> I think this statement is incorrect -- the only figure 1 is in section
> 5.1, and does not show internal retargeting.  I suspect this statement
> is left over from an earlier version.

REMOVED.

>=20
> 9.2.  Sending a Request with History-Info
>=20
>   In addition, the SIP entity MUST add a new
>   hi-entry to the outgoing request, but the SIP entity MUST NOT add =
the
>   hi-entry to the outgoing request(but not the cache) populating the
>   new hi-entry to the cache at this time.
>=20
> This sentence clearly has been damaged during editing.  I think it
> should be
>=20
>   In addition, the SIP entity MUST add a new hi-entry to the outgoing
>   request, but the SIP entity MUST NOT add the hi-entry to the cache
>   at this time.

 FIXED.

>=20
> At this point, we need to specify the ordering of the hi-entries in
> the outgoing History-Info, since the ordering of the hi-entries in the
> cache isn't specified:
>=20
>   The hi-entries in the outgoing request's History-Info header field
>   is the preorder of the tree of hi-entries, that is, by the
>   lexicographic ordering of the hi-indexes.

 DONE.

>=20
> This needs to be updated to add "np":
>=20
>   rc/mp>>>/np<<<:  The SIP entity MUST include an "rc", "mp" or "np" =
header
>      field parameter in the hi-entry, if applicable, per the =
procedures
>      in Section 10.4.

 DONE.

>=20
> 9.4.  Sending History-Info in Responses
>=20
>   When sending a response other than a 100, a SIP entity MUST include
>   all the cached hi-entries in the response, subject to the privacy
>   consideration in Section 10.1.2 with the following exception: If the
>   received request contained no hi-entries and there is no "histinfo"
>   option tag in the Supported header field, the SIP entity MUST NOT
>   include hi-entries in the response.
>=20
> I think this can be clarified by
>=20
>   When sending a response other than a 100, a SIP entity MUST include
>   all the cached hi-entries in the response, subject to the privacy
>   consideration in Section 10.1.2>>>, and<<< with the following
>   exception: If the
>   received request contained no hi-entries and there is no "histinfo"
>   option tag in the Supported header field, the SIP entity MUST NOT
>   include >>>History-Info<<< in the response.

 FIXED.

>=20
> 10.2.  Reason in the History-info Header Field
>=20
>   If a request has timed out (instead of being explicitly rejected),
>   the SIP entity MUST include a Reason header field, containing a SIP
>   error response code of 408 "Request Timeout".
>=20
> More exact would be to say
>=20
>   If a request has timed out (instead of being explicitly rejected),
>   the SIP entity MUST update the cache as if the request received a=20
>   SIP error response code of 408 "Request Timeout".

 FIXED.

>=20
> That allows tel: URIs to be handled correctly, i.e., not receiving a
> Reason header.
>=20
> There needs to be some discussion of the fact that a request can
> receive multiple "non-100 non-2xx" responses, and that their Reason
> headers can have duplicate or non-duplicate reason-values.  That could
> be done by adding at this point in 10.2 this paragraph:
>=20
>   A request can receive multiple non-100 non-2xx responses which
>   carry or imply (for responses without Reason headers, and for
>   timeouts) multiple, possibly duplicated, reason-values to be
>   applied to a non-tel hi-targeted-to-uri.  In these situations, the
>   SIP entity must accumulate without duplications all the distinct
>   reason-values in the received or implied Reason header fields.
>=20
> ... or whatever rule we want to adopt.

 Added this with some modified text.=20

>=20
> 10.3.  Indexing in the History-Info Header Field
>=20
>   Per the syntax in Section 5, the hi-index consists of a series of
>   digits separated by dots (e.g., 1.1.2).
>=20
> As far as I can tell, "digit" should be replaced with "nonnegative
> integer" (or perhaps just "integer", since the BNF enforces
> nonnegativeness) everywhere in the document.

 FIXED.

>=20
>   The highest digit at each
>   hop reflects the number of entities to which the request has been
>   retargeted at the specific hop (i.e., the number of branches).
>=20
> This should be expanded by adding "at the time that the request
> represented by this hi-entry was generated", since the "number of
> entities to which..." changes over time, but of course, the integer in
> the sent request cannot.
>=20
>   Thus,
>   the indexing results in a logical tree representation for the =
history
>   of the request.
>=20
> I think it would be useful to add "and the hi-entries are given in the
> preorder of the tree".

 FIXED.

>=20
>   In the case that a SIP entity (intermediary or UAS) adds an hi-entry
>   on behalf of the previous hop, the hi-index MUST be set to 1.
>=20
> I'm not certain of the meaning of this, but I think it's incorrect as
> written.  At the least, if the intermediary adds an additional
> hi-entry to an existing History-Info, then it has to follow the "0.1"
> rule.
>=20
> If a first-hop proxy adds History-Info on behalf of a UAC, then it can
> use index 1 for the initial hi-entry.
>=20
> If a non-first-hop proxy adds History-Info on behalf of the upstream
> entities, the situation is messier.  In principle, it should probably
> apply the "0.1" rule, so the first hi-entry does not have index "1".
>=20
> However, in that case, the intermediary probably won't be sending
> History-Info upstream in the response, so there is no way that the
> upstream entities can get confused by receiving multiple replies with
> different History-Info trees that use the same root index (1).
>=20
> (Actually, a UAC that acts on 3xx or initially forks can cause the
> first-hop proxy to be in the same situation as a non-first-hop proxy.
> But the same analysis shows that always using an initial index of 1
> won't confuse the UAC.)
>=20
> So I think the statement should be
>=20
>   In the case that a SIP entity (intermediary or UAS) adds >>>a
>   first<<< hi-entry on behalf of the previous hop, the hi-index MUST
>   be set to 1.
>=20

FIXED.

>   For
>   each forward hop (i.e., each new level of indexing), the hi-index
>   MUST start at 1.  An increment of 1 MUST be used for advancing to a
>   new branch.
>=20
> Actually, that should be "the last integers of the hi-indexes of the
> new requests MUST be generated starting at 1 and incrementing by 1 for
> each additional request".  The hi-indexes *themselves* don't start at
> 1.

FIXED.

>=20
>       Note, that in the case of parallel forking,
>       only the hi-entry corresponding to the fork is included in the
>       request.
>=20
> It might be useful to add "because no response can yet have been
> received for any of the parallel forked requests" -- that may not be
> obvious to the first-time reader.

FIXED.

>=20
>   6.  Missing entity: If the request clearly has a gap in the hi-entry
>       (e.g. by evaluating via header to the existing request history =
to
>       see if it traversed a domain which doesn't exist in History-Info
>       header field.), the entity adding an hi-entry MUST add an index =
a
>       digit of "0" to the current index prior to adding appropriate
>       index for the action to be taken.  If the index of the last hi-
>       entry in the request received was 1.1.2 and there was a missing
>       hi-entry and the request was being forwarded to the next hop, =
the
>       resulting index will be 1.1.2.0.1.
>=20
> This rule doesn't really belong here.  Properly, it describes an
> additional condition that triggers the adding of an hi-entry described
> in section 9.1.

 Section 9.1 already describes the condition as to when it=20
adds hi-entry.=20

 " If the Request-URI of the incoming request does not match the hi-
   targeted-to-uri in the last hi-entry (i.e., the previous SIP entity
   that sent the request did not include a History-Info header field),
   the SIP entity MUST add a hi-entry to end of the cache, on behalf of
   the previous SIP entity"=20

 Above is strictly talking about condition when "0" is to be used for=20
index.=20

 The fact that last hi-entry and R-URI doesn't match is a rule for=20
adding an hi-entry there is no additional rule for adding an=20
hi-entry on behalf of the previous hop.=20

>=20
> It's also not clear to me that this trigger condition can be specified
> sufficiently accurately to allow this to become a MUST.  If we are
> relaxed about this, the revision would be to delete this item 6 and
> update section 9.1:

 We can change it to SHOULD. =20

 Regards
  Shida

>=20
>   The entity MUST attempt, in the following manner, to determine if =
the
>   previous SIP entity that sent the request did not add a hi-entry to
>   the History-Info header field:
>=20
>   1. It MUST detect if the Request-URI of the incoming request does
>   not match the hi-targeted-to-uri in the last hi-entry.
>=20
>   2. It MAY detect the condition via other means, for example,=20
>   if the topmost Via header shows traversing a domain which is
>   different from the last hi-entry, or if the request was received
>   from an address to which the last hi-targeted-to-uri could not have
>   addressed the request.
>=20
> (I expect that border devices may often be able to apply the latter =
condition.)
>=20
>   When this condition is detected, the SIP entity MUST add a hi-entry
>   to end of the cache, on behalf of the previous SIP entity, as
>   follows:
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From nataraju.sip@gmail.com  Thu Apr 19 07:22:31 2012
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Date: Thu, 19 Apr 2012 19:52:25 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--f46d04016a77aab15004be08e66a
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On Wed, Apr 18, 2012 at 10:28 PM, Paul Kyzivat <pkyzivat@alum.mit.edu>wrote=
:

> I=F1aki,
>
> I agree that there is nothing in this draft that requires someone
> implementing it to violate 3261.
>
> However, as you mention below, a typical use case for this draft is
> javascript sip stacks running in a browser that would not be able to
> implement sip tcp and udp transports. AFAIK those would indeed be
> non-compliant to 3261. If we want to encourage such implementations, then
> we should do something so that they can be compliant. ISTM that this will
> require a revision to 3261. That could be done in a separate draft, or it
> could be done in this draft that is already revising 3261.
>
> One disadvantage to doing it in this draft is that then compliance to the
> revised language would be advertised by advertising support for *this*
> document. If the relaxation is to apply to other implementations of sip
> that don't want to implement the websocket transport, then it should be
> done separately.
>

   [ABN] I agree, it is better to address these issues in a new draft to
specify the deviations from 3261 due to web socket transport support. At
least there should be a statement in this doc/draft mentioning the
deviation from 3261 like,

   5.4.  Deviations from RFC 3261

   These deviations shall be conditional, non support for UDP / TCP
acceptable only when WebSocket transport or equivalent one is supported. In
no other cases this relaxation applicable.


>        Thanks,
>        Paul
>
>
> On 4/18/12 12:13 PM, I=F1aki Baz Castillo wrote:
>
>> 2012/4/18 Paul Kyzivat<pkyzivat@alum.mit.edu>**:
>>
>>> I think we can find a way to achieve those objectives while loosening t=
he
>>> requirements to support the sorts of constrained implementation
>>> environments
>>> we are seeing.
>>>
>>> Both the websocket transport and outbound rely upon a neighboring sip
>>> element to bridge the gap to interoperability. So rather than
>>> implementing
>>> UDP and TCP itself, the websocket client depends upon the websocket
>>> server
>>> to implement them on its behalf.
>>>
>>> I think it will take a bit work to craft new language that provides the
>>> needed additional flexibility while not giving license to create
>>> implementations that lack basic interoperability.
>>>
>>
>> Hi Paul, the scenario is similar to the one in which a SIP UA behind
>> NAT registers using UDP through its Outbound Edge Proxy, so it can
>> only receive incoming requests (from outside its local network) via
>> its Outbound proxy. Neither it can receive inbound TCP connections.
>>
>> So, perhaps we could extract some related text from RFC 5626? But note
>> that mandating RFC 5626 was removes in Paris due the got consensus.
>>
>>
>> However, this discussion assumes that the SIP UA is a SIP WebSocket
>> Client just capable of using WebSocket transport (so probably a web
>> browser), and that the SIP WebSocket Server is a SIP proxy which
>> cannot open outbound WS connections (and speaks other transports).
>> This assumption does not exist in the draft. If a UA or a proxy
>> implements both a SIP WebSocket Client and a SIP WebSocket Server then
>> all this discussion becomes "invalid" since topology constrains are
>> relaxed.
>>
>> So:
>>
>> - This draft does not assume network topology.
>> - This draft does not assume clients and proxies/servers nature.
>> - RFC 3261 mandates UDP and TCP.
>>
>>
>> JavaScripts SIP stacks running in browsers will not implement SIP UDP/TC=
P.
>> Are those SIP stacks "violating" RFC 3261? well, things change since
>> 2002, and probably a SIP transport like WebSocket was not expected
>> when RFC 3261 was written. But anyhow, from the point of view of the
>> draft itself, IMHO the spec is not violating RFC 3261 since it does
>> not assume the "web browser" scenario.
>>
>>
>> Best regards.
>>
>>
>>
> ______________________________**_________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/**listinfo/sipcore<https://www.ietf.org/mail=
man/listinfo/sipcore>
>



--=20
Thanks,
Nataraju A.B.

--f46d04016a77aab15004be08e66a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><br><div class=3D"gmail_quote">On Wed, Apr 18, 2012 at 10:28 PM, Paul K=
yzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzi=
vat@alum.mit.edu</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">
I=F1aki,<br>
<br>
I agree that there is nothing in this draft that requires someone implement=
ing it to violate 3261.<br>
<br>
However, as you mention below, a typical use case for this draft is javascr=
ipt sip stacks running in a browser that would not be able to implement sip=
 tcp and udp transports. AFAIK those would indeed be non-compliant to 3261.=
 If we want to encourage such implementations, then we should do something =
so that they can be compliant. ISTM that this will require a revision to 32=
61. That could be done in a separate draft, or it could be done in this dra=
ft that is already revising 3261.<br>

<br>
One disadvantage to doing it in this draft is that then compliance to the r=
evised language would be advertised by advertising support for *this* docum=
ent. If the relaxation is to apply to other implementations of sip that don=
&#39;t want to implement the websocket transport, then it should be done se=
parately.<br>
</blockquote><div><br></div><div><font face=3D"&#39;courier new&#39;, monos=
pace" color=3D"#000099">=A0 =A0[ABN] I agree, it is better to address these=
 issues in a new draft to specify the deviations from 3261 due to=A0web soc=
ket=A0transport support.=A0At least=A0there should be a statement in this d=
oc/draft mentioning the deviation from 3261 like,</font></div>
<div><pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D=
"&#39;courier new&#39;, monospace" color=3D"#000099">   5.4.  Deviations fr=
om RFC 3261</font></pre></div><div><font face=3D"&#39;courier new&#39;, mon=
ospace" color=3D"#000099">=A0 =A0These deviations shall be conditional, non=
 support for UDP / TCP acceptable only when WebSocket transport or equivale=
nt one is supported. In no other cases this relaxation applicable.</font></=
div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<br>
 =A0 =A0 =A0 =A0Thanks,<br>
 =A0 =A0 =A0 =A0Paul<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 4/18/12 12:13 PM, I=F1aki Baz Castillo wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
2012/4/18 Paul Kyzivat&lt;<a href=3D"mailto:pkyzivat@alum.mit.edu" target=
=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;<u></u>:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think we can find a way to achieve those objectives while loosening the<b=
r>
requirements to support the sorts of constrained implementation environment=
s<br>
we are seeing.<br>
<br>
Both the websocket transport and outbound rely upon a neighboring sip<br>
element to bridge the gap to interoperability. So rather than implementing<=
br>
UDP and TCP itself, the websocket client depends upon the websocket server<=
br>
to implement them on its behalf.<br>
<br>
I think it will take a bit work to craft new language that provides the<br>
needed additional flexibility while not giving license to create<br>
implementations that lack basic interoperability.<br>
</blockquote>
<br>
Hi Paul, the scenario is similar to the one in which a SIP UA behind<br>
NAT registers using UDP through its Outbound Edge Proxy, so it can<br>
only receive incoming requests (from outside its local network) via<br>
its Outbound proxy. Neither it can receive inbound TCP connections.<br>
<br>
So, perhaps we could extract some related text from RFC 5626? But note<br>
that mandating RFC 5626 was removes in Paris due the got consensus.<br>
<br>
<br>
However, this discussion assumes that the SIP UA is a SIP WebSocket<br>
Client just capable of using WebSocket transport (so probably a web<br>
browser), and that the SIP WebSocket Server is a SIP proxy which<br>
cannot open outbound WS connections (and speaks other transports).<br>
This assumption does not exist in the draft. If a UA or a proxy<br>
implements both a SIP WebSocket Client and a SIP WebSocket Server then<br>
all this discussion becomes &quot;invalid&quot; since topology constrains a=
re<br>
relaxed.<br>
<br>
So:<br>
<br>
- This draft does not assume network topology.<br>
- This draft does not assume clients and proxies/servers nature.<br>
- RFC 3261 mandates UDP and TCP.<br>
<br>
<br>
JavaScripts SIP stacks running in browsers will not implement SIP UDP/TCP.<=
br>
Are those SIP stacks &quot;violating&quot; RFC 3261? well, things change si=
nce<br>
2002, and probably a SIP transport like WebSocket was not expected<br>
when RFC 3261 was written. But anyhow, from the point of view of the<br>
draft itself, IMHO the spec is not violating RFC 3261 since it does<br>
not assume the &quot;web browser&quot; scenario.<br>
<br>
<br>
Best regards.<br>
<br>
<br>
</blockquote>
<br></div></div><div class=3D"HOEnZb"><div class=3D"h5">
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>

--f46d04016a77aab15004be08e66a--

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Date: Thu, 19 Apr 2012 20:04:03 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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--90e6ba308f0c4169d604be0910da
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On Wed, Apr 18, 2012 at 10:37 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:

> 2012/4/18 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> > However, as you mention below, a typical use case for this draft is
> > javascript sip stacks running in a browser that would not be able to
> > implement sip tcp and udp transports. AFAIK those would indeed be
> > non-compliant to 3261. If we want to encourage such implementations,
> then we
> > should do something so that they can be compliant. ISTM that this will
> > require a revision to 3261. That could be done in a separate draft, or =
it
> > could be done in this draft that is already revising 3261.
> >
> > One disadvantage to doing it in this draft is that then compliance to t=
he
> > revised language would be advertised by advertising support for *this*
> > document. If the relaxation is to apply to other implementations of sip
> that
> > don't want to implement the websocket transport, then it should be done
> > separately.
>
> I agree. A new draft should address this issue by covering the following
> topics:
>
> - Scenarios in which just *secure* SIP transport is required (so UDP
> has no place). This could occur in local networks, providers networks
> or whatever.
>
> - Scenarios in which a single transport can be used (i.e. in Outbound
> topology in which UA's are behind different NAT's and connect to the
> same Edge Proxy using TCP or UDP, so they cannot receive requests via
> a different transport).
>
> - Special cases in which it's clear that ~100% of implementations are
> not able to support SIP UDP/TCP transports (as in this draft).
>
>
> Basically it would say "in certain cases a SIP entity MAY NOT
> implement UDP and/or TCP transports".
>

    [ABN] In addition, we can also add the change for deprecated transport
parameter, *transport=3Dtls*.

transport-param   =3D  "transport=3D"
                     ( "udp" / "tcp" / "sctp" / other-transport)

     --> "tls" removed

This should have been taken care when rfc5630 was introduced (but some
how missed). Atleast we can take care off through this new draft.



>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



--=20
Thanks,
Nataraju A.B.

--90e6ba308f0c4169d604be0910da
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<br><br><div class=3D"gmail_quote">On Wed, Apr 18, 2012 at 10:37 PM, I=F1ak=
i Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net">ibc@a=
liax.net</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">
2012/4/18 Paul Kyzivat &lt;<a href=3D"mailto:pkyzivat@alum.mit.edu">pkyziva=
t@alum.mit.edu</a>&gt;:<br>
<div class=3D"im">&gt; However, as you mention below, a typical use case fo=
r this draft is<br>
&gt; javascript sip stacks running in a browser that would not be able to<b=
r>
&gt; implement sip tcp and udp transports. AFAIK those would indeed be<br>
&gt; non-compliant to 3261. If we want to encourage such implementations, t=
hen we<br>
&gt; should do something so that they can be compliant. ISTM that this will=
<br>
&gt; require a revision to 3261. That could be done in a separate draft, or=
 it<br>
&gt; could be done in this draft that is already revising 3261.<br>
&gt;<br>
&gt; One disadvantage to doing it in this draft is that then compliance to =
the<br>
&gt; revised language would be advertised by advertising support for *this*=
<br>
&gt; document. If the relaxation is to apply to other implementations of si=
p that<br>
&gt; don&#39;t want to implement the websocket transport, then it should be=
 done<br>
&gt; separately.<br>
<br>
</div>I agree. A new draft should address this issue by covering the follow=
ing topics:<br>
<br>
- Scenarios in which just *secure* SIP transport is required (so UDP<br>
has no place). This could occur in local networks, providers networks<br>
or whatever.<br>
<br>
- Scenarios in which a single transport can be used (i.e. in Outbound<br>
topology in which UA&#39;s are behind different NAT&#39;s and connect to th=
e<br>
same Edge Proxy using TCP or UDP, so they cannot receive requests via<br>
a different transport).<br>
<br>
- Special cases in which it&#39;s clear that ~100% of implementations are<b=
r>
not able to support SIP UDP/TCP transports (as in this draft).<br>
<br>
<br>
Basically it would say &quot;in certain cases a SIP entity MAY NOT<br>
implement UDP and/or TCP transports&quot;.<br></blockquote><div>=A0</div><d=
iv><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">=A0 =
=A0 [ABN] In addition, we can also add the change for deprecated transport =
parameter, <b>transport=3Dtls</b>.</font></div>
<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"><br>=
</font></div><div><pre style=3D"word-wrap:break-word;white-space:pre-wrap">=
<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">transport=
-param   =3D  &quot;transport=3D&quot;
                     ( &quot;udp&quot; / &quot;tcp&quot; / &quot;sctp&quot;=
 / other-transport) =A0</font></pre><pre style=3D"word-wrap:break-word;whit=
e-space:pre-wrap"><font face=3D"&#39;courier new&#39;, monospace" color=3D"=
#000099">     --&gt; &quot;tls&quot; removed</font></pre>
<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" color=3D"#000099">This should have been taken=
 care when rfc5630 was introduced (but some how missed). Atleast we can tak=
e care off through this new draft.</font></pre>
<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" color=3D"#000099"><br></font></pre></div><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">

<div class=3D"im HOEnZb"><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>

--90e6ba308f0c4169d604be0910da--

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From: Brett Tate <brett@broadsoft.com>
To: Nataraju A.B <nataraju.sip@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 19 Apr 2012 08:01:19 -0700
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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>=A0[ABN] In addition, we can also add the change for deprecated=20
> transport parameter, transport=3Dtls.
>
> transport-param   =3D  "transport=3D"
>                     ( "udp" / "tcp" / "sctp" / other-transport) =A0
>     --> "tls" removed
> This should have been taken care when rfc5630 was introduced=20
> (but some how missed). Atleast we can take care off through=20
> this new draft.

RFC 5630 and RFC 3261 indicate why "tls" was not to removed from ABNF.

RFC 5630:

3.1.4.  Usage of the transport=3Dtls URI Parameter and the TLS Via
        Parameter

   [RFC3261], Section 26.2.2 deprecated the "transport=3Dtls" URI
   transport parameter in SIPS or SIP URIs:

      Note that in the SIPS URI scheme, transport is independent of TLS,
      and thus "sips:alice@atlanta.com;transport=3DTCP" and
      "sips:alice@atlanta.com;transport=3Dsctp" are both valid (although
      note that UDP is not a valid transport for SIPS).  The use of
      "transport=3Dtls" has consequently been deprecated, partly because
      it was specific to a single hop of the request.  This is a change
      since RFC 2543.

   The "tls" parameter has not been eliminated from the ABNF in
   [RFC3261], Section 25, since the parameter needs to remain in the
   ABNF for backward compatibility in order for parsers to be able to
   process the parameter correctly.  The transport=3Dtls parameter has
   never been defined in an RFC, but only in some of the Internet drafts
   between [RFC2543] and [RFC3261].

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 04/18/2012 08:56 PM, Ravindran, Parthasarathi wrote:
> Inaki,
>
> I agree with Christer for mandating path in case REGISTER is the first message
> and it is not required to be mandated in case INVITE is the first message.
>
> In the mentioned below callflow, P1 should be some prior knowledge about Bob
> that Bob supports websocket transport which is not always feasible through
> standard mechanism. Also, the callflow is not generic enough to handle
> SIP entity which supports UDP&  TCP transport only.

This has already been covered extensively on the list; this document 
should not *normatively* make any such statements. The requirement to 
use Outbound, Path, or any other non-RFC 3261 mechanism is dependent on 
the architecture of the SIP deployment, and it's up to the implementor 
to determine whether any such mechanisms are necessary and how they 
should be used. The document describing the WebSocket transport has no 
place mandating such usage.

In addition, whether these requests are the 'first message' over the 
WebSocket connection or not is not relevant. If a UA wants to open a 
SIP-over-WebSocket connection and then send four OPTIONS requests before 
sending a REGISTER, there's no reason for that to be a problem.

If the informative sections of the document need to be expanded to 
include such guidance and recommendations that could be useful, but 
that's a separate discussion.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From nataraju.sip@gmail.com  Thu Apr 19 08:17:34 2012
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Date: Thu, 19 Apr 2012 20:47:28 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: vpascual@acmepacket.com, jmillan@aliax.net, sipcore@ietf.org
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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--e89a8f2348a787b1fa04be09ab43
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On Wed, Apr 18, 2012 at 11:21 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrot=
e:

> 2012/4/18 Nataraju A.B <nataraju.sip@gmail.com>:
> > Hi Inaki,
> >
> > Please find some more comments inline...
> >
> >
> > Please let me know, the reasoning behind single transport parameter and
> two
> > sent-protocol values.
> > sent-protocol : WS and WSS
> > transport =3D ws (only one new value)
> >
> > This might be confusing in future...
>
> Not really. In the same way that using SIP over TLS-TCP means
> ;transport=3Dtcp with sips schema, or sending SIP over TLS-SCTP means
> ;transport=3Dsctp with sips schema. So sending SIP over TLS-WebSocket
> must involve ;transport=3Dws as sips schema.
>
> Note that ;transport=3Dtls has never been defined (nor in RFC 2543) and
> it seems to appear in RFC 3261 due to dubious reasons, but it is not
> used in RFC 3261.
>
> Please check http://tools.ietf.org/html/rfc5630#section-3.1.4.
>
>
>
>
>
>
> > if we want to use SIP URI with any other secure communication mechanism
> like
> > WSS, might require
> > update to RFC5630. like
> > <rfc5630> sec 3.1.3.
> >
> >    If one wants to use "best-effort TLS/WSS" for SIP, one just needs to
> use
> >    a SIP URI, and send the request over TLS ** or any other secure
> transport
> > mechanism like WSS**
> >
> > </rfc5630>
>
> TLS is a extra-layer on top of TCP and also on top of WebSocket. When
> applied on top of TCP transport it is commonly named "TLS", but it
> should be better to call it "TLS-TCP" (in the same way that TLS over
> SCTP is named "TLS-SCTP" for Via transport value in RFC 4168.
>
>
>
>
>
> >> >    Extra closing double quote in contact header.
> >>
> >> Typo, thanks :)
> >>
> > [ABN] similar typo mistake in message F3 as well.
>
> Good catch.
>
>
>
>
> >> The example uses a secure WS connection (on top of TLS) but does not
> >> use SIPS schema.
> >>
> >
> > [ABN] Probably for the sake of clarity, we can give call flow for Secur=
e
> Web
> > Socket (WSS)
> > and non-secure Web Socket (WS). this content is already covered in
> different
> > sections (this doc and rfc6455 as well). But putting it together with a
> call
> > flow would make the concept more clearer.
>
> The problem is that, trying to include a full secure call would lead
> to multiple mail threads discussing SIPS stuf that is really difficult
> to understand and implement. There is a RFC trying to improve/explain
> it (RFC 5630). I don't think the mission of this draft is to clarify
> SIPS usage.
>

[ABN] Inaki, I understand your point. Anyway you would have analyzed
thoroughly to address the secure and non-secure websocket scenarios. The
same information can serve as a ready reference for people to use this
draft.

Not a problem, even if you drop this comment (rfc4168 don't do this....:)

>
>
>
> Thanks a lot.
>
>
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>



--=20
Thanks,
Nataraju A.B.

--e89a8f2348a787b1fa04be09ab43
Content-Type: text/html; charset=ISO-8859-1
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<br><br><div class=3D"gmail_quote">On Wed, Apr 18, 2012 at 11:21 PM, I=F1ak=
i Baz Castillo <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" targe=
t=3D"_blank">ibc@aliax.net</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex">

2012/4/18 Nataraju A.B &lt;<a href=3D"mailto:nataraju.sip@gmail.com" target=
=3D"_blank">nataraju.sip@gmail.com</a>&gt;:<br>
<div>&gt; Hi Inaki,<br>
&gt;<br>
&gt; Please find some more comments inline...<br>
&gt;<br>
&gt;<br>
&gt; Please let me know, the reasoning behind single transport parameter an=
d two<br>
&gt; sent-protocol values.<br>
&gt; sent-protocol : WS and WSS<br>
&gt; transport =3D ws (only one new value)<br>
&gt;<br>
&gt; This might be confusing in future...<br>
<br>
</div>Not really. In the same way that using SIP over TLS-TCP means<br>
;transport=3Dtcp with sips schema, or sending SIP over TLS-SCTP means<br>
;transport=3Dsctp with sips schema. So sending SIP over TLS-WebSocket<br>
must involve ;transport=3Dws as sips schema.<br>
<br>
Note that ;transport=3Dtls has never been defined (nor in RFC 2543) and<br>
it seems to appear in RFC 3261 due to dubious reasons, but it is not<br>
used in RFC 3261.<br>
<br>
Please check <a href=3D"http://tools.ietf.org/html/rfc5630#section-3.1.4" t=
arget=3D"_blank">http://tools.ietf.org/html/rfc5630#section-3.1.4</a>.<br>
<div><br>
<br>
<br>
<br>
<br>
<br>
&gt; if we want to use SIP URI with any other secure communication mechanis=
m like<br>
&gt; WSS, might require<br>
&gt; update to RFC5630. like<br>
&gt; &lt;rfc5630&gt; sec=A03.1.3.<br>
&gt;<br>
&gt; =A0 =A0If one wants to use &quot;best-effort TLS/WSS&quot; for SIP, on=
e just needs to use<br>
&gt; =A0 =A0a SIP URI, and send the request over TLS ** or any other secure=
 transport<br>
&gt; mechanism like WSS**<br>
&gt;<br>
&gt; &lt;/rfc5630&gt;<br>
<br>
</div>TLS is a extra-layer on top of TCP and also on top of WebSocket. When=
<br>
applied on top of TCP transport it is commonly named &quot;TLS&quot;, but i=
t<br>
should be better to call it &quot;TLS-TCP&quot; (in the same way that TLS o=
ver<br>
SCTP is named &quot;TLS-SCTP&quot; for Via transport value in RFC 4168.<br>
<div><br>
<br>
<br>
<br>
<br>
&gt;&gt; &gt; =A0 =A0Extra closing double quote in contact header.<br>
&gt;&gt;<br>
&gt;&gt; Typo, thanks :)<br>
&gt;&gt;<br>
&gt; [ABN] similar typo mistake in message F3 as well.<br>
<br>
</div>Good catch.<br>
<div><br>
<br>
<br>
<br>
&gt;&gt; The example uses a secure WS connection (on top of TLS) but does n=
ot<br>
&gt;&gt; use SIPS schema.<br>
&gt;&gt;<br>
&gt;<br>
&gt; [ABN] Probably for the sake of clarity, we can give call flow for Secu=
re Web<br>
&gt; Socket (WSS)<br>
&gt; and non-secure Web Socket (WS). this content is already covered in dif=
ferent<br>
&gt; sections (this doc and rfc6455 as well). But putting it together with =
a call<br>
&gt; flow would make the concept more clearer.<br>
<br>
</div>The problem is that, trying to include a full secure call would lead<=
br>
to multiple mail threads discussing SIPS stuf that is really difficult<br>
to understand and implement. There is a RFC trying to improve/explain<br>
it (RFC 5630). I don&#39;t think the mission of this draft is to clarify<br=
>
SIPS usage.<br></blockquote><div>=A0</div><div>[ABN] Inaki, I understand yo=
ur point. Anyway you would have analyzed thoroughly to address the secure a=
nd non-secure websocket scenarios. The same information can serve as a read=
y reference for people to use this draft.</div>

<div><br></div><div>Not a problem, even if you drop this comment (rfc4168 d=
on&#39;t do this....:)</div><blockquote class=3D"gmail_quote" style=3D"marg=
in:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
<br>
<br>
Thanks a lot.<br>
<div><div><br>
<br>
<br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>

<br>

--e89a8f2348a787b1fa04be09ab43--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/19 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> I agree with Christer for mandating path in case REGISTER is the first me=
ssage
> and it is not required to be mandated in case INVITE is the first message=
.

Hi Partha, honestly I don't think it's a good idea to mandate SIP
topology requirements in a draft describing a SIP transport (just it).
We tryed to mandate Outbound RFC 5626 and such a requirement was
removed in IETF #83 due to consensus. So mandating Path is mostly the
same "error".

It was clearly agreed in Paris that the draft should not assume a SIP
network topology, but also that it should include a *informative*
section for the most expected use case (clients like SIP browsers).
This is already done in Appendix I which describes such an scenario
and all the SIP standards (Outbound, Path, GRUU) that could help to
make it to work. But *just* it.

Again, the draft does NOT assume that clients are web browsers so
please consider it for future suggestions about the draft.


> In the mentioned below callflow, P1 should be some prior knowledge about =
Bob
> that Bob supports websocket transport which is not always feasible throug=
h
> standard mechanism. Also, the callflow is not generic enough to handle
> SIP entity which supports UDP & TCP transport only.

I don't agree. But if it is easier to understand imagine that alice,
bob and the proxy/registrar all of them implement a WebSocket client
and server, so alice can open a WS connection to proxy and also to
bob, and the same for bob. So no Record-Route required. That's all.



Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Thu, 19 Apr 2012 21:34:47 +0530
Message-ID: <CA+rAfUMyAw0pSy2bHr1cxhAX05mKOahW5mK5d0TpkYt_cNiN5A@mail.gmail.com>
From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Content-Type: multipart/alternative; boundary=e89a8f235097b85d6104be0a543a
Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Incoming call notification for websocket transport
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--e89a8f235097b85d6104be0a543a
Content-Type: text/plain; charset=ISO-8859-1
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Partha,
After some thinking, I also found we need incoming SIP requests @ websocket
clients for presence services like gtalk, yahoo, skype messengers...

On Thu, Apr 19, 2012 at 8:17 AM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

>  Hi Nataraju,****
>
> ** **
>
> [ABN] Partha, ****
>
>           Can you name some scenario which need this incoming SIP request=
s
> @ the WebSocket client ???? ****
>
> ** **
>
> There are two scenario which I could think of****
>
> ** **
>
> **1)      **Skype sort of websocket application in endpoint ****
>
> **2)      **B2BUA which listen on websocket to receive the calls without
> registration****
>
> ** **
>
> Thanks****
>
> Partha****
>
> ** **
>
> *From:* Nataraju A.B [mailto:nataraju.sip@gmail.com]
> *Sent:* Wednesday, April 18, 2012 3:50 PM
> *To:* Ravindran, Parthasarathi
> *Cc:* Sa=FAl Ibarra Corretg=E9; SIPCORE (Session Initiation Protocol Core=
)
> WG; SIPCORE Chairs; I=F1aki Baz Castillo
> *Subject:* Re: [sipcore] Stateless or Transaction stateful proxy possible
> in Websocket transport? [RE: Call for Adoption:
> draft-ibc-sipcore-sip-websocket-02]****
>
> ** **
>
> ** **
>
> On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasarathi <
> pravindran@sonusnet.com> wrote:****
>
> Even if it is the limitations of websocket, SIP UA MUST listen in
> UDP & TCP as per RFC 3261 and not adhere to this requirement which MUST
> be considered as a violation of RFC 3261. In case you think that this dra=
ft
> does not violate any RFC 3261 for SIP UA, the Current text in the draft i=
s
> not
> clear enough, Please mention it explicitly in the draft that
>
> "As per this specification, SIP UA MUST support UDP, TCP transport apart
> from Websocket as a transport"
>
> This information helps for the implementers who wish to use this draft fo=
r
> RTCWeb client to SIP Interop in the standard way. I'm writing this mail a=
s
> most of the folks support this draft for interworking RTCWeb client to SI=
P
> in the standard manner as per my discussion in IETF-83.
>
> I'll write the separate mail thread for discussing incoming SIP over
> websocket
> as it is one another major open item in this draft.****
>
> [ABN] Partha, ****
>
>           Can you name some scenario which need this incoming SIP request=
s
> @ the WebSocket client ???? ****
>
>
> Thanks
> Partha****
>
>
> >-----Original Message-----
> >From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
> >Sent: Wednesday, April 18, 2012 3:23 PM
> >To: Ravindran, Parthasarathi****
>
> >Cc: I=F1aki Baz Castillo; SIPCORE (Session Initiation Protocol Core) WG;
> >SIPCORE Chairs; Nataraju A.B
> >Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
> >in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
> >websocket-02]
> >
> >****
>
> >On Apr 18, 2012, at 11:35 AM, Ravindran, Parthasarathi wrote:
> >
> >> As per your below statement, JavaScript with SIP and RTCWeb Browser
> >MUST NOT use this draft as it violates RFC 3261.
> >>
> >
> >It doesn't, see my previous email explaining why.
> >
> >> Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websocket
> >will not listen for the incoming call without registration but in case
> >of your draft is written with SIP softphone (smartphone) in mind , it is
> >possible to listen in Port 80 or any other Websocket port which is not
> >addressed in this draft till now. We need to discuss and sort it out.
> >>
> >
> >The fact that a JS SIP client (running in a web browser) does not listen
> >for incoming connections is a limitation imposed by the WebSocket API,
> >not by this draft. This specification defines 2 entities: a SIP
> >WebSocket Client and a SIP WebSocket Server, so nobody stops you from
> >creating an entity that implements both and can accept incoming
> >WebSocket connections. Of course, such an application would not run in a
> >browser.
> >
> >
> >Regards,
> >
> >--
> >Sa=FAl Ibarra Corretg=E9
> >AG Projects
> >
> >****
>
>
>
> ****
>
> ** **
>
> --
> Thanks,****
>
> Nataraju A.B.****
>
> ** **
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>


--=20
Thanks,
Nataraju A.B.

--e89a8f235097b85d6104be0a543a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Partha,=A0<div>After some thinking, I also found we need incoming SIP reque=
sts @ websocket clients for presence services like gtalk, yahoo, skype mess=
engers...</div><div><br><div class=3D"gmail_quote">On Thu, Apr 19, 2012 at =
8:17 AM, Ravindran, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:p=
ravindran@sonusnet.com">pravindran@sonusnet.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">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Hi Nataraju,<u></u><u></u=
></span></p>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal">[ABN] Partha,=A0<u></u><u></u></p>
<p class=3D"MsoNormal">=A0 =A0 =A0 =A0 =A0 Can you name some scenario which=
 need this incoming SIP requests @ the WebSocket client ????=A0<u></u><u></=
u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">There are two scenario wh=
ich I could think of<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>1)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=A0=A0=A0=A0=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">Skype sort of websoc=
ket application in endpoint
<u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>2)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=A0=A0=A0=A0=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">B2BUA which listen o=
n websocket to receive the calls without registration<u></u><u></u></span><=
/p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Thanks<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Partha<u></u><u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=A0<u></u></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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Nataraju=
 A.B [mailto:<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">na=
taraju.sip@gmail.com</a>]
<br>
<b>Sent:</b> Wednesday, April 18, 2012 3:50 PM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> Sa=FAl Ibarra Corretg=E9; SIPCORE (Session Initiation Protocol C=
ore) WG; SIPCORE Chairs; I=F1aki Baz Castillo<br>
<b>Subject:</b> Re: [sipcore] Stateless or Transaction stateful proxy possi=
ble in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-w=
ebsocket-02]<u></u><u></u></span></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">On Wed, Apr 18, 2012 at 3:37 PM, Ravindran, Parthasa=
rathi &lt;<a href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">prav=
indran@sonusnet.com</a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">Even if it is the limitations of websocket, SIP UA M=
UST listen in<br>
UDP &amp; TCP as per RFC 3261 and not adhere to this requirement which MUST=
<br>
be considered as a violation of RFC 3261. In case you think that this draft=
<br>
does not violate any RFC 3261 for SIP UA, the Current text in the draft is =
not<br>
clear enough, Please mention it explicitly in the draft that<br>
<br>
&quot;As per this specification, SIP UA MUST support UDP, TCP transport apa=
rt<br>
from Websocket as a transport&quot;<br>
<br>
This information helps for the implementers who wish to use this draft for<=
br>
RTCWeb client to SIP Interop in the standard way. I&#39;m writing this mail=
 as<br>
most of the folks support this draft for interworking RTCWeb client to SIP<=
br>
in the standard manner as per my discussion in IETF-83.<br>
<br>
I&#39;ll write the separate mail thread for discussing incoming SIP over we=
bsocket<br>
as it is one another major open item in this draft.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">[ABN] Partha,=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=A0 =A0 =A0 =A0 =A0 Can you name some scenario which=
 need this incoming SIP requests @ the WebSocket client ????=A0<u></u><u></=
u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<p class=3D"MsoNormal"><br>
Thanks<br>
Partha<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><br>
&gt;-----Original Message-----<br>
&gt;From: Sa=FAl Ibarra Corretg=E9 [mailto:<a href=3D"mailto:saul@ag-projec=
ts.com" target=3D"_blank">saul@ag-projects.com</a>]<br>
&gt;Sent: Wednesday, April 18, 2012 3:23 PM<br>
&gt;To: Ravindran, Parthasarathi<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&gt;Cc: I=F1aki Baz Castillo; SIPCORE (Session Initi=
ation Protocol Core) WG;<br>
&gt;SIPCORE Chairs; Nataraju A.B<br>
&gt;Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=
<br>
&gt;in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-<=
br>
&gt;websocket-02]<br>
&gt;<br>
&gt;<u></u><u></u></p>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt">&gt;On Apr 18, 2012, =
at 11:35 AM, Ravindran, Parthasarathi wrote:<br>
&gt;<br>
&gt;&gt; As per your below statement, JavaScript with SIP and RTCWeb Browse=
r<br>
&gt;MUST NOT use this draft as it violates RFC 3261.<br>
&gt;&gt;<br>
&gt;<br>
&gt;It doesn&#39;t, see my previous email explaining why.<br>
&gt;<br>
&gt;&gt; Also in IETF-83 SIPCore meeting, I noticed that SIP UA over websoc=
ket<br>
&gt;will not listen for the incoming call without registration but in case<=
br>
&gt;of your draft is written with SIP softphone (smartphone) in mind , it i=
s<br>
&gt;possible to listen in Port 80 or any other Websocket port which is not<=
br>
&gt;addressed in this draft till now. We need to discuss and sort it out.<b=
r>
&gt;&gt;<br>
&gt;<br>
&gt;The fact that a JS SIP client (running in a web browser) does not liste=
n<br>
&gt;for incoming connections is a limitation imposed by the WebSocket API,<=
br>
&gt;not by this draft. This specification defines 2 entities: a SIP<br>
&gt;WebSocket Client and a SIP WebSocket Server, so nobody stops you from<b=
r>
&gt;creating an entity that implements both and can accept incoming<br>
&gt;WebSocket connections. Of course, such an application would not run in =
a<br>
&gt;browser.<br>
&gt;<br>
&gt;<br>
&gt;Regards,<br>
&gt;<br>
&gt;--<br>
&gt;Sa=FAl Ibarra Corretg=E9<br>
&gt;AG Projects<br>
&gt;<br>
&gt;<span class=3D"HOEnZb"><font color=3D"#888888"><u></u><u></u></font></s=
pan></p><span class=3D"HOEnZb"><font color=3D"#888888">
</font></span></div><span class=3D"HOEnZb"><font color=3D"#888888">
</font></span></div><span class=3D"HOEnZb"><font color=3D"#888888">
</font></span></blockquote><span class=3D"HOEnZb"><font color=3D"#888888">
</font></span></div><span class=3D"HOEnZb"><font color=3D"#888888">
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><u></u><u></u></p>
</div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</font></span></div>
</div>
</div>

<br>_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
<br></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font co=
lor=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">=
Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courie=
r new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>
</div>

--e89a8f235097b85d6104be0a543a--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 19 Apr 2012 18:10:58 +0200
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To: "Nataraju A.B" <nataraju.sip@gmail.com>
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] Incoming call notification for websocket transport
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2012/4/19 Nataraju A.B <nataraju.sip@gmail.com>:
> After some thinking, I also found we need incoming SIP requests @ websock=
et
> clients for presence services like gtalk, yahoo, skype messengers...

I don't understand. Presence in SIP is based on PUBLISH, SUBSCRIBE,
NOTIFY. If the SIP client is connected to an Outbound proxy the SIP
client would reuse the same connection for sending PUBLISH, SUBSCRIBES
and so (since there is already a open connection open to the outbound
proxy as RFC 3261 states) and therefore also for receiving NOTIFY
requests (within the subscription dialog).

But again, nothing prevents somebody to code a SIP WebSocket entity
capable of creating outbound WS connections and listening for inbound
WS connections. Please don't assume that the draft just talks about
web browsers.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From ibc@aliax.net  Thu Apr 19 09:23:11 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 19 Apr 2012 18:22:46 +0200
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Cc: vpascual@acmepacket.com, jmillan@aliax.net, sipcore@ietf.org
Subject: Re: [sipcore] Call flow for Secure Web Socket creation --- draft-ibc-sipcore-sip-websocket-02.txt
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2012/4/17 Nataraju A.B <nataraju.sip@gmail.com>:
> Hi All,
>
> I have written a call flow for the secure web socket usage with TCP.
>
> Please go through the same and share if any changes required.

Hi Nataraju, I think the flow is ok, but honestly we wouldn't like to
deal too much with SIPS stuf within the draft, is not the mission of
the draft to clarify all the SIPS complexity and issues. IMHO RFC 5630
is the good place for that, and RFC 5630 is already referenced by RFC
3261 (it updates RFC 3261 in fact) so IMHO the draft does not need to
mention it (we don't want to open Pandora'x box and I think we
shouldn't).

As said before in a previous mail, using a secure transport without
SIPS schema is perfectly valid according to RFC 3261 and RFC 5630:

- SIPS requires secure transport in all the path,
- but a secure transport does not require SIPS schema.

http://tools.ietf.org/html/rfc5630#section-3.1.3


Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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2012/4/19 Nataraju A.B <nataraju.sip@gmail.com>:
>> The problem is that, trying to include a full secure call would lead
>> to multiple mail threads discussing SIPS stuf that is really difficult
>> to understand and implement. There is a RFC trying to improve/explain
>> it (RFC 5630). I don't think the mission of this draft is to clarify
>> SIPS usage.
>
>
> [ABN] Inaki, I understand your point. Anyway you would have analyzed
> thoroughly to address the secure and non-secure websocket scenarios. The
> same information can serve as a ready reference for people to use this
> draft.


In fact, examples in the draft use secure WebSocket connections, so
IMHO we "help" people implementing secure SIP. But SIPS schema is
another history nobody wants to deal with (that is my opinion after
many threads about it).

The most realistic scenario is that in which clients connect to their
proxy/server using a secure transport and the proxy decides whether to
use a secure or non-secure transport for communicating with other
servers, gateways or SIP networks, and examples in the draft show that
case.

If somebody wants to use SIPS with WebSocket transport, then just use
SIPS schema everywhere and WSS connections (and read RFC 5630) :)


Regards.
--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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From dworley@avaya.com  Thu Apr 19 19:34:06 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 19 Apr 2012 22:33:16 -0400
Thread-Topic: Comments on draft-ietf-sipcore-rfc4244bis-07 - sections 10.1 and 11-end
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Subject: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 - sections 10.1 and 11-end
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Correction of my comments on 10.3:  I proposed removing all of item 6:

   6.  Missing entity: If the request clearly has a gap in the hi-entry
       (e.g. by evaluating via header to the existing request history to
       see if it traversed a domain which doesn't exist in History-Info
       header field.), the entity adding an hi-entry MUST add an index a
       digit of "0" to the current index prior to adding appropriate
       index for the action to be taken.  If the index of the last hi-
       entry in the request received was 1.1.2 and there was a missing
       hi-entry and the request was being forwarded to the next hop, the
       resulting index will be 1.1.2.0.1.

But I should have restricted it to removing the "condition", namely
"(e.g. by evaluating via header to the existing request history to see
if it traversed a domain which doesn't exist in History-Info header
field.)"

10.1.1.  Indicating Privacy

   In this case, the
   intermediary MUST set a Privacy header field to a priv-value of
   "history" and include the Privacy header field in the hi-targeted-to-
   uri, for each new hi-entry added by the intermediary, as the request
   is retargeted within the domain for which the SIP entity is
   responsible.  Note, that the hi-entries that are added to a request
   from the cache are excluded in this case since the appropriate
   privacy was set for those hi-entries when they were originally added
   to a request.  The intermediary MUST NOT include any other priv-
   values in this Privacy header field.

The phrasing here is awkward.  I think this would be better:

   In this case, the intermediary MUST construct a Privacy header
   field with the single priv-value of "history" and include the
   Privacy header field in the hi-targeted-to-uri, for each new
   hi-entry created by the intermediary, as the request is retargeted
   within the domain for which the SIP entity is responsible.
   (Note that hi-entries that are added to a request
   from the cache are not modified by this process.)

However, I'm uncertain of the phrase "is retargeted within the
domain".  Does that mean retargeting outside the domain of interest
cannot have this process applied?  Perhaps we want to leave the
determination to the intermediary, since it is doing the modification
on its own behalf.  In that case, we would change the end of the
sentence to:

   ... for each new hi-entry created by the intermediary whose
   hi-targeted-to-uri it wishes to privacy protect.

   Finally, the terminator of the request may not want to reveal the

I think it would be more standard to say

   A UAS may not want to reveal the

and replace "terminator" with "UAS" in the rest of the paragraph.

10.1.2.  Applying Privacy

   When a request is retargeted to a URI associated with a domain for
   which the SIP intermediary is not responsible or a response is
   forwarded to a domain for which the SIP intermediary is not
   responsible, a Privacy Service at the boundary of the domain applies

It would be shorter to say

   When a SIP message is forwarded to a domain for which the SIP
   intermediary is not responsible ...

I think that has the same effect.

   which a SIP intermediary is responsible, are anonymized by the

probably should be "which >>>the<<< SIP intermediary is responsible",
referring to the intermediary discussed in the first paragraph.

   URIs containing a domain of anonymous.invalid (e.g.,
   anonymous@anonymous.invalid).  If there is a Privacy header field in

Strictly, that should be "sip:anonymous@anonymous.invalid" and/or
"sips:anonymous@anonymous.invalid".  Perhaps it would be better to not
describe the URI at all but just refer to RFC 3323.  However, the
algorithm requires that H-I processing can *recognize* entries that
have already been anonymized, so we have to add a condition:

   The Privacy Service MUST change any hi-targeted-to-uris in
   >>>these<<< hi-entries that have not been anonymized (evidenced by
   their domain not being "anonymous.invalid") to anonymous URIs
   containing a domain of "anonymous.invalid" as described in RFC 3323
   section 4.1.

also:

   If there is a Privacy header field in the "headers" component of
   the hi-targeted-to-uri in >>>these<<< hi-entries(?), then the
   Privacy header field value MUST be removed from the hi-entry.

Similarly in the next paragraph:

   If there is not a Privacy header field in the message header of the
   request or response that is being forwarded, but there is a Privacy
   header field with a priv-value of "history" in the "headers"
   component in any of the hi-targeted-uris in the hi-entries
   associated with the domain for which a SIP intermediary is
   responsible, then the Privacy Service MUST update those
   hi-targeted-to-uris >>>as described above.<<<  Any other priv-values
   in the Privacy header field in the "headers" component of the
   hi-targeted-to-uris in the hi-entries MUST be ignored.
   >>>[Last sentence deleted, as it is included in the "as above"
   prescription.]<<<

Maybe there is a better way to factor out the "anonymize and remove
Privacy" operation.

11.  Application Considerations

   The SIP entity MUST determine if there are gaps in the indices.

This is strictly true, but I don't think it's the sense we want.
There is also the problem that the two types of "gap" are distinctly
different in what they reveal.  Maybe something fuzzier like:

   The SIP entity MUST be prepared to process effectively messages
   whose hi-entries show evidence of "gaps", that is, situations that
   reveal that not all of the forks of the request have been recorded
   in the hi-entries.

   Gaps are possible if the request is forwarded through
   intermediaries that do not support the History-info header field and
   are reflected by the existence of multiple hi-entries with an digit
   of "0" e.g. "1.1.0.1".

Replace "digit" with "integer" as discussed before, and change "an",
or rather leave it as "an" because we're changing "digit" to integer.

But "multiple" should be deleted.  In particular, when a UAS receives
a request whose request-URI does not match the last hi-entry, it adds
a "gap" hi-entry, but since the UAS does not forward the request
onward, it does not add any further hi-entries with "0"s.  But in this
case, the request *has* had a gap in H-I processing.

   Gaps are also possible in the case of
   parallel forking if there is an outstanding request at the time the
   SIP entity sends a response as described in Section 9.4, in which
   case the gap will not be visible as the branch responsible for the
   gap wasn't on the path of the request received.

I think this sentence needs to be modified strongly:

   Gaps are also possible in the case of
   parallel forking if there is an outstanding request at the time the
   SIP entity sends a >>>message<<<.

A gap (missing branch) due to an outstanding request can be revealed
in both requests and responses.  The final part of the sentence I
can't understand.  I wonder if there was a further sentence which was
damaged and joined to this previous sentence.

12.  Application Specific Usage

It would be wise to mark this section as non-normative:

   Following are possible application-specific usages of History-Info.

and we could add:

   See also the example call flows in Appendix B.

Dale

From dworley@avaya.com  Thu Apr 19 19:35:21 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 19 Apr 2012 22:34:10 -0400
Thread-Topic: Comments on draft-ietf-sipcore-rfc4244bis-07 - substantival questions
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Subject: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 - substantival questions
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Here are a few questions about how 4244bis operates in certain
situations.  They may have been discussed already.  If so, I hope that
they have straightforward answers (which would show that the mechanism
is clean and thought through).

I. Numbering of gaps

When an entity receives a request, it checks to see whether the last
hi-entry matches the request-URI, and if not, adds an hi-entry to
document the difference:

   6.  Missing entity: If the request clearly has a gap in the hi-entry
       (e.g. by evaluating via header to the existing request history to
       see if it traversed a domain which doesn't exist in History-Info
       header field.), the entity adding an hi-entry MUST add an index a
       digit of "0" to the current index prior to adding appropriate
       index for the action to be taken.  If the index of the last hi-
       entry in the request received was 1.1.2 and there was a missing
       hi-entry and the request was being forwarded to the next hop, the
       resulting index will be 1.1.2.0.1.

However, if in the un-logged processing of the request, the request
forked, the two forks can then receive hi-entries with the *same*
gap-indicating hi-index.  These hi-entries can be carried upstream in
responses and accumulate in the cache at some point upstream of the
gap, resulting in two entries in the cache with the same hi-index.

Modifying the example in Figure 1, assume that biloxi.example.com does
not understand H-I.  If I understand the details correctly, Bob@pc and
Bob@phone will independently create hi-entries with hi-index 1.1.0.
The two entries will be put into the cache at atlanta.exmple.com:

   Alice   atlanta.example.com  biloxi.example.com   Bob@pc  Bob@phone
   |                |                |                |          |
   |   INVITE sip:bob@biloxi.example.com;p=3Dx          |          |
   |--------------->|                |                |          |
   | Supported: histinfo             |                |          |
   |                |                |                |          |
   |                |   INVITE sip:bob@biloxi.example.com;p=3Dx    |
   |                |--------------->|                |          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   |                |                |                |          |
   |                |                |   INVITE sip:bob@192.0.2.3|
   |                |                |--------------->|          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   |                |                |                |          |
   | As modified by Bob@pc due to section 9.1:        |          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.3>;index=3D1.1.0    |          |
   |                |                |                |          |
   |                |                |   INVITE sip:bob@192.0.2.7|
   |                |                |-------------------------->|
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   |                |                |                |          |
   | As modified by Bob@phone due to section 9.1:     |          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.7>;index=3D1.1.0    |          |
   |                |                |     4xx        |          |
   |                |                |<---------------|          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.3>;index=3D1.1.0    |          |
   |                |                |                |          |
   |                |     4xx        |                |          |
   |                |<---------------|                |          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.3>;index=3D1.1.0    |          |
   |                |                |                |          |
   |                |                |      200       |          |
   |                |                |<--------------------------|
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.7>;index=3D1.1.0    |          |
   |                |                |                |          |
   |     200        |                |                |          |
   |<---------------|                |                |          |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1      |
   | History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   | History-Info: <sip:bob@192.0.2.3&Reason=3DSIP%3Bcause%3D4xx>;index=3D1=
.1.0
   | History-Info: <sip:bob@192.0.2.7>;index=3D1.1.0    |          |
   |                |                |                |          |
   |     ACK        |                |                |          |
   |--------------->|    ACK         |                |          |
   |                |--------------->|     ACK        |          |
   |                |                |--------------->|          |

One way to avoid this problem is to add a random number after the "0",
so the hi-entries created by the two UASs are (probably) different:

   History-Info: <sip:bob@192.0.2.3>;index=3D1.1.0.42949
   History-Info: <sip:bob@192.0.2.7>;index=3D1.1.0.34317

To do this consistently and statelessly, one can hash the branch
parameter of the topmost Via of the incoming request, extract 16 bits
and add 1.  By the Birthday Paradox, using 16 bits for the number,
twenty forks can be accommodated with a low probability (0.3%) of
collision.

With this change, the final response would have this H-I:

   History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1
   History-Info: <sip:bob@biloxi.example.com;p=3Dx>;np=3D1;index=3D1.1
   History-Info: <sip:bob@192.0.2.7>;index=3D1.1.0.34317
   History-Info: <sip:bob@192.0.2.3&Reason=3DSIP%3Bcause%3D4xx>;index=3D1.1=
.0.42949

This approach implies that H-I processing must allow index values of
at least 65537; otherwise 100 would suffice in practice and the limit
would not need to be specified.

II. Reason headers added by processing provisional responses

Looking at the example in section B.1 ... I don't see what I need,
which is the H-I from the 180 Ringing response or the 200 OK response
which is sent to Alice.  It would be useful to have those shown.

But based on section 10.2:

   If the SIP response does not
   contain a Reason header field, the SIP entity MUST include a Reason
   header field, containing the SIP Response Code, in the "headers"
   component of the hi-targeted-to-uri in the last hi-entry added to the
   cache, unless the hi-targeted-to-uri is a Tel-URI.

there must be a "Reason=3DSIP%3Bcause%3D180" attached to the URI in
hi-entry 1.2 because request 1.2 had a 180 response.

I suspect that my error is that the above rule is only intended to be
applied for (non-2xx) *final* responses.

III. What is added to "Reason" elements?

What gets added to "Reason" elements is:

- "Reason" header values received in responses

- SIP (final) failure response codes, if the response did not have a
  "Reason" header value

Have we deliberately excluded "SIP 2xx" reason values and SIP response
codes if an explicit Reason was provided?  At first glance, it seems
to me that a SIP failure final response code is always useful to know,
even if there was an accompanying Reason value.  But perhaps brevity
is more valuable.

IV. Locating the hi-entry corresponding to a message

By the algorithms, in a request, the final hi-entry corresponds to the
request and shows the request's position in the forking structure.
(Or rather, it will correspond if all SIP entities support H-I.)

But there is no such correspondence in a response, because a response
can carry hi-entries from responses to requests generated after the
request that generated the response.

As a simple case, a 4xx response with

   History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1
   History-Info: <sip:bob@192.0.2.3&Reason=3DSIP%3Bcause%3D4xx>;index=3D1.1
   History-Info: <sip:bob@10.11.12.13&Reason=3DSIP%3Bcause%3D4xx>;index=3D1=
.2

could have ultimately been generated by either sip:bob@192.0.2.3 or
sip:bob@10.11.12.13; there is no way to distinguish the two cases.

There is no harm in this as long as we don't need the recipient of a
response to be able to determine the ultimate source of a response.

---------------------------------------------------------------------------=
----------

That is the last of my review comments.  Now I have to respond to the
authors' comments on my review comments.

Dale

From saul@ag-projects.com  Fri Apr 20 01:04:42 2012
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <4F8ED41A.6060703@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E2281EE@inba-mail01.sonusnet.com> <CALiegfkBxu_RDFoFhwSchAeqni-ZAVY6tT6tjUP810_rNZuMpg@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C428E2FA4@ESESSCMS0356.eemea.ericsson.se> <387F9047F55E8C42850AD6B3A7A03C6C0E228315@inba-mail01.sonusnet.com> <CALiegf=Hm+LEoh7jz-0BGiU5gNzPi9vQmSjJZ0f6yZnNwi_v5w@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E22882A@inba-mail01.sonusnet.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi Partha,

On Apr 20, 2012, at 2:39 AM, Ravindran, Parthasarathi wrote:

> Hi Inaki,
>=20
> There are only two options which I could think of forward for this =
draft
>=20
> 1) As I earlier mentioned, Websocket transport works well only in =
specific SIP topology and not in all topology as it is not capable of =
supporting UDP/TCP transport. In case the draft is only going to specify =
transport, then SIP entities support this draft has to have compliance =
with RFC 3261 (UDP, TCP transport) as well and particularly you have to =
mention how Javascript based SIP stack will compliance with RFC 3261.
>=20

SIP over UDP also doesn't work on all topologies, just think about NAT. =
IMO this draft should not target at using SIP in web browsers using =
JavaScript, it should just define a new transport for the SIP protocol, =
which is exactly what it does. There are, indeed, inherent limitations =
when using SIP in a web browser, but these limitations are imposed by =
the WebSocket API, *not* by this draft.

Also note that there is no "IETF police", if I would implement this =
draft for using a SIP client within a browser I would not implement UDP =
or TCP support whatsoever, because they can't be used *in a browser*. =
Sue me :-)

> 2) Alternative is to specify SIP topology requirement wherein =
websocket really applies. Include the related header like record-route & =
path with the detail information where these headers are mandated.
>=20
> IMHO, the current draft is not clear enough about SIP over Websocket =
transport usage.
>=20

I fail to understand what is not clear about it. Of course WebSockets =
are used in browsers nearly 100% of the time, but this draft doesn't =
restrict itself to that model. You could write a desktop SIP client =
using WebSocket as a transport, and you could listen for incoming =
connections since limitations such as those in web browsers wouldn't =
apply. So, please, can you clarify *what* is it not clear about it?


Thanks and regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From pravindran@sonusnet.com  Fri Apr 20 01:59:06 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Saul,

My reading of your statement "There are, indeed, inherent limitations when =
using SIP in a web browser, but these limitations are imposed by the WebSoc=
ket API, *not* by this draft." is that websocket does not qualify as a tran=
sport for SIP as it is not possible to meet RFC 3261 requirement.

NAT is not SIP endpoint issue but it is a network issue. All NAT are not co=
nsidered as part of RFC 3261 design but it is resolved as part of Managing =
Client-Initiated Connections in the Session Initiation Protocol (SIP) RFC (=
5626).=20

Thanks
Partha

Note: AFAIK, NAT issue in SIP is one of the reason for not selecting SIP as=
 part of signaling/transport protocol in RTCWeb.

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>Sent: Friday, April 20, 2012 1:35 PM
>To: Ravindran, Parthasarathi
>Cc: I=F1aki Baz Castillo; sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>Hi Partha,
>
>On Apr 20, 2012, at 2:39 AM, Ravindran, Parthasarathi wrote:
>
>> Hi Inaki,
>>
>> There are only two options which I could think of forward for this
>draft
>>
>> 1) As I earlier mentioned, Websocket transport works well only in
>specific SIP topology and not in all topology as it is not capable of
>supporting UDP/TCP transport. In case the draft is only going to specify
>transport, then SIP entities support this draft has to have compliance
>with RFC 3261 (UDP, TCP transport) as well and particularly you have to
>mention how Javascript based SIP stack will compliance with RFC 3261.
>>
>
>SIP over UDP also doesn't work on all topologies, just think about NAT.
>IMO this draft should not target at using SIP in web browsers using
>JavaScript, it should just define a new transport for the SIP protocol,
>which is exactly what it does. There are, indeed, inherent limitations
>when using SIP in a web browser, but these limitations are imposed by
>the WebSocket API, *not* by this draft.
>
>Also note that there is no "IETF police", if I would implement this
>draft for using a SIP client within a browser I would not implement UDP
>or TCP support whatsoever, because they can't be used *in a browser*.
>Sue me :-)
>
>> 2) Alternative is to specify SIP topology requirement wherein
>websocket really applies. Include the related header like record-route &
>path with the detail information where these headers are mandated.
>>
>> IMHO, the current draft is not clear enough about SIP over Websocket
>transport usage.
>>
>
>I fail to understand what is not clear about it. Of course WebSockets
>are used in browsers nearly 100% of the time, but this draft doesn't
>restrict itself to that model. You could write a desktop SIP client
>using WebSocket as a transport, and you could listen for incoming
>connections since limitations such as those in web browsers wouldn't
>apply. So, please, can you clarify *what* is it not clear about it?
>
>
>Thanks and regards,
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>


From christer.holmberg@ericsson.com  Fri Apr 20 02:01:47 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Date: Fri, 20 Apr 2012 11:01:42 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Why not write a separate draft, which updates 3261 by saying that all trans=
ports are optional... That's how the real world looks, anyway.

Regards,

Christer

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ravindran, Parthasarathi
Sent: 20. huhtikuuta 2012 11:59
To: Sa=FAl Ibarra Corretg=E9
Cc: I=F1aki Baz Castillo; sipcore@ietf.org
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Saul,

My reading of your statement "There are, indeed, inherent limitations when =
using SIP in a web browser, but these limitations are imposed by the WebSoc=
ket API, *not* by this draft." is that websocket does not qualify as a tran=
sport for SIP as it is not possible to meet RFC 3261 requirement.

NAT is not SIP endpoint issue but it is a network issue. All NAT are not co=
nsidered as part of RFC 3261 design but it is resolved as part of Managing =
Client-Initiated Connections in the Session Initiation Protocol (SIP) RFC (=
5626).=20

Thanks
Partha

Note: AFAIK, NAT issue in SIP is one of the reason for not selecting SIP as=
 part of signaling/transport protocol in RTCWeb.

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>Sent: Friday, April 20, 2012 1:35 PM
>To: Ravindran, Parthasarathi
>Cc: I=F1aki Baz Castillo; sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=20
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-=20
>websocket-02]
>
>Hi Partha,
>
>On Apr 20, 2012, at 2:39 AM, Ravindran, Parthasarathi wrote:
>
>> Hi Inaki,
>>
>> There are only two options which I could think of forward for this
>draft
>>
>> 1) As I earlier mentioned, Websocket transport works well only in
>specific SIP topology and not in all topology as it is not capable of=20
>supporting UDP/TCP transport. In case the draft is only going to=20
>specify transport, then SIP entities support this draft has to have=20
>compliance with RFC 3261 (UDP, TCP transport) as well and particularly=20
>you have to mention how Javascript based SIP stack will compliance with RF=
C 3261.
>>
>
>SIP over UDP also doesn't work on all topologies, just think about NAT.
>IMO this draft should not target at using SIP in web browsers using=20
>JavaScript, it should just define a new transport for the SIP protocol,=20
>which is exactly what it does. There are, indeed, inherent limitations=20
>when using SIP in a web browser, but these limitations are imposed by=20
>the WebSocket API, *not* by this draft.
>
>Also note that there is no "IETF police", if I would implement this=20
>draft for using a SIP client within a browser I would not implement UDP=20
>or TCP support whatsoever, because they can't be used *in a browser*.
>Sue me :-)
>
>> 2) Alternative is to specify SIP topology requirement wherein
>websocket really applies. Include the related header like record-route=20
>& path with the detail information where these headers are mandated.
>>
>> IMHO, the current draft is not clear enough about SIP over Websocket
>transport usage.
>>
>
>I fail to understand what is not clear about it. Of course WebSockets=20
>are used in browsers nearly 100% of the time, but this draft doesn't=20
>restrict itself to that model. You could write a desktop SIP client=20
>using WebSocket as a transport, and you could listen for incoming=20
>connections since limitations such as those in web browsers wouldn't=20
>apply. So, please, can you clarify *what* is it not clear about it?
>
>
>Thanks and regards,
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On Apr 20, 2012, at 11:01 AM, Christer Holmberg wrote:

> Why not write a separate draft, which updates 3261 by saying that all =
transports are optional... That's how the real world looks, anyway.
>=20

+1

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi Pharta,

El d=EDa 20 de abril de 2012 10:58, Ravindran, Parthasarathi
<pravindran@sonusnet.com> escribi=F3:
> Saul,
>
> My reading of your statement "There are, indeed, inherent limitations whe=
n using SIP in a web browser, but these limitations are imposed by the WebS=
ocket API, *not* by this draft." is that websocket does not qualify as a tr=
ansport for SIP as it is not possible to meet RFC 3261 requirement.
>

Please, have a look at the response given by Paul in this thread .

http://www.ietf.org/mail-archive/web/sipcore/current/msg04802.html


> NAT is not SIP endpoint issue but it is a network issue. All NAT are not =
considered as part of RFC 3261 design but it is resolved as part of Managin=
g Client-Initiated Connections in the Session Initiation Protocol (SIP) RFC=
 (5626).
>
> Thanks
> Partha
>
> Note: AFAIK, NAT issue in SIP is one of the reason for not selecting SIP =
as part of signaling/transport protocol in RTCWeb.
>
>>-----Original Message-----
>>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>>Sent: Friday, April 20, 2012 1:35 PM
>>To: Ravindran, Parthasarathi
>>Cc: I=F1aki Baz Castillo; sipcore@ietf.org
>>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>>websocket-02]
>>
>>Hi Partha,
>>
>>On Apr 20, 2012, at 2:39 AM, Ravindran, Parthasarathi wrote:
>>
>>> Hi Inaki,
>>>
>>> There are only two options which I could think of forward for this
>>draft
>>>
>>> 1) As I earlier mentioned, Websocket transport works well only in
>>specific SIP topology and not in all topology as it is not capable of
>>supporting UDP/TCP transport. In case the draft is only going to specify
>>transport, then SIP entities support this draft has to have compliance
>>with RFC 3261 (UDP, TCP transport) as well and particularly you have to
>>mention how Javascript based SIP stack will compliance with RFC 3261.
>>>
>>
>>SIP over UDP also doesn't work on all topologies, just think about NAT.
>>IMO this draft should not target at using SIP in web browsers using
>>JavaScript, it should just define a new transport for the SIP protocol,
>>which is exactly what it does. There are, indeed, inherent limitations
>>when using SIP in a web browser, but these limitations are imposed by
>>the WebSocket API, *not* by this draft.
>>
>>Also note that there is no "IETF police", if I would implement this
>>draft for using a SIP client within a browser I would not implement UDP
>>or TCP support whatsoever, because they can't be used *in a browser*.
>>Sue me :-)
>>
>>> 2) Alternative is to specify SIP topology requirement wherein
>>websocket really applies. Include the related header like record-route &
>>path with the detail information where these headers are mandated.
>>>
>>> IMHO, the current draft is not clear enough about SIP over Websocket
>>transport usage.
>>>
>>
>>I fail to understand what is not clear about it. Of course WebSockets
>>are used in browsers nearly 100% of the time, but this draft doesn't
>>restrict itself to that model. You could write a desktop SIP client
>>using WebSocket as a transport, and you could listen for incoming
>>connections since limitations such as those in web browsers wouldn't
>>apply. So, please, can you clarify *what* is it not clear about it?
>>
>>
>>Thanks and regards,
>>
>>--
>>Sa=FAl Ibarra Corretg=E9
>>AG Projects
>>
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <4F8C5EC6.8090202@nostrum.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227EAB@inba-mail01.sonusnet.com> <CA+rAfUPj4h0R_qHKw53SFotAQSe83m3jZyucBPxcXqVxfEqMCw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <4F8ED41A.6060703@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E2281EE@inba-mail01.sonusnet.com> <CALiegfkBxu_RDFoFhwSchAeqni-ZAVY6tT6tjUP810_rNZuMpg@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C428E2FA4@ESESSCMS0356.eemea.ericsson.se> <387F9047F55E8C42850AD6B3A7A03C6C0E228315@inba-mail01.sonusnet.com> <CALiegf=Hm+LEoh7jz-0BGiU5gNzPi9vQmSjJZ0f6yZnNwi_v5w@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E22882A@inba-mail01.sonusnet.com> <436967A4-20DC-42A6-9BA2-578155B55FD9@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C 6C0E2289A3@inba-mail01.sonusnet.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Partha,

On Apr 20, 2012, at 10:58 AM, Ravindran, Parthasarathi wrote:

> Saul,
>=20
> My reading of your statement "There are, indeed, inherent limitations =
when using SIP in a web browser, but these limitations are imposed by =
the WebSocket API, *not* by this draft." is that websocket does not =
qualify as a transport for SIP as it is not possible to meet RFC 3261 =
requirement.
>=20

Your reading is incorrect. The fact that *a device* be it a browser, a =
calculator or a softphone is unable to use a specific transport due to =
environment limitations doesn't render the transport unqualifiable.

Again, this is not the "SIP transport for web browsers" draft.


> NAT is not SIP endpoint issue but it is a network issue. All NAT are =
not considered as part of RFC 3261 design but it is resolved as part of =
Managing Client-Initiated Connections in the Session Initiation Protocol =
(SIP) RFC (5626).=20
>=20

Web browser limitations are not a SIP endpoint issue either.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 - sections 10.1 and 11-end
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Hi Dale;

 Thanks again for your comments.=20

 My comments inline..

On Apr 20, 2012, at 11:33 AM, Worley, Dale R (Dale) wrote:

> Correction of my comments on 10.3:  I proposed removing all of item 6:
>=20
>   6.  Missing entity: If the request clearly has a gap in the hi-entry
>       (e.g. by evaluating via header to the existing request history =
to
>       see if it traversed a domain which doesn't exist in History-Info
>       header field.), the entity adding an hi-entry MUST add an index =
a
>       digit of "0" to the current index prior to adding appropriate
>       index for the action to be taken.  If the index of the last hi-
>       entry in the request received was 1.1.2 and there was a missing
>       hi-entry and the request was being forwarded to the next hop, =
the
>       resulting index will be 1.1.2.0.1.
>=20
> But I should have restricted it to removing the "condition", namely
> "(e.g. by evaluating via header to the existing request history to see
> if it traversed a domain which doesn't exist in History-Info header
> field.)"

 Okay, that is more clear but I still see no harm in leaving it in 10.3.

 The text could reside in 9.2 but I personally think it provides more =
uses=20
by having it in 10.3. =20

>=20
> 10.1.1.  Indicating Privacy
>=20
>   In this case, the
>   intermediary MUST set a Privacy header field to a priv-value of
>   "history" and include the Privacy header field in the =
hi-targeted-to-
>   uri, for each new hi-entry added by the intermediary, as the request
>   is retargeted within the domain for which the SIP entity is
>   responsible.  Note, that the hi-entries that are added to a request
>   from the cache are excluded in this case since the appropriate
>   privacy was set for those hi-entries when they were originally added
>   to a request.  The intermediary MUST NOT include any other priv-
>   values in this Privacy header field.
>=20
> The phrasing here is awkward.  I think this would be better:
>=20
>   In this case, the intermediary MUST construct a Privacy header
>   field with the single priv-value of "history" and include the
>   Privacy header field in the hi-targeted-to-uri, for each new
>   hi-entry created by the intermediary, as the request is retargeted
>   within the domain for which the SIP entity is responsible.
>   (Note that hi-entries that are added to a request
>   from the cache are not modified by this process.)

 Looks good.=20

>=20
> However, I'm uncertain of the phrase "is retargeted within the
> domain".  Does that mean retargeting outside the domain of interest
> cannot have this process applied?  Perhaps we want to leave the
> determination to the intermediary, since it is doing the modification
> on its own behalf.  In that case, we would change the end of the
> sentence to:

 Well, for requests that are retargeted outside the domain, if the=20
intermediary want privacy, it should anonymize the hi-entries=20
before retargeting outside the domain of interest but following=20
modification does leave room for intermediary to make a choice,=20
so I will use your suggestion with the following modification.=20

>=20
>   ... for each new hi-entry created by the intermediary whose
>   hi-targeted-to-uri it wishes to privacy protect.
>=20
>   Finally, the terminator of the request may not want to reveal the
>=20
> I think it would be more standard to say
>=20
>   A UAS may not want to reveal the
>=20
> and replace "terminator" with "UAS" in the rest of the paragraph.

 DONE.

>=20
> 10.1.2.  Applying Privacy
>=20
>   When a request is retargeted to a URI associated with a domain for
>   which the SIP intermediary is not responsible or a response is
>   forwarded to a domain for which the SIP intermediary is not
>   responsible, a Privacy Service at the boundary of the domain applies
>=20
> It would be shorter to say
>=20
>   When a SIP message is forwarded to a domain for which the SIP
>   intermediary is not responsible ...
>=20
> I think that has the same effect.

FIXED.

>=20
>   which a SIP intermediary is responsible, are anonymized by the
>=20
> probably should be "which >>>the<<< SIP intermediary is responsible",
> referring to the intermediary discussed in the first paragraph.
>=20

FIXED.

>   URIs containing a domain of anonymous.invalid (e.g.,
>   anonymous@anonymous.invalid).  If there is a Privacy header field in
>=20
> Strictly, that should be "sip:anonymous@anonymous.invalid" and/or
> "sips:anonymous@anonymous.invalid".  Perhaps it would be better to not
> describe the URI at all but just refer to RFC 3323.  However, the
> algorithm requires that H-I processing can *recognize* entries that
> have already been anonymized, so we have to add a condition:
>=20
>   The Privacy Service MUST change any hi-targeted-to-uris in
>>>> these<<< hi-entries that have not been anonymized (evidenced by
>   their domain not being "anonymous.invalid") to anonymous URIs
>   containing a domain of "anonymous.invalid" as described in RFC 3323
>   section 4.1.

 FIXED.

>=20
> also:
>=20
>   If there is a Privacy header field in the "headers" component of
>   the hi-targeted-to-uri in >>>these<<< hi-entries(?), then the
>   Privacy header field value MUST be removed from the hi-entry.

 I think it is fine as is.=20

>=20
> Similarly in the next paragraph:
>=20
>   If there is not a Privacy header field in the message header of the
>   request or response that is being forwarded, but there is a Privacy
>   header field with a priv-value of "history" in the "headers"
>   component in any of the hi-targeted-uris in the hi-entries
>   associated with the domain for which a SIP intermediary is
>   responsible, then the Privacy Service MUST update those
>   hi-targeted-to-uris >>>as described above.<<<  Any other priv-values
>   in the Privacy header field in the "headers" component of the
>   hi-targeted-to-uris in the hi-entries MUST be ignored.
>>>> [Last sentence deleted, as it is included in the "as above"
>   prescription.]<<<
>=20
> Maybe there is a better way to factor out the "anonymize and remove
> Privacy" operation.
>=20
> 11.  Application Considerations
>=20
>   The SIP entity MUST determine if there are gaps in the indices.
>=20
> This is strictly true, but I don't think it's the sense we want.
> There is also the problem that the two types of "gap" are distinctly
> different in what they reveal.  Maybe something fuzzier like:
>=20
>   The SIP entity MUST be prepared to process effectively messages
>   whose hi-entries show evidence of "gaps", that is, situations that
>   reveal that not all of the forks of the request have been recorded
>   in the hi-entries.
>=20
>   Gaps are possible if the request is forwarded through
>   intermediaries that do not support the History-info header field and
>   are reflected by the existence of multiple hi-entries with an digit
>   of "0" e.g. "1.1.0.1".
>=20
> Replace "digit" with "integer" as discussed before, and change "an",
> or rather leave it as "an" because we're changing "digit" to integer.

FIXED.

>=20
> But "multiple" should be deleted.  In particular, when a UAS receives
> a request whose request-URI does not match the last hi-entry, it adds
> a "gap" hi-entry, but since the UAS does not forward the request
> onward, it does not add any further hi-entries with "0"s.  But in this
> case, the request *has* had a gap in H-I processing.
>=20
>   Gaps are also possible in the case of
>   parallel forking if there is an outstanding request at the time the
>   SIP entity sends a response as described in Section 9.4, in which
>   case the gap will not be visible as the branch responsible for the
>   gap wasn't on the path of the request received.
>=20
> I think this sentence needs to be modified strongly:
>=20
>   Gaps are also possible in the case of
>   parallel forking if there is an outstanding request at the time the
>   SIP entity sends a >>>message<<<.
>=20

FIXED.

> A gap (missing branch) due to an outstanding request can be revealed
> in both requests and responses.  The final part of the sentence I
> can't understand.  I wonder if there was a further sentence which was
> damaged and joined to this previous sentence.
>=20

 We were just trying to describe how there could be an index missing=20
not a "0" but  a branch can start from 2 instead of 1 due to parallel =
forking.=20

 Many Thanks indeed for all of your suggestions and comments.

 Regards
  Shida


> 12.  Application Specific Usage
>=20
> It would be wise to mark this section as non-normative:
>=20
>   Following are possible application-specific usages of History-Info.
>=20
> and we could add:
>=20
>   See also the example call flows in Appendix B.
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Date: Fri, 20 Apr 2012 16:39:28 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: =?ISO-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
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Cc: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On Fri, Apr 20, 2012 at 3:41 PM, Sa=FAl Ibarra Corretg=E9
<saul@ag-projects.com>wrote:

> Partha,
>
> On Apr 20, 2012, at 10:58 AM, Ravindran, Parthasarathi wrote:
>
> > Saul,
> >
> > My reading of your statement "There are, indeed, inherent limitations
> when using SIP in a web browser, but these limitations are imposed by the
> WebSocket API, *not* by this draft." is that websocket does not qualify a=
s
> a transport for SIP as it is not possible to meet RFC 3261 requirement.
> >
>
> Your reading is incorrect. The fact that *a device* be it a browser, a
> calculator or a softphone is unable to use a specific transport due to
> environment limitations doesn't render the transport unqualifiable.
>
> Again, this is not the "SIP transport for web browsers" draft.
>

[ABN] If that is the case, then it is illogical to compare sip over sctp /
 sip over ws  / sip over tcp as equivalent in some of the earlier mail
threads. isn't it ???


>
> > NAT is not SIP endpoint issue but it is a network issue. All NAT are no=
t
> considered as part of RFC 3261 design but it is resolved as part of
> Managing Client-Initiated Connections in the Session Initiation Protocol
> (SIP) RFC (5626).
> >
>
> Web browser limitations are not a SIP endpoint issue either.
>
>
> Regards,
>
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



--=20
Thanks,
Nataraju A.B.

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<br><br><div class=3D"gmail_quote">On Fri, Apr 20, 2012 at 3:41 PM, Sa=FAl =
Ibarra Corretg=E9 <span dir=3D"ltr">&lt;<a href=3D"mailto:saul@ag-projects.=
com">saul@ag-projects.com</a>&gt;</span> wrote:<br><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex">
Partha,<br>
<div class=3D"im"><br>
On Apr 20, 2012, at 10:58 AM, Ravindran, Parthasarathi wrote:<br>
<br>
&gt; Saul,<br>
&gt;<br>
&gt; My reading of your statement &quot;There are, indeed, inherent limitat=
ions when using SIP in a web browser, but these limitations are imposed by =
the WebSocket API, *not* by this draft.&quot; is that websocket does not qu=
alify as a transport for SIP as it is not possible to meet RFC 3261 require=
ment.<br>

&gt;<br>
<br>
</div>Your reading is incorrect. The fact that *a device* be it a browser, =
a calculator or a softphone is unable to use a specific transport due to en=
vironment limitations doesn&#39;t render the transport unqualifiable.<br>

<br>
Again, this is not the &quot;SIP transport for web browsers&quot; draft.<br=
></blockquote><div><br></div><div>[ABN] If that is the case, then it is ill=
ogical to compare sip over sctp / =A0sip over ws =A0/ sip over tcp as equiv=
alent in some of the earlier mail threads. isn&#39;t it ???</div>
<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br>
<br>
&gt; NAT is not SIP endpoint issue but it is a network issue. All NAT are n=
ot considered as part of RFC 3261 design but it is resolved as part of Mana=
ging Client-Initiated Connections in the Session Initiation Protocol (SIP) =
RFC (5626).<br>

&gt;<br>
<br>
</div>Web browser limitations are not a SIP endpoint issue either.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
Regards,<br>
<br>
--<br>
Sa=FAl Ibarra Corretg=E9<br>
AG Projects<br>
<br>
<br>
<br>
_______________________________________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>

--f46d042d059670326804be1a5241--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 20 Apr 2012 13:56:45 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi Partha,

I would like to comment you that there are other technologies that
allow coding applications to be executed in web browsers (apart from
JavaScript), those are Flash, Java, SilverLight...

I'm not an expert in them but I do know that some of them gives the
developer access to the TCP/UDP stack of the host, so you are able to
code a SIP TCP/UDP stack with them (in fact there are SIP TCP/UDP
implementations running as Java applets). Those technologies
(regardless they are proprietary or not) also implement, or will
implement soon, a WebSocket stack so they can connect to WebSocket
servers.

This means that you can build a SIP TCP/UDP/WebSocket client and run
it in a web browser.

And of course, there are WebSocket libraries available for many
languages (C, C++, Java, Ruby, Python, Perl...). Somebody could
implement SIP over WebSocket (along with other transports such a
TCP/UDP) and code a SIP client to run as a desktop application or
smartphone app or whatever.


Unfortunately you insist on "The WebSocket API" [1] which is a W3C
specification that enables WebSocket support in web browsers to be
managed via a JavaScript API. You assume that the "SIP WebSocket
Transport" is for web browsers and, not just it, but also that it is
for implementing a SIP stack in JavaScript language. Please check the
draft again and realize that all the references to web browsers,
JavaScript and "The WebSocket API" are just informative and not
normative.

The draft MUST NOT mention that a web browser + JavaScript "is not SIP
compliant since it cannot run SIP over TCP/UDP". The draft just
defines a SIP transport and does not assume not web browsers scenarios
neither JavaScript usage, and as said above, there are more
technologies for developing web applications.


In the other side, the draft does not define a signaling mechanism for
WebRTC. rtcweb WG specifications define the media transport (audio,
video, data channel) for web based applications but does not mandate a
signaling protocol between browsers and servers, neither it mandates
that such a protocol should be managed via JavaScript or that it
should be carried on top of HTTP or WebSocket (you could implement
your custom signaling protocol via RTMP or TCP by using Flash, Java
applet, SilverLight or whatever web technology, and then make use of
the WebRTC stak of the browser for enabling audio/video in your web
app).


So, summarizing:

a) Our draft does not assume web browsers scenario.

b) The web browser scenario must not assume JavaScript and "The
WebSocket API" only ingredients.

c) The limitation you mean (not SIP TCP/UDP transport) just applies in
"web browsers + JavaScript + The WebSocket API". That is a very
specific scope the draft does NOT assume.



Best regards.



[1] "The WebSocket API": http://dev.w3.org/html5/websockets/


PS: About "mandating" Path stuff in the draft, you were are ready
replied by me in a previous mail. Please consider reading it.







2012/4/20 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> Hi Inaki,
>
> There are only two options which I could think of forward for this draft
>
> 1) As I earlier mentioned, Websocket transport works well only in specifi=
c SIP topology and not in all topology as it is not capable of supporting U=
DP/TCP transport. In case the draft is only going to specify transport, the=
n SIP entities support this draft has to have compliance with RFC 3261 (UDP=
, TCP transport) as well and particularly you have to mention how Javascrip=
t based SIP stack will compliance with RFC 3261.
>
> 2) Alternative is to specify SIP topology requirement wherein websocket r=
eally applies. Include the related header like record-route & path with the=
 detail information where these headers are mandated.
>
> IMHO, the current draft is not clear enough about SIP over Websocket tran=
sport usage.
>
> Thanks
> Partha




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Fri, 20 Apr 2012 14:00:30 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 04/20/2012 07:00 AM, Christer Holmberg wrote:
> Hi,
>
> In my opinion, the issue is not whether one has access to UDP/TCP or not - but the fact that it would be stupid and complex being mandated to implement so many different transports.
>
> So, again, let's make every transport optional, and people will then implement based on customer requirements, network environments, platform characteristics, etc etc etc.

Let's just get this started... it's already relevant since SCTP has been 
defined as a transport for SIP. A UA that wants to solely use SCTP 
should be allowed to do that without being labeled non-compliant with 
RFC 3261.

How would this work? Would this just be a two-page document updating 
that one small section of RFC 3261?

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 04/20/2012 03:58 AM, Ravindran, Parthasarathi wrote:

> My reading of your statement "There are, indeed, inherent limitations when using SIP in a web browser, but these limitations are imposed by the WebSocket API, *not* by this draft." is that websocket does not qualify as a transport for SIP as it is not possible to meet RFC 3261 requirement.

This statement is not supported by any facts. It's completely possible 
and practical to use WebSocket between two *non-web-browser* UAs, if 
there is value in doing so. In fact, I can already see value in 
softphones and hardphones supporting SIP over WebSocket (in addition to, 
or instead of, UDP and plain TCP).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From christer.holmberg@ericsson.com  Fri Apr 20 05:50:25 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 20 Apr 2012 14:50:21 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Hi,

>> In my opinion, the issue is not whether one has access to UDP/TCP or not=
 - but the fact that it would be stupid and complex being mandated to imple=
ment so many different transports.
>>
>> So, again, let's make every transport optional, and people will then imp=
lement based on customer requirements, network environments, platform chara=
cteristics, etc etc etc.
>
> Let's just get this started... it's already relevant since SCTP has been =
defined as a transport for SIP. A UA that wants to solely use SCTP should b=
e allowed to do that without being labeled non-compliant with RFC 3261.
>
> How would this work? Would this just be a two-page document updating that=
 one small section of RFC 3261?

Pretty much, yes :)

Of course, in addition to the actual update there would need to be some jus=
tification text.

Regards,

Christer


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 20 Apr 2012 14:54:55 +0200
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "Kevin P. Fleming" <kpfleming@digium.com>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/20 Christer Holmberg <christer.holmberg@ericsson.com>:
>>> So, again, let's make every transport optional, and people will then im=
plement based on customer requirements, network environments, platform char=
acteristics, etc etc etc.
>>
>> Let's just get this started... it's already relevant since SCTP has been=
 defined as a transport for SIP. A UA that wants to solely use SCTP should =
be allowed to do that without being labeled non-compliant with RFC 3261.
>>
>> How would this work? Would this just be a two-page document updating tha=
t one small section of RFC 3261?
>
> Pretty much, yes :)
>
> Of course, in addition to the actual update there would need to be some j=
ustification text.


Hi, in a previous mail I suggested some points for such a new work.
This future draft could address this topic by covering the following
cases:

- Scenarios in which just *secure* SIP transport is required (so UDP
has no place). This could occur in local networks, providers networks
or whatever.

- Scenarios in which just a single transport can be used due to
network topology (i.e. in Outbound topology in which UA's are behind
different NAT's and connect to the same Edge Proxy using TCP or UDP,
so they cannot receive requests via a different transport than the one
they use to connect to their Outbound proxy).

- Special cases in which it's clear that, given enviroment/plattform
constrains, some implementations are not able to support SIP UDP/TCP
transports.

Basically such a draft could state that "in certain cases (such as
those described above) a SIP entity MAY NOT implement UDP and/or TCP
transports, being this an update to RFC 3261".

IMHO just one page :)


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
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Date: Fri, 20 Apr 2012 15:05:03 +0200
Thread-Topic: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "Kevin P. Fleming" <kpfleming@digium.com>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Date: Fri, 20 Apr 2012 15:11:24 +0200
Message-ID: <CALiegfmzT3Xe=pP1XXcFw_ixMnzVqTS8vP1RnWJ6L5TouCc=xQ@mail.gmail.com>
From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "Kevin P. Fleming" <kpfleming@digium.com>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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Rigth. Then let me add this scenario:

- Scenarios in which large SIP messages are required so UDP is no valid.

;)
El 20/04/2012 15:05, "Christer Holmberg" <christer.holmberg@ericsson.com>
escribi=C3=B3:

> You can use IPSec to get secure SIP traffic with UDP :)
>
> Regards,
>
> Christer
>
> -----Original Message-----
> From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: 20. huhtikuuta 2012 15:55
> To: Christer Holmberg
> Cc: Kevin P. Fleming; sipcore@ietf.org
> Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible i=
n
> Websocket transport? [RE: Call for Adoption:
> draft-ibc-sipcore-sip-websocket-02]
>
> 2012/4/20 Christer Holmberg <christer.holmberg@ericsson.com>:
> >>> So, again, let's make every transport optional, and people will then
> implement based on customer requirements, network environments, platform
> characteristics, etc etc etc.
> >>
> >> Let's just get this started... it's already relevant since SCTP has
> been defined as a transport for SIP. A UA that wants to solely use SCTP
> should be allowed to do that without being labeled non-compliant with RFC
> 3261.
> >>
> >> How would this work? Would this just be a two-page document updating
> that one small section of RFC 3261?
> >
> > Pretty much, yes :)
> >
> > Of course, in addition to the actual update there would need to be some
> justification text.
>
>
> Hi, in a previous mail I suggested some points for such a new work.
> This future draft could address this topic by covering the following
> cases:
>
> - Scenarios in which just *secure* SIP transport is required (so UDP has
> no place). This could occur in local networks, providers networks or
> whatever.
>
> - Scenarios in which just a single transport can be used due to network
> topology (i.e. in Outbound topology in which UA's are behind different
> NAT's and connect to the same Edge Proxy using TCP or UDP, so they cannot
> receive requests via a different transport than the one they use to conne=
ct
> to their Outbound proxy).
>
> - Special cases in which it's clear that, given enviroment/plattform
> constrains, some implementations are not able to support SIP UDP/TCP
> transports.
>
> Basically such a draft could state that "in certain cases (such as those
> described above) a SIP entity MAY NOT implement UDP and/or TCP transports=
,
> being this an update to RFC 3261".
>
> IMHO just one page :)
>
>
> --
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
>

--20cf307ca4ba8ce6b804be1c065f
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<p>Rigth. Then let me add this scenario:</p>
<p>- Scenarios in which large SIP messages are required so UDP is no valid.=
</p>
<p>;)</p>
<div class=3D"gmail_quote">El 20/04/2012 15:05, &quot;Christer Holmberg&quo=
t; &lt;<a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@=
ericsson.com</a>&gt; escribi=C3=B3:<br type=3D"attribution"><blockquote cla=
ss=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pa=
dding-left:1ex">
You can use IPSec to get secure SIP traffic with UDP :)<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
-----Original Message-----<br>
From: I=C3=B1aki Baz Castillo [mailto:<a href=3D"mailto:ibc@aliax.net">ibc@=
aliax.net</a>]<br>
Sent: 20. huhtikuuta 2012 15:55<br>
To: Christer Holmberg<br>
Cc: Kevin P. Fleming; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org<=
/a><br>
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]<br>
<br>
2012/4/20 Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsso=
n.com">christer.holmberg@ericsson.com</a>&gt;:<br>
&gt;&gt;&gt; So, again, let&#39;s make every transport optional, and people=
 will then implement based on customer requirements, network environments, =
platform characteristics, etc etc etc.<br>
&gt;&gt;<br>
&gt;&gt; Let&#39;s just get this started... it&#39;s already relevant since=
 SCTP has been defined as a transport for SIP. A UA that wants to solely us=
e SCTP should be allowed to do that without being labeled non-compliant wit=
h RFC 3261.<br>

&gt;&gt;<br>
&gt;&gt; How would this work? Would this just be a two-page document updati=
ng that one small section of RFC 3261?<br>
&gt;<br>
&gt; Pretty much, yes :)<br>
&gt;<br>
&gt; Of course, in addition to the actual update there would need to be som=
e justification text.<br>
<br>
<br>
Hi, in a previous mail I suggested some points for such a new work.<br>
This future draft could address this topic by covering the following<br>
cases:<br>
<br>
- Scenarios in which just *secure* SIP transport is required (so UDP has no=
 place). This could occur in local networks, providers networks or whatever=
.<br>
<br>
- Scenarios in which just a single transport can be used due to network top=
ology (i.e. in Outbound topology in which UA&#39;s are behind different NAT=
&#39;s and connect to the same Edge Proxy using TCP or UDP, so they cannot =
receive requests via a different transport than the one they use to connect=
 to their Outbound proxy).<br>

<br>
- Special cases in which it&#39;s clear that, given enviroment/plattform co=
nstrains, some implementations are not able to support SIP UDP/TCP transpor=
ts.<br>
<br>
Basically such a draft could state that &quot;in certain cases (such as tho=
se described above) a SIP entity MAY NOT implement UDP and/or TCP transport=
s, being this an update to RFC 3261&quot;.<br>
<br>
IMHO just one page :)<br>
<br>
<br>
--<br>
I=C3=B1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</blockquote></div>

--20cf307ca4ba8ce6b804be1c065f--

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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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This is a going to be a Standards Track RFC, therefore it can "update" 3261=
 with text that relaxes the "UDP and TCP" requirement in circumstances wher=
e it is not possible (e.g. in a browser) or not practical. I don't think yo=
u need a separate draft.

When we were working on 3261 about 10 years ago, we had discussions/debates=
 about whether UDP should be required. IIRC, there were some people advocat=
ing UDP be deprecated altogether. There were also discussions about removin=
g the requirement for UDP at some point in the future. The standards need t=
o evolve with the state of the art.


cheers,
(-:bob




-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Christer Holmberg
Sent: Friday, April 20, 2012 5:02 AM
To: Ravindran, Parthasarathi; Sa=FAl Ibarra Corretg=E9
Cc: I=F1aki Baz Castillo; sipcore@ietf.org
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Why not write a separate draft, which updates 3261 by saying that all trans=
ports are optional... That's how the real world looks, anyway.

Regards,

Christer

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ravindran, Parthasarathi
Sent: 20. huhtikuuta 2012 11:59
To: Sa=FAl Ibarra Corretg=E9
Cc: I=F1aki Baz Castillo; sipcore@ietf.org
Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in =
Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocke=
t-02]

Saul,

My reading of your statement "There are, indeed, inherent limitations when =
using SIP in a web browser, but these limitations are imposed by the WebSoc=
ket API, *not* by this draft." is that websocket does not qualify as a tran=
sport for SIP as it is not possible to meet RFC 3261 requirement.

NAT is not SIP endpoint issue but it is a network issue. All NAT are not co=
nsidered as part of RFC 3261 design but it is resolved as part of Managing =
Client-Initiated Connections in the Session Initiation Protocol (SIP) RFC (=
5626).=20

Thanks
Partha

Note: AFAIK, NAT issue in SIP is one of the reason for not selecting SIP as=
 part of signaling/transport protocol in RTCWeb.

>-----Original Message-----
>From: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]
>Sent: Friday, April 20, 2012 1:35 PM
>To: Ravindran, Parthasarathi
>Cc: I=F1aki Baz Castillo; sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible=20
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-=20
>websocket-02]
>
>Hi Partha,
>
>On Apr 20, 2012, at 2:39 AM, Ravindran, Parthasarathi wrote:
>
>> Hi Inaki,
>>
>> There are only two options which I could think of forward for this
>draft
>>
>> 1) As I earlier mentioned, Websocket transport works well only in
>specific SIP topology and not in all topology as it is not capable of=20
>supporting UDP/TCP transport. In case the draft is only going to=20
>specify transport, then SIP entities support this draft has to have=20
>compliance with RFC 3261 (UDP, TCP transport) as well and particularly=20
>you have to mention how Javascript based SIP stack will compliance with RF=
C 3261.
>>
>
>SIP over UDP also doesn't work on all topologies, just think about NAT.
>IMO this draft should not target at using SIP in web browsers using=20
>JavaScript, it should just define a new transport for the SIP protocol,=20
>which is exactly what it does. There are, indeed, inherent limitations=20
>when using SIP in a web browser, but these limitations are imposed by=20
>the WebSocket API, *not* by this draft.
>
>Also note that there is no "IETF police", if I would implement this=20
>draft for using a SIP client within a browser I would not implement UDP=20
>or TCP support whatsoever, because they can't be used *in a browser*.
>Sue me :-)
>
>> 2) Alternative is to specify SIP topology requirement wherein
>websocket really applies. Include the related header like record-route=20
>& path with the detail information where these headers are mandated.
>>
>> IMHO, the current draft is not clear enough about SIP over Websocket
>transport usage.
>>
>
>I fail to understand what is not clear about it. Of course WebSockets=20
>are used in browsers nearly 100% of the time, but this draft doesn't=20
>restrict itself to that model. You could write a desktop SIP client=20
>using WebSocket as a transport, and you could listen for incoming=20
>connections since limitations such as those in web browsers wouldn't=20
>apply. So, please, can you clarify *what* is it not clear about it?
>
>
>Thanks and regards,
>
>--
>Sa=FAl Ibarra Corretg=E9
>AG Projects
>
>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 20 Apr 2012 16:03:36 +0200
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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/20 Bob Penfield <BPenfield@acmepacket.com>:
> This is a going to be a Standards Track RFC, therefore it can "update" 32=
61 with text that relaxes the "UDP and TCP" requirement in circumstances wh=
ere it is not possible (e.g. in a browser) or not practical. I don't think =
you need a separate draft.

Hi Bob. Taking into account that new SIP transports could appear in
the future, would not make more sense to write such an statement in a
separate draft? Just wondering. As Paul said, otherwise each new SIP
transport specification should include the same update to RFC 3261.

Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From pkyzivat@alum.mit.edu  Fri Apr 20 08:03:51 2012
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Subject: [sipcore] Should we change mandatory-to-implement transports?
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I've been looking for a suitable place to jump back into this thread.
Maybe this is as good a place as any. I've changed the subject to 
reflect the current focus of the discussion.

We have two ways forward on this - the narrow way, and the broad way.

The narrow way it to craft a specific exemption for implementers of the 
websocket transport - that they aren't required to implement UDP and 
TCP. Such an exemption could be included in the websocket transport draft.

The broad way is to do a standalone update to 3261 that removes/relaxes 
the requirement to implement UDP and TCP transports.

The narrow way will be easier to achieve. Its impact is narrower and 
easier to understand. So there is likely to be much less objection to 
it. And of course it doesn't require a separate document.

The broad way will cover many more cases all in one swipe of the pen. 
But that means that it will require much more scrutiny to avoid 
unintended consequences. As a result it will be harder to craft and 
harder to pass. My main concern is that it has to potential to erect new 
barriers to interoperability. The point of having some must-implement 
transports is to ensure some common basis for interop. This has the 
potential to lose that. I am myself uncertain if I would support such a 
change. This would depend on some very careful analysis.

I can't promise that the chairs or the AD will even approve of the broad 
approach.

(I guess there is a third way - to do *nothing* about the 
mandatory-to-implement transport requirement. Leave 
javascript-browser-based sip implementations that can't do UDP or TCP 
non-compliant. IMO that is an irresponsible direction that I don't want 
to consider.)

Given all of that, I would like to have some discussion of which way the 
group would like to go. Please do more that vote - actually make the 
case for why you want to go the way you do.

	Thanks,
	Paul [as co-chair]

On 4/20/12 8:54 AM, IĆ±aki Baz Castillo wrote:
> 2012/4/20 Christer Holmberg<christer.holmberg@ericsson.com>:
>>>> So, again, let's make every transport optional, and people will then implement based on customer requirements, network environments, platform characteristics, etc etc etc.
>>>
>>> Let's just get this started... it's already relevant since SCTP has been defined as a transport for SIP. A UA that wants to solely use SCTP should be allowed to do that without being labeled non-compliant with RFC 3261.
>>>
>>> How would this work? Would this just be a two-page document updating that one small section of RFC 3261?
>>
>> Pretty much, yes :)
>>
>> Of course, in addition to the actual update there would need to be some justification text.
>
>
> Hi, in a previous mail I suggested some points for such a new work.
> This future draft could address this topic by covering the following
> cases:
>
> - Scenarios in which just *secure* SIP transport is required (so UDP
> has no place). This could occur in local networks, providers networks
> or whatever.
>
> - Scenarios in which just a single transport can be used due to
> network topology (i.e. in Outbound topology in which UA's are behind
> different NAT's and connect to the same Edge Proxy using TCP or UDP,
> so they cannot receive requests via a different transport than the one
> they use to connect to their Outbound proxy).
>
> - Special cases in which it's clear that, given enviroment/plattform
> constrains, some implementations are not able to support SIP UDP/TCP
> transports.
>
> Basically such a draft could state that "in certain cases (such as
> those described above) a SIP entity MAY NOT implement UDP and/or TCP
> transports, being this an update to RFC 3261".
>
> IMHO just one page :)
>
>


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 20 Apr 2012 18:49:35 +0200
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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2012/4/20 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> We have two ways forward on this - the narrow way, and the broad way.
>
> The narrow way it to craft a specific exemption for implementers of the
> websocket transport - that they aren't required to implement UDP and TCP.
> Such an exemption could be included in the websocket transport draft.
>
> The broad way is to do a standalone update to 3261 that removes/relaxes t=
he
> requirement to implement UDP and TCP transports.
>
> The narrow way will be easier to achieve. Its impact is narrower and easi=
er
> to understand. So there is likely to be much less objection to it. And of
> course it doesn't require a separate document.
>
> The broad way will cover many more cases all in one swipe of the pen. But
> that means that it will require much more scrutiny to avoid unintended
> consequences. As a result it will be harder to craft and harder to pass. =
My
> main concern is that it has to potential to erect new barriers to
> interoperability. The point of having some must-implement transports is t=
o
> ensure some common basis for interop. This has the potential to lose that=
. I
> am myself uncertain if I would support such a change. This would depend o=
n
> some very careful analysis.
>
> I can't promise that the chairs or the AD will even approve of the broad
> approach.
>
> (I guess there is a third way - to do *nothing* about the
> mandatory-to-implement transport requirement. Leave javascript-browser-ba=
sed
> sip implementations that can't do UDP or TCP non-compliant. IMO that is a=
n
> irresponsible direction that I don't want to consider.)
>
> Given all of that, I would like to have some discussion of which way the
> group would like to go. Please do more that vote - actually make the case
> for why you want to go the way you do.


Hi Paul, thanks for opening this thread. Here my opinion:


IMHO there are cases enough to go for the broad way (a separate
document updating RFC 3261), let me repeat them and include some new
ones:

- Scenarios in which just *secure* SIP transport is required and IPsec
is not available (so UDP has no place). This could occur in local
networks, providers networks or whatever. Note that UDP-DTLS for SIP
is an expired draft.

- Scenarios in which just a single transport can be used due to
network topology (i.e. in Outbound scenarios in which UA's are behind
different NAT's and connect to the same Edge Proxy using TCP or UDP,
so they cannot receive requests via a different transport than the one
they used to connect to their Outbound proxy).

- Special cases in which it's clear that, given enviroment/plattform
constrains, some implementations are not able to support SIP UDP/TCP
transports (i.e. JavaScript SIP stacks in web browsers supporting The
WebSocket API).

- Scenarios in which large SIP messages are required. For example,
when SIMPLE/XCAP presence is used, a NOTIFY with a long resource-list
body would not fit into a UDP datagram.

- Scenarios in which SIP interconnection is "private" or negotiated by
other means (i.e. within a SIP PSTN provider infrastructure in which
SIP over SCTP is used).



*But* given you explanation about how hard would be this specification
to be designed and approved, and given the fact that I don't expect so
many new SIP transports to appear in next years, I vote for the narrow
way (a text in our SIP WebSocket Transport draft updating RFC 3261).
Such a text would be placed within "Updates to RFC 3261 section" and
could state the following (text to be improved):




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

5.2.  Updates to RFC 3261


5.2.4  Relaxing TCP/UDP Implementation Requirement


Section 18 "Transport" in RFC 3261 states that all SIP elements MUST
implement UDP and TCP transports:

   All SIP elements MUST implement UDP and TCP.  SIP elements MAY
   implement other protocols.

It is expected that the one of the use cases for the SIP WebSocket
transport will take place within web applications. JavaScript is one
of the existing alternatives for developing applications that run
within the browser, and "The WebSocket API" provides WebSocket access
for JavaScript applications. However the JavaScript stack in web
browsers does not allow direct access to the UDP and TCP layers so
these SIP stacks cannot implement SIP UDP and TCP transports, and must
rely on the WebSocket connection they have opened to the WebSocket
server for sending and receiving SIP messages.

Therefore this specification relaxes the requirement in section 18 of
RFC 3261 by stating that a SIP WebSocket Client (as defined in section
2.1 "Definitions" of this document) MAY NOT implement UDP and TCP
transports, and instead it MAY rely on the SIP WebSocket Server it
connects to for communicating with other SIP entities implementing
different SIP transports such as UDP and TCP.

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



So being pragmatic, my vote for the narrow way.


Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From pkyzivat@alum.mit.edu  Fri Apr 20 10:26:53 2012
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Date: Fri, 20 Apr 2012 13:26:49 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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IĆ±aki,

Thanks for putting good arguments on the table.
I could nit pick them, but that wouldn't enhance the discussion.

I'd like to pose one more subject for discussion in this thread:

Can the "broad way" be constrained in a way that minimizes abuse - 
failure to interoperate for lack of a common transport when there is no 
compelling reason why a common transport could not have been used.

As that relates to your examples below:

- is *no* communication always preferable to insecure communication?
- how often are large sip messages universally required by a UA,
   rather than being required in certain contexts?

I know cases can be contrived, but does that apply to a piece of 
equipment, or does it apply to the deployment of that equipment in a 
particular context?

My point is to figure out if we can construct a revised requirement that 
provides sufficient flexibility where it is needed to create 
implementations, yet doesn't let implementers off the hook when its just 
easier for them to omit support.

For instance, I think we would probably still want to say that a 
transport supporting large messages must always be supported.

	Thanks,
	Paul

On 4/20/12 12:49 PM, IĆ±aki Baz Castillo wrote:
> 2012/4/20 Paul Kyzivat<pkyzivat@alum.mit.edu>:
>> We have two ways forward on this - the narrow way, and the broad way.
>>
>> The narrow way it to craft a specific exemption for implementers of the
>> websocket transport - that they aren't required to implement UDP and TCP.
>> Such an exemption could be included in the websocket transport draft.
>>
>> The broad way is to do a standalone update to 3261 that removes/relaxes the
>> requirement to implement UDP and TCP transports.
>>
>> The narrow way will be easier to achieve. Its impact is narrower and easier
>> to understand. So there is likely to be much less objection to it. And of
>> course it doesn't require a separate document.
>>
>> The broad way will cover many more cases all in one swipe of the pen. But
>> that means that it will require much more scrutiny to avoid unintended
>> consequences. As a result it will be harder to craft and harder to pass. My
>> main concern is that it has to potential to erect new barriers to
>> interoperability. The point of having some must-implement transports is to
>> ensure some common basis for interop. This has the potential to lose that. I
>> am myself uncertain if I would support such a change. This would depend on
>> some very careful analysis.
>>
>> I can't promise that the chairs or the AD will even approve of the broad
>> approach.
>>
>> (I guess there is a third way - to do *nothing* about the
>> mandatory-to-implement transport requirement. Leave javascript-browser-based
>> sip implementations that can't do UDP or TCP non-compliant. IMO that is an
>> irresponsible direction that I don't want to consider.)
>>
>> Given all of that, I would like to have some discussion of which way the
>> group would like to go. Please do more that vote - actually make the case
>> for why you want to go the way you do.
>
>
> Hi Paul, thanks for opening this thread. Here my opinion:
>
>
> IMHO there are cases enough to go for the broad way (a separate
> document updating RFC 3261), let me repeat them and include some new
> ones:
>
> - Scenarios in which just *secure* SIP transport is required and IPsec
> is not available (so UDP has no place). This could occur in local
> networks, providers networks or whatever. Note that UDP-DTLS for SIP
> is an expired draft.
>
> - Scenarios in which just a single transport can be used due to
> network topology (i.e. in Outbound scenarios in which UA's are behind
> different NAT's and connect to the same Edge Proxy using TCP or UDP,
> so they cannot receive requests via a different transport than the one
> they used to connect to their Outbound proxy).
>
> - Special cases in which it's clear that, given enviroment/plattform
> constrains, some implementations are not able to support SIP UDP/TCP
> transports (i.e. JavaScript SIP stacks in web browsers supporting The
> WebSocket API).
>
> - Scenarios in which large SIP messages are required. For example,
> when SIMPLE/XCAP presence is used, a NOTIFY with a long resource-list
> body would not fit into a UDP datagram.
>
> - Scenarios in which SIP interconnection is "private" or negotiated by
> other means (i.e. within a SIP PSTN provider infrastructure in which
> SIP over SCTP is used).
>
>
>
> *But* given you explanation about how hard would be this specification
> to be designed and approved, and given the fact that I don't expect so
> many new SIP transports to appear in next years, I vote for the narrow
> way (a text in our SIP WebSocket Transport draft updating RFC 3261).
> Such a text would be placed within "Updates to RFC 3261 section" and
> could state the following (text to be improved):
>
>
>
>
> --------------------------------------------------------------------
>
> 5.2.  Updates to RFC 3261
>
>
> 5.2.4  Relaxing TCP/UDP Implementation Requirement
>
>
> Section 18 "Transport" in RFC 3261 states that all SIP elements MUST
> implement UDP and TCP transports:
>
>     All SIP elements MUST implement UDP and TCP.  SIP elements MAY
>     implement other protocols.
>
> It is expected that the one of the use cases for the SIP WebSocket
> transport will take place within web applications. JavaScript is one
> of the existing alternatives for developing applications that run
> within the browser, and "The WebSocket API" provides WebSocket access
> for JavaScript applications. However the JavaScript stack in web
> browsers does not allow direct access to the UDP and TCP layers so
> these SIP stacks cannot implement SIP UDP and TCP transports, and must
> rely on the WebSocket connection they have opened to the WebSocket
> server for sending and receiving SIP messages.
>
> Therefore this specification relaxes the requirement in section 18 of
> RFC 3261 by stating that a SIP WebSocket Client (as defined in section
> 2.1 "Definitions" of this document) MAY NOT implement UDP and TCP
> transports, and instead it MAY rely on the SIP WebSocket Server it
> connects to for communicating with other SIP entities implementing
> different SIP transports such as UDP and TCP.
>
> --------------------------------------------------------------------
>
>
>
> So being pragmatic, my vote for the narrow way.
>
>
> Thanks a lot.
>
>


From christer.holmberg@ericsson.com  Fri Apr 20 14:08:16 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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Hi,

Eventhough I would give my hum for the broad way, if that is difficult to g=
et accepted there is also a middle way: RFC 3261 mandates support of TCP an=
d UDP, but is updated to explicitly allow specifications for new transports=
 (SCTP, WebSocket etc) to relax that requirement for implementations of tho=
se specific transports.

The fact would still exist, that there are cases (see Inaki's examples - ev=
enthough I could also nit pick on some of them :)  where an implementation =
won't use, and/or cannot be reached using both TCP and UDP.

Regards,

Christer

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Paul=
 Kyzivat [pkyzivat@alum.mit.edu]
Sent: Friday, April 20, 2012 8:26 PM
To: I=F1aki Baz Castillo
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Should we change mandatory-to-implement transports?

I=F1aki,

Thanks for putting good arguments on the table.
I could nit pick them, but that wouldn't enhance the discussion.

I'd like to pose one more subject for discussion in this thread:

Can the "broad way" be constrained in a way that minimizes abuse -
failure to interoperate for lack of a common transport when there is no
compelling reason why a common transport could not have been used.

As that relates to your examples below:

- is *no* communication always preferable to insecure communication?
- how often are large sip messages universally required by a UA,
   rather than being required in certain contexts?

I know cases can be contrived, but does that apply to a piece of
equipment, or does it apply to the deployment of that equipment in a
particular context?

My point is to figure out if we can construct a revised requirement that
provides sufficient flexibility where it is needed to create
implementations, yet doesn't let implementers off the hook when its just
easier for them to omit support.

For instance, I think we would probably still want to say that a
transport supporting large messages must always be supported.

        Thanks,
        Paul

On 4/20/12 12:49 PM, I=F1aki Baz Castillo wrote:
> 2012/4/20 Paul Kyzivat<pkyzivat@alum.mit.edu>:
>> We have two ways forward on this - the narrow way, and the broad way.
>>
>> The narrow way it to craft a specific exemption for implementers of the
>> websocket transport - that they aren't required to implement UDP and TCP=
.
>> Such an exemption could be included in the websocket transport draft.
>>
>> The broad way is to do a standalone update to 3261 that removes/relaxes =
the
>> requirement to implement UDP and TCP transports.
>>
>> The narrow way will be easier to achieve. Its impact is narrower and eas=
ier
>> to understand. So there is likely to be much less objection to it. And o=
f
>> course it doesn't require a separate document.
>>
>> The broad way will cover many more cases all in one swipe of the pen. Bu=
t
>> that means that it will require much more scrutiny to avoid unintended
>> consequences. As a result it will be harder to craft and harder to pass.=
 My
>> main concern is that it has to potential to erect new barriers to
>> interoperability. The point of having some must-implement transports is =
to
>> ensure some common basis for interop. This has the potential to lose tha=
t. I
>> am myself uncertain if I would support such a change. This would depend =
on
>> some very careful analysis.
>>
>> I can't promise that the chairs or the AD will even approve of the broad
>> approach.
>>
>> (I guess there is a third way - to do *nothing* about the
>> mandatory-to-implement transport requirement. Leave javascript-browser-b=
ased
>> sip implementations that can't do UDP or TCP non-compliant. IMO that is =
an
>> irresponsible direction that I don't want to consider.)
>>
>> Given all of that, I would like to have some discussion of which way the
>> group would like to go. Please do more that vote - actually make the cas=
e
>> for why you want to go the way you do.
>
>
> Hi Paul, thanks for opening this thread. Here my opinion:
>
>
> IMHO there are cases enough to go for the broad way (a separate
> document updating RFC 3261), let me repeat them and include some new
> ones:
>
> - Scenarios in which just *secure* SIP transport is required and IPsec
> is not available (so UDP has no place). This could occur in local
> networks, providers networks or whatever. Note that UDP-DTLS for SIP
> is an expired draft.
>
> - Scenarios in which just a single transport can be used due to
> network topology (i.e. in Outbound scenarios in which UA's are behind
> different NAT's and connect to the same Edge Proxy using TCP or UDP,
> so they cannot receive requests via a different transport than the one
> they used to connect to their Outbound proxy).
>
> - Special cases in which it's clear that, given enviroment/plattform
> constrains, some implementations are not able to support SIP UDP/TCP
> transports (i.e. JavaScript SIP stacks in web browsers supporting The
> WebSocket API).
>
> - Scenarios in which large SIP messages are required. For example,
> when SIMPLE/XCAP presence is used, a NOTIFY with a long resource-list
> body would not fit into a UDP datagram.
>
> - Scenarios in which SIP interconnection is "private" or negotiated by
> other means (i.e. within a SIP PSTN provider infrastructure in which
> SIP over SCTP is used).
>
>
>
> *But* given you explanation about how hard would be this specification
> to be designed and approved, and given the fact that I don't expect so
> many new SIP transports to appear in next years, I vote for the narrow
> way (a text in our SIP WebSocket Transport draft updating RFC 3261).
> Such a text would be placed within "Updates to RFC 3261 section" and
> could state the following (text to be improved):
>
>
>
>
> --------------------------------------------------------------------
>
> 5.2.  Updates to RFC 3261
>
>
> 5.2.4  Relaxing TCP/UDP Implementation Requirement
>
>
> Section 18 "Transport" in RFC 3261 states that all SIP elements MUST
> implement UDP and TCP transports:
>
>     All SIP elements MUST implement UDP and TCP.  SIP elements MAY
>     implement other protocols.
>
> It is expected that the one of the use cases for the SIP WebSocket
> transport will take place within web applications. JavaScript is one
> of the existing alternatives for developing applications that run
> within the browser, and "The WebSocket API" provides WebSocket access
> for JavaScript applications. However the JavaScript stack in web
> browsers does not allow direct access to the UDP and TCP layers so
> these SIP stacks cannot implement SIP UDP and TCP transports, and must
> rely on the WebSocket connection they have opened to the WebSocket
> server for sending and receiving SIP messages.
>
> Therefore this specification relaxes the requirement in section 18 of
> RFC 3261 by stating that a SIP WebSocket Client (as defined in section
> 2.1 "Definitions" of this document) MAY NOT implement UDP and TCP
> transports, and instead it MAY rely on the SIP WebSocket Server it
> connects to for communicating with other SIP entities implementing
> different SIP transports such as UDP and TCP.
>
> --------------------------------------------------------------------
>
>
>
> So being pragmatic, my vote for the narrow way.
>
>
> Thanks a lot.
>
>

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From pravindran@sonusnet.com  Sat Apr 21 19:55:57 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Websocket is a new transport or new URI?
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Date: Sun, 22 Apr 2012 02:55:49 +0000
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F5D@inba-mail01.sonusnet.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227FFD@inba-mail01.sonusnet.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com> <FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228098@inba-mail01.sonusnet.com> <CALiegf=q1HE4bn1dgDz7RqKFe3NCDxr3vOWtR5DTevWAo0rK6A@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C42DF5E46@ESESSCMS0356.eemea.ericsson.se> <4F8ECB82.7070805@ericsson.com>
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Subject: [sipcore] Websocket is a new transport or new URI?
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Sal & others,

After reading lot of mails and thinking further, I agree with you that Webs=
ocket is an application protocol that is used as an transport for SIP. TLS =
is one another application protocol on top of TCP which was considered as t=
ransport (transport=3Dtls) earlier and deprecated as part of RFC 3261 in Se=
c 26.2.2. TLS and Websocket has some unique behavior like hop-by-hop protoc=
ol rather than end-to-end.=20

TLS is adopted within SIP using new URI namely SIPS with transport=3Dtcp. I=
n the same way, I'm proposing SIPWS which implies websocket over TCP, SIPWS=
S which implies websocket over TLS over TCP. Please let me know your opinio=
n on the same.=20

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Salvatore Loreto
>Sent: Wednesday, April 18, 2012 7:41 PM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>On 4/18/12 4:40 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I may have misunderstood something in the conversation, but my
>suggestion would be to consider *NOT* mandating SIP WebSocket clients to
>implement SIP UDP and TCP.
>>
>> Especially if I write a JavaScript SIP browser app, I very likely
>would want to support only WebSocket.
>>
>> Also, I don't think we can directly compare with SCTP, since the
>typical use-case for WebSocket is browser apps (eventhough a native app
>can of course also use WS).
>
>I think that in general we can not compare SCTP to WebSocket at all
>because the first is a pure Transport Protocol, the latter is an
>application protocol that is used as transport.
>
> From a pure WebSocket prospective, what you are going to define is SIP
>as a websocket sub-protocol that is transported on top of.
>
>The important thing to standardize from a SIPCore prospective is the
>interaction/translations that happens between the WebSocket server and
>the SIP proxy that are two different logical entities.
>
>my two cents
>Sal
>
>--
>Salvatore Loreto, PhD
>www.sloreto.com
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

From pravindran@sonusnet.com  Sat Apr 21 20:01:37 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Should we change mandatory-to-implement transports?
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Date: Sun, 22 Apr 2012 03:01:28 +0000
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <4F8ED41A.6060703@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E2281EE@inba-mail01.sonusnet.com> <CALiegfkBxu_RDFoFhwSchAeqni-ZAVY6tT6tjUP810_rNZuMpg@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C428E2FA4@ESESSCMS0356.eemea.ericsson.se> <387F9047F55E8C42850AD6B3A7A03C6C0E228315@inba-mail01.sonusnet.com> <CALiegf=Hm+LEoh7jz-0BGiU5gNzPi9vQmSjJZ0f6yZnNwi_v5w@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E22882A@inba-mail01.sonusnet.com> <CALiegfkJP=aKgvEF3OWYvU+A5V+Qo2tr5NqqU+KybW4eC_s0=w@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C42E5A5A7@ESESSCMS0356.eemea.ericsson.se> <4F915A2C.1080506@digium.com> <7F2072F1E0DE894DA4B517B93C6A05852C42E5A609@ESESSCMS0356.eemea.ericsson.se> <CALiegfngJ2gDFEE4b0UJWc-f1oys41UTZTDxnqfnZM4z=ahuQg@mail.gmail.com> <4F917AD4.2040500@alum.mit.edu>
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From pravindran@sonusnet.com  Sat Apr 21 20:33:45 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Salvatore Loreto <salvatore.loreto@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Interaction between Websocket server & SIP entity
Thread-Index: AQHNIDf9m3t7ssNkiEKLUwiKn7kQsQ==
Date: Sun, 22 Apr 2012 03:27:52 +0000
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Subject: [sipcore] Interaction between Websocket server & SIP entity
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Hi all,

<snip from Sal>

>The important thing to standardize from a SIPCore prospective is the
>interaction/translations that happens between the WebSocket server and
>the SIP proxy that are two different logical entities.

</Snip from Sal>

Websocket server is capable of performing real time communication using RTC=
Web Specification without using SIP. Websocket server will reuse SDP and fo=
llowing the mechanism mentioned in JSEP (Updated RFC 3264 mechanism) in som=
e form. =20

I'm of the same opinion as Sal that interworking functionality between webs=
ocket server which is capable of performing real time communication & SIP e=
ntities is the important part of standardization.  draft-ibc-sipcore-sip-we=
bsocket-02 does not address how JSEP to SIP interaction occurs as it focus =
on the transport mechanism. I'll write new draft about the interworking bet=
ween SIP and RTCWeb entity. Please let me other opinion on the same.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Salvatore Loreto
>Sent: Wednesday, April 18, 2012 7:41 PM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>websocket-02]
>
>On 4/18/12 4:40 PM, Christer Holmberg wrote:
>> Hi,
>>
>> I may have misunderstood something in the conversation, but my
>suggestion would be to consider *NOT* mandating SIP WebSocket clients to
>implement SIP UDP and TCP.
>>
>> Especially if I write a JavaScript SIP browser app, I very likely
>would want to support only WebSocket.
>>
>> Also, I don't think we can directly compare with SCTP, since the
>typical use-case for WebSocket is browser apps (eventhough a native app
>can of course also use WS).
>
>I think that in general we can not compare SCTP to WebSocket at all
>because the first is a pure Transport Protocol, the latter is an
>application protocol that is used as transport.
>
> From a pure WebSocket prospective, what you are going to define is SIP
>as a websocket sub-protocol that is transported on top of.
>
>The important thing to standardize from a SIPCore prospective is the
>interaction/translations that happens between the WebSocket server and
>the SIP proxy that are two different logical entities.
>
>my two cents
>Sal
>
>--
>Salvatore Loreto, PhD
>www.sloreto.com
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 22 Apr 2012 15:20:47 +0200
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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2012/4/20 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> I=C3=B1aki,
>
> Thanks for putting good arguments on the table.
> I could nit pick them, but that wouldn't enhance the discussion.

Hi Paul, let me some comments inline:



> I'd like to pose one more subject for discussion in this thread:
>
> Can the "broad way" be constrained in a way that minimizes abuse - failur=
e
> to interoperate for lack of a common transport when there is no compellin=
g
> reason why a common transport could not have been used.
>
> As that relates to your examples below:
>
> - is *no* communication always preferable to insecure communication?

I could imagine a "like-Skype" service using SIP protocol and
providing both the server infrastructure and the SIP client (i.e. a
SIP softphone). Such a provider could be interested in secure
communications so its SIP client would just implement TLS-TCP
transport for signaling and SRTP (with DTLS or SDES) for media. The
provider does not want and does not accept that their SIP clients are
reachable via UDP, neither it accepts that their SIP clients
communicate with SIP entities other than their "official" SIP
proxies/servers.



> - how often are large sip messages universally required by a UA,
> =C2=A0rather than being required in certain contexts?

A SIP UA typically just registers using a single transport and, in
common scenarios, it is just reachable through the proxy it is
connected to (i.e. Outbound scenarios). Therefore the UA would prefer
to use TCP rather than UDP and then allow big SIP messages (in case
they are needed, as for example when receiving a NOTIFY with a
resource-list XML body).

If a vendor or provider offers such a service, it might consider a
waste of time to implement UDP transport.


> I know cases can be contrived, but does that apply to a piece of equipmen=
t,
> or does it apply to the deployment of that equipment in a particular
> context?

IMHO it's more related to the network context and security requirements.

UDP is not valid for all the SIP scenarios (it is not valid for big
messages and it cannot provide security). If a client wants
"security", why should its equipment support and listen into UDP
transport?



Also let me some questions coming to my mind:

1) If a SIP phone implements both UDP and TCP, but allows the user to
disable UDP (so it no longer listens into a UDP socket), is that phone
"violating" RFC 3261 or not?

2) If a SIP domain announces DNS NAPTR records just including SCTP
transport or just SCTP and TCP, is it "violating" RFC 3261? Note that
RFC 3263 states that, when NAPTR records are present, those are the
options to reach the server.




> My point is to figure out if we can construct a revised requirement that
> provides sufficient flexibility where it is needed to create
> implementations, yet doesn't let implementers off the hook when its just
> easier for them to omit support.

I understand your point in favour of interoperability. Maybe the
following usecase seems a bit ironic:

- A scenario in which *direct* interoperability with other SIP
transports won't take place (i.e. a SIP stack in JavaScript using
WebSocket transport, and whose SIP outbound proxy provides
interoperability with other SIP transports when required).



> For instance, I think we would probably still want to say that a transpor=
t
> supporting large messages must always be supported.

I agree.


Said that, I also consider that such a new document requires lot of
work and analysis (as you predicted). Otherwise it could become a
risk. So for now I would vote for the narrow way. IMHO nothing prevent
the broad way to take place latter.


Best regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From pkyzivat@alum.mit.edu  Sun Apr 22 12:49:04 2012
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Partha,

IMO the main thing we learned about sips is that it was a bad idea.
I don't support making that mistake again.

	Thanks,
	Paul

On 4/21/12 10:55 PM, Ravindran, Parthasarathi wrote:
> Sal&  others,
>
> After reading lot of mails and thinking further, I agree with you that Websocket is an application protocol that is used as an transport for SIP. TLS is one another application protocol on top of TCP which was considered as transport (transport=tls) earlier and deprecated as part of RFC 3261 in Sec 26.2.2. TLS and Websocket has some unique behavior like hop-by-hop protocol rather than end-to-end.
>
> TLS is adopted within SIP using new URI namely SIPS with transport=tcp. In the same way, I'm proposing SIPWS which implies websocket over TCP, SIPWSS which implies websocket over TLS over TCP. Please let me know your opinion on the same.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Salvatore Loreto
>> Sent: Wednesday, April 18, 2012 7:41 PM
>> To: sipcore@ietf.org
>> Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible
>> in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-
>> websocket-02]
>>
>> On 4/18/12 4:40 PM, Christer Holmberg wrote:
>>> Hi,
>>>
>>> I may have misunderstood something in the conversation, but my
>> suggestion would be to consider *NOT* mandating SIP WebSocket clients to
>> implement SIP UDP and TCP.
>>>
>>> Especially if I write a JavaScript SIP browser app, I very likely
>> would want to support only WebSocket.
>>>
>>> Also, I don't think we can directly compare with SCTP, since the
>> typical use-case for WebSocket is browser apps (eventhough a native app
>> can of course also use WS).
>>
>> I think that in general we can not compare SCTP to WebSocket at all
>> because the first is a pure Transport Protocol, the latter is an
>> application protocol that is used as transport.
>>
>>  From a pure WebSocket prospective, what you are going to define is SIP
>> as a websocket sub-protocol that is transported on top of.
>>
>> The important thing to standardize from a SIPCore prospective is the
>> interaction/translations that happens between the WebSocket server and
>> the SIP proxy that are two different logical entities.
>>
>> my two cents
>> Sal
>>
>> --
>> Salvatore Loreto, PhD
>> www.sloreto.com
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/22/12 9:20 AM, IĆ±aki Baz Castillo wrote:
> 2012/4/20 Paul Kyzivat<pkyzivat@alum.mit.edu>:
>> IĆ±aki,
>>
>> Thanks for putting good arguments on the table.
>> I could nit pick them, but that wouldn't enhance the discussion.
>
> Hi Paul, let me some comments inline:
>
>
>
>> I'd like to pose one more subject for discussion in this thread:
>>
>> Can the "broad way" be constrained in a way that minimizes abuse - failure
>> to interoperate for lack of a common transport when there is no compelling
>> reason why a common transport could not have been used.
>>
>> As that relates to your examples below:
>>
>> - is *no* communication always preferable to insecure communication?
>
> I could imagine a "like-Skype" service using SIP protocol and
> providing both the server infrastructure and the SIP client (i.e. a
> SIP softphone). Such a provider could be interested in secure
> communications so its SIP client would just implement TLS-TCP
> transport for signaling and SRTP (with DTLS or SDES) for media. The
> provider does not want and does not accept that their SIP clients are
> reachable via UDP, neither it accepts that their SIP clients
> communicate with SIP entities other than their "official" SIP
> proxies/servers.

A service such as you describe, where all the pieces are provided by a 
single provider, is a proprietary solution, even if it is built using 
public standards.

The more interesting case is when the service is more open, allowing 
differing clients, and maybe differing servers as well. That allows 
tradeoffs to be made.

For instance, maybe somebody wants to make a client for your service 
that can be used by prisoners in a prison. And then maybe there is a law 
forbidding encryption from being used. (And don't get all over me about 
the details of the example - it is just the first one that came to mind.)

>> - how often are large sip messages universally required by a UA,
>>   rather than being required in certain contexts?
>
> A SIP UA typically just registers using a single transport and, in
> common scenarios, it is just reachable through the proxy it is
> connected to (i.e. Outbound scenarios). Therefore the UA would prefer
> to use TCP rather than UDP and then allow big SIP messages (in case
> they are needed, as for example when receiving a NOTIFY with a
> resource-list XML body).

But the UA *may* register with multiple addresses. Or it can register 
with an fqdn that has entries for multiple transports.

But even if it only registers with one, which one to use could be 
decided at deployment time rather than at implementation time.

> If a vendor or provider offers such a service, it might consider a
> waste of time to implement UDP transport.

I am certainly willing to have a debate of the pros and cons of this. 
Making support of UDP optional might help to phase it out.

>> I know cases can be contrived, but does that apply to a piece of equipment,
>> or does it apply to the deployment of that equipment in a particular
>> context?
>
> IMHO it's more related to the network context and security requirements.
>
> UDP is not valid for all the SIP scenarios (it is not valid for big
> messages and it cannot provide security). If a client wants
> "security", why should its equipment support and listen into UDP
> transport?
>
>
>
> Also let me some questions coming to my mind:
>
> 1) If a SIP phone implements both UDP and TCP, but allows the user to
> disable UDP (so it no longer listens into a UDP socket), is that phone
> "violating" RFC 3261 or not?

This is a valid question to ask - 3261 isn't clear on what it means by 
"implement".

IMO its currently mandatory to implement but not mandatory to use, so 
its valid to allow disabling it.

> 2) If a SIP domain announces DNS NAPTR records just including SCTP
> transport or just SCTP and TCP, is it "violating" RFC 3261? Note that
> RFC 3263 states that, when NAPTR records are present, those are the
> options to reach the server.

Again I am inclined to think mandatory to implement, optional to use. So 
as long as the NAPTR config is a deployment-time decision its ok.

>> My point is to figure out if we can construct a revised requirement that
>> provides sufficient flexibility where it is needed to create
>> implementations, yet doesn't let implementers off the hook when its just
>> easier for them to omit support.
>
> I understand your point in favour of interoperability. Maybe the
> following usecase seems a bit ironic:
>
> - A scenario in which *direct* interoperability with other SIP
> transports won't take place (i.e. a SIP stack in JavaScript using
> WebSocket transport, and whose SIP outbound proxy provides
> interoperability with other SIP transports when required).

ISTM that this is a good example of why relaxing the requirements in a 
careful way could still allow the sorts of interop we need. E.g. relax 
the requirements on UAs that connect using outbound.

>> For instance, I think we would probably still want to say that a transport
>> supporting large messages must always be supported.
>
> I agree.
>
>
> Said that, I also consider that such a new document requires lot of
> work and analysis (as you predicted). Otherwise it could become a
> risk. So for now I would vote for the narrow way. IMHO nothing prevent
> the broad way to take place latter.

I'm generally in favor of general approaches to things rather than 
narrow ones. But I can also see why people might not want to expend the 
time and effort to sort out all the issues of the general issue, and so 
do the narrow way for now. And certainly doing that doesn't preclude 
doing something broader later.

	Thanks,
	Paul

From nataraju.sip@gmail.com  Sun Apr 22 21:58:44 2012
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Subject: [sipcore] comments on draft-ietf-sipcore-rfc3265bis-08.txt
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All,

Some logical comments on the draft-ietf-sipcore-rfc3265bis-08.txt.

-----------------------------------------------------
5.3.2.  State Deltas

CurrentText:

   If a NOTIFY request arrives that has a version number that is
   incremented by more than one, the subscriber knows that a state delta
   has been missed; it ignores the NOTIFY request containing the state
   delta (except for the version number, which it retains to detect
   message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
   request containing a complete state snapshot.

Comment:
   From this text, it infers that if any state delta lost, then subscriber
   sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
request inturn lead
   to fetching the complete state information.

   It may be a good diea to differentiate the re-SUBSCRIBE triggered by
different scenarios.
   a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE duration.
In this
      case it is not necessary to fetch the complete state information
   b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec 5.3.2.
      VERSION number in received NOTIFY is increased by greater than ONE).
      In this case subscriber wants to fetch the complete state information.

   If we don't differentiate these two cases, then natural re-SUBSCRIBE
request
   unnecessarily fetch the complete state information, which may not be
necessary / expected.
   But uses higher network bandwidth to fetch the complete state
information.
-----------------------------------------------------
-----------------------------------------------------
4.4.1.  Dialog Creation and Termination

CurrentText:

      A subscriber may send a SUBSCRIBE request with an "Expires" header
      field of 0 in order to trigger the sending of such a NOTIFY
      request; however, for the purposes of subscription and dialog
      lifetime, the subscription is not considered terminated until the
      NOTIFY transaction with a "Subscription-State" of "terminated"
      completes.

Comment:
We may have to mention how long the dialog must be kept alive, before
being cleaned up. Should we delete the dialog immediately after 200-SUB
 or wait till NOTIFY with "Subscription-State"="terminated". If we always
wait till Last Notify for dialog termination, then 200-SUB don't have any
 impact on the dialog termination. Atleast some normative text is of
necessity, for better understanding of the topic.
-----------------------------------------------------
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:

   SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
   in [RFC3261]).  This expires value indicates the duration of the
   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions on a periodic basis using a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].

Comment:
We can add a normative text saying, the refresh time will be
 normally 50% to 75% of subscription duration.

-----------------------------------------------------
-----------------------------------------------------
4.1.2.1.  Requesting a Subscription

CurrentText:
   [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
   an "Accept" header field, but that field does not indicate a media
   type that the notifier is capable of generating in its NOTIFY
   requests, then the proper error response is 406 (Not Acceptable).

COMMENT:
It is better to mention it as a normative text (indented).

-----------------------------------------------------
-----------------------------------------------------
4.1.2.  Creating and Maintaining Subscriptions


CurrentText:

   From the subscriber's perspective, a subscription proceeds according
   to the following state diagram:

Comment:
Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
 received in notify_wait state (because very often this is possible).
If no change in state of FSM, then some normative text shall convey the
same.
        It may be no-operation, but better to mention the same.
-----------------------------------------------------
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:
      In addition to being a request to unsubscribe, a SUBSCRIBE request
      with "Expires" of 0 also causes a fetch of state; see
      Section 4.4.3.

Comment:
     The above text has been mentioned to be normative text. But I feel
this
     shall be written as a regualr text in the draft/rfc. because this is
the
     expected behavior from the notifier.
-----------------------------------------------------
-----------------------------------------------------
3.2.1.

CurrentText:
   This body will contain either the state of the subscribed resource or
   a pointer to such state in the form of a URI (see Section 5.4.13).

ReplacementText/Comment:
   This body **SHALL** contain either the state of the subscribed resource
or
   a pointer to such state in the form of a **URL** (see Section 5.4.13).
-----------------------------------------------------
-----------------------------------------------------
4.2.1.  Subscription Establishment and Maintenance

CurrentText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty performing one
      action without the other.  Additionally, the proper exchange of

ReplacementText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty **in** performing one
      action without the other.  Additionally, the proper exchange of

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

Thanks,
Nataraju A B

---------- Forwarded message ----------
From: Nataraju A.B <nataraju.sip@gmail.com>
Date: Fri, Apr 13, 2012 at 10:53 PM
Subject: draft-ietf-sipcore-rfc3265bis-08.txt - Review comments (Nataraju A
B)
To: Adam Roach <adam@nostrum.com>, sipcore@ietf.org


Adam,

Please find some editorial comments on the draft
"draft draft-ietf-sipcore-rfc3265bis-08.txt"

For simplicity, I have mentioned both the current text and proposed text.

The comments / corrective text can be found with prefix and suffix  --- **

-----------------------------------------------------
1.1.  Overview of Operation
CurrentText:
   The general concept is that entities in the network can subscribe to
   resource or call state for various resources or calls in the network,
   and those entities (or entities acting on their behalf) can send
   notifications when those states change.

ReplacementText:
   The general concept is that entities in the network can subscribe to
   resource or call state **at / of** various resources or calls in the
network,
   and those entities (or entities acting on their behalf) **who** can send
   notifications when those states change.
-----------------------------------------------------

1.1.  Overview of Operation

CurrentText:
   Subscriptions are expired and must be refreshed by subsequent
   SUBSCRIBE requests.

ReplacementText:
   Subscriptions are expired and must be refreshed by subsequent
   SUBSCRIBE requests **before expiration**.
-----------------------------------------------------

2.  Definitions

CurrentText:
   Subscriber:  A subscriber is a user agent which receives NOTIFY
      requests from notifiers; these NOTIFY requests contain information
      about the state of a resource in which the subscriber is
      interested.  Subscribers typically also generate SUBSCRIBE
      requests and send them to notifiers to create subscriptions.

Comment:
After this information, we can add a normative text which indicates
 when a SIP entity might get NOTIFY messages without actually
 sending SUBSCRIBE request. For example, REFER request might end up with
implicit subscription. Also mention any other requests which may lead to
 implicit subscriptions (other than REFER).
-----------------------------------------------------
2.  Definitions

CurrentText:
   Subscription:  A subscription is a set of application state
      associated with a dialog.  This application state includes a
      pointer to the associated dialog, the event package name, and
      possibly an identification token.  Event packages will define
      additional subscription state information.  By definition,
      subscriptions exist in both a subscriber and a notifier.

ReplacementText:
   Subscription:  A subscription is a set of application **states**
      associated with a dialog.  **The** application state includes a
      pointer to the associated dialog, the event package name, and
      possibly an identification token.  Event packages will define
      additional subscription state information.  By definition,
      subscriptions exist in both a subscriber and a notifier.
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:

   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions on a periodic basis using a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].

ReplacementText
3.1.1.  Subscription Duration

   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions **before expiration** on a periodic basis using
a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].
-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:
      In addition to being a request to unsubscribe, a SUBSCRIBE request
      with "Expires" of 0 also causes a fetch of state; see
      Section 4.4.3.

Comment:
     The above text has been mentioned to be normative text. But I feel
this
     shall be written as a regualr text in the draft/rfc. because this is
the
     expected behavior from the notifier.
-----------------------------------------------------
3.1.2.  Identification of Subscribed Events and Event Classes

CurrentText:
   Subscribers MUST include exactly one "Event" header field in
   SUBSCRIBE requests, indicating to which event or class of events they
   are subscribing.  The "Event" header field will contain a token which
   indicates the type of state for which a subscription is being
   requested.  This token will be registered with the IANA and will
   correspond to an event package which further describes the semantics
   of the event or event class.


ReplacementText:
   Subscribers MUST include exactly one "Event" header field in
   SUBSCRIBE requests, indicating to which event or class of events they
   are subscribing.  The "Event" header field **MUST** contain a token which
   indicates the type of state for which a subscription is being
   requested.  This token will be registered with the IANA and will
   correspond to an event package which further describes the semantics
   of the event or event class.
-----------------------------------------------------
3.1.  SUBSCRIBE

CurrentText:
   The SUBSCRIBE method is used to request current state and state
   updates from a remote node.  SUBSCRIBE requests are target refresh


ReplacementText:
   The SUBSCRIBE method is used to request current state and **subsequent**
state
   updates from a remote node.  SUBSCRIBE requests are target refresh


-----------------------------------------------------
3.1.1.  Subscription Duration

CurrentText:

   SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
   in [RFC3261]).  This expires value indicates the duration of the
   subscription.  In order to keep subscriptions effective beyond the
   duration communicated in the "Expires" header field, subscribers need
   to refresh subscriptions on a periodic basis using a new SUBSCRIBE
   request on the same dialog as defined in [RFC3261].

Comment:
We can add a normative text saying, the refresh time will be
 normally 50% to 75% of subscription duration.

-----------------------------------------------------
3.2.1.

CurrentText:
   This body will contain either the state of the subscribed resource or
   a pointer to such state in the form of a URI (see Section 5.4.13).

ReplacementText/Comment:
   This body **SHALL** contain either the state of the subscribed resource
or
   a pointer to such state in the form of a **URL** (see Section 5.4.13).
-----------------------------------------------------
4.1.2.  Creating and Maintaining Subscriptions


CurrentText:

   From the subscriber's perspective, a subscription proceeds according
   to the following state diagram:

Comment:
Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
 received in notify_wait state (because very often this is possible).
 If no change in state of FSM, then some normative text shall convey the
same.
        It may be no-operation, but better to mention the same.
-----------------------------------------------------
4.1.2.1.  Requesting a Subscription

CurrentText:
   When a subscriber wishes to subscribe to a particular state for a
   resource, it forms a SUBSCRIBE request.  If the initial SUBSCRIBE

ReplacementText:
   When a subscriber wishes to subscribe to a particular state **of** a
   resource, it forms a SUBSCRIBE request **according to sec 3.1 and send
it across to notifier**.  If the initial SUBSCRIBE

-----------------------------------------------------
4.1.2.1.  Requesting a Subscription

CurrentText:
   [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
   an "Accept" header field, but that field does not indicate a media
   type that the notifier is capable of generating in its NOTIFY
   requests, then the proper error response is 406 (Not Acceptable).

COMMENT:
It is better to mention it as a normative text (indented).

-----------------------------------------------------
4.1.3.  Receiving and Processing State Information

CurrentText:

   If the "Subscription-State" value is "terminated", the subscriber
   MUST consider the subscription terminated.  The "expires" parameter
   has no semantics for "terminated" -- notifiers SHOULD NOT include an

ReplacementText:

   If the "Subscription-State" value is "terminated", the subscriber
   MUST consider the subscription terminated.  The "expires" parameter
   has no semantics for "terminated" **state**. Notifiers SHOULD NOT
include an
-----------------------------------------------------

CurrentText:
   deactivated:  The subscription has been terminated, but the
      subscriber SHOULD retry immediately with a new subscription.  One

ReplacementText:
   deactivated:  The subscription has been terminated, but the
      subscriber SHOULD retry with a new subscription.  One

   Is It mandatory to resubscribe immediately. if not remove the word
"immediately".
   If it is delayed, what are the consequences ????

-----------------------------------------------------
4.2.1.  Subscription Establishment and Maintenance

CurrentText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty performing one
      action without the other.  Additionally, the proper exchange of

ReplacementText:
      inability to signal the success of a REFER request while signaling
      a problem with the subscription; and difficulty **in** performing one
      action without the other.  Additionally, the proper exchange of

-----------------------------------------------------
4.2.1.4.  Refreshing of Subscriptions

CurrentText:
   If no refresh for a notification address is received before its
   expiration time, the subscription is removed.  When removing a

ReplacementText:
   If no refresh for a notification is received before its
   expiration time, the subscription is removed.  When removing a
 Comment: Word "address" needs to be removed.
-----------------------------------------------------
4.3.  Proxy Behavior

CurrentText:
   Proxies that did not add a Record-Route header field to the initial
   SUBSCRIBE request MUST NOT add a Record-Route header field to any of
   the associated NOTIFY requests.

ReplacementText:
   Proxies that did not add a Record-Route header field to the initial
   SUBSCRIBE request MUST NOT add a Record-Route header field to any of
   the associated NOTIFY requests **or any SUBSCRIBE refresh requests**.
-----------------------------------------------------
4.4.1.  Dialog Creation and Termination

CurrentText:

      A subscriber may send a SUBSCRIBE request with an "Expires" header
      field of 0 in order to trigger the sending of such a NOTIFY
      request; however, for the purposes of subscription and dialog
      lifetime, the subscription is not considered terminated until the
      NOTIFY transaction with a "Subscription-State" of "terminated"
      completes.

Comment:
We may have to mention how long the dialog must be kept alive, before
 being cleaned up. Do we delete the dialog immediately after 200-SUB
or wait till NOTIFY with "Subscription-State"="terminated". If we always
 wait till Last Notify for dialog termination, then 200-SUB don't have any
impact on the dialog termination. Atleast some normative text is of
 necessity, for better understanding of the topic.
-----------------------------------------------------
4.4.3.  Polling Resource State

CurrentText:

   Note that the NOTIFY requests triggered by SUBSCRIBE requests with
   "Expires" header fields of 0 will contain a "Subscription-State"
   value of "terminated", and a "reason" parameter of "timeout".

ReplacementText:
   Note that the NOTIFY requests triggered by SUBSCRIBE requests with
   "Expires" header fields of 0 **shall** contain a "Subscription-State"
   value of "terminated", and a "reason" parameter of "timeout".
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
      for a dialog (e.g., route set handling).  In particular, multiple
      subscriptions within a dialog are expire independently, and
      require independent subscription refreshes.

ReplacementText:
      for a dialog (e.g., route set handling).  In particular, multiple
      subscriptions within a dialog **operates** independently, and
      require independent subscription refreshes.
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
   o  Subscribers MAY include an "id" parameter in SUBSCRIBE request
      "Event" header field to allow differentiation between multiple

ReplacementText:
   o  Subscribers MAY include an "id" parameter in SUBSCRIBE **request's**
      "Event" header field to allow differentiation between multiple
OR
   o  Subscribers MAY include an "id" parameter in **"Event" header field
      of SUBSCRIBE request** to allow differentiation between multiple
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:
   o  When a subscriber refreshes a the subscription timer, the
      SUBSCRIBE request MUST contain the same "Event" header field "id"
      parameter as was present in the SUBSCRIBE request that created the
      subscription.  (Otherwise, the notifier will interpret the

ReplacementText:
   o  When a subscriber refreshes a the subscription timer, the
      SUBSCRIBE request MUST contain the same **"id" parameter in "Event"
      header field, which** was present in the SUBSCRIBE request that
created the
      subscription.  (Otherwise, the notifier will interpret the
-----------------------------------------------------
4.5.2.  Sharing Dialogs

CurrentText:

   o  When a subscription is created in the notifier, it stores any
      "Event" header field "id" parameter as part of the subscription
      information (along with the event package name).

ReplacementText:
   o  When a subscription is created in the notifier, it stores **the**
      **"id" parameter in "Event" header field** as part of the subscription
      information (along with the event package name).
-----------------------------------------------------
4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions

CurrentText:

   Neither SUBSCRIBE nor NOTIFY requests can be canceled.  If a UAS
   receives a CANCEL request that matches a known SUBSCRIBE or NOTIFY
   transaction, it MUST respond to the CANCEL request, but otherwise
   ignore it.  In particular, the CANCEL request MUST NOT affect
   processing of the SUBSCRIBE or NOTIFY request in any way.

ReplacementText/Comment:
What should be the response code for the CANCEL's reply ???
 Preferably 200-OK shall be used.
-----------------------------------------------------
4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions

CurrentText:

   UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY
   transactions.

ReplacementText:
   UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY
   transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY
   or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**
-----------------------------------------------------
5.3.2.  State Deltas

CurrentText:

   In the case that the state information to be conveyed is large, the
   event package may choose to detail a scheme by which NOTIFY requests
   contain state deltas instead of complete state.


ReplacementText:
   In the case that the state information to be conveyed is large, the
   event package may choose a scheme by which NOTIFY requests
   contain state deltas instead of complete state.

comment: deleted words "to detail" in 2nd line.

-----------------------------------------------------
5.3.2.  State Deltas

CurrentText:

   If a NOTIFY request arrives that has a version number that is
   incremented by more than one, the subscriber knows that a state delta
   has been missed; it ignores the NOTIFY request containing the state
   delta (except for the version number, which it retains to detect
   message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
   request containing a complete state snapshot.

Comment:
   From this text, it infers that if any state delta lost, then subscriber
   sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
request inturn lead
   to fetching the complete state information.

   It may be a good diea to differentiate the re-SUBSCRIBE triggered by
different scenarios.
   a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE duration.
In this
      case it is *not *necessary to fetch the complete state information
   b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec 5.3.2.
      VERSION number in received NOTIFY is increased by greater than ONE).
      In this case subscriber wants to fetch the complete state information.

   If we don't differentiate these two cases, then natural re-SUBSCRIBE
request
   unnecessarily fetch the complete state information, which may not be
necessary / expected.
   But uses higher network bandwidth to fetch the complete state
information.
-----------------------------------------------------
5.4.7.  Notifier generation of NOTIFY requests

CurrentText:
   This section may optionally describe the behavior used to process the
   subsequent response.

Comment:
   In this section 5.4.7, the above statement is not clear, what is really
meant to be.

-----------------------------------------------------
5.4.13.  Use of URIs to Retrieve State

CurrentText:

ReplacementText:
   In this context, I feel Use of "URLs" is more appropriate than "URIs".
-----------------------------------------------------


Thanks,
Nataraju A.B.
E-ID : nataraju.ab@gmail.com / nataraju.sip@gmail.com
 +91-98455-95744



-- 
Thanks,
Nataraju A.B.

--f46d0401fde7b61d7b04be517da3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_extra">All,</div><div class=3D"gmail_extra"><br></div><=
div class=3D"gmail_extra">Some logical comments on the draft-ietf-sipcore-r=
fc3265bis-08.txt.</div><div class=3D"gmail_extra"><br></div><div class=3D"g=
mail_extra">

<div class=3D"gmail_extra">------------------------------------------------=
-----</div><div class=3D"gmail_extra">5.3.2. =A0State Deltas</div><div clas=
s=3D"gmail_extra"><br></div><div class=3D"gmail_extra">CurrentText:</div><d=
iv class=3D"gmail_extra">

<br></div><div class=3D"gmail_extra">=A0 =A0If a NOTIFY request arrives tha=
t has a version number that is</div><div class=3D"gmail_extra">=A0 =A0incre=
mented by more than one, the subscriber knows that a state delta</div><div =
class=3D"gmail_extra">

=A0 =A0has been missed; it ignores the NOTIFY request containing the state<=
/div><div class=3D"gmail_extra">=A0 =A0delta (except for the version number=
, which it retains to detect</div><div class=3D"gmail_extra">=A0 =A0message=
 loss), and re-sends a SUBSCRIBE request to force a NOTIFY</div>

<div class=3D"gmail_extra">=A0 =A0request containing a complete state snaps=
hot.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">C=
omment:</div><div class=3D"gmail_extra">=A0 =A0From this text, it infers th=
at if any state delta lost, then subscriber=A0</div>

<div class=3D"gmail_extra">=A0 =A0sends refresh SUBSCRIBE (re-SUBSCRIBE) re=
quest. This re-SUBSCRIBE request inturn lead=A0</div><div class=3D"gmail_ex=
tra">=A0 =A0to fetching the complete state information.=A0</div><div class=
=3D"gmail_extra">

<br></div><div class=3D"gmail_extra">=A0 =A0It may be a good diea to differ=
entiate the re-SUBSCRIBE triggered by different scenarios.</div><div class=
=3D"gmail_extra">=A0 =A0a) natural re-SUBSCRIBE request, which extends the =
SUBSCRIBE duration. In this=A0</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 case it is not necessary to fetch th=
e complete state information</div><div class=3D"gmail_extra">=A0 =A0b) re-S=
UBSCRIBE triggered because of lost NOTIFY request (ref sec 5.3.2.=A0</div><=
div class=3D"gmail_extra">

=A0 =A0 =A0 VERSION number in received NOTIFY is increased by greater than =
ONE).=A0</div><div class=3D"gmail_extra">=A0 =A0 =A0 In this case subscribe=
r wants to fetch the complete state information.</div><div class=3D"gmail_e=
xtra"><br></div>

<div class=3D"gmail_extra">=A0 =A0If we don&#39;t differentiate these two c=
ases, then natural re-SUBSCRIBE request</div><div class=3D"gmail_extra">=A0=
 =A0unnecessarily fetch the complete state information, which may not be ne=
cessary / expected.=A0</div>

<div class=3D"gmail_extra">=A0 =A0But uses higher network bandwidth to fetc=
h the complete state information.</div><div class=3D"gmail_extra">---------=
--------------------------------------------</div><div class=3D"gmail_extra=
">-----------------------------------------------------</div>

<div class=3D"gmail_extra">4.4.1. =A0Dialog Creation and Termination</div><=
div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">CurrentText:=
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">=A0 =
=A0 =A0 A subscriber may send a SUBSCRIBE request with an &quot;Expires&quo=
t; header</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 field of 0 in order to trigger the s=
ending of such a NOTIFY</div><div class=3D"gmail_extra">=A0 =A0 =A0 request=
; however, for the purposes of subscription and dialog</div><div class=3D"g=
mail_extra">=A0 =A0 =A0 lifetime, the subscription is not considered termin=
ated until the</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 NOTIFY transaction with a &quot;Subs=
cription-State&quot; of &quot;terminated&quot;</div><div class=3D"gmail_ext=
ra">=A0 =A0 =A0 completes.</div><div class=3D"gmail_extra"><br></div><div c=
lass=3D"gmail_extra">

Comment:</div><div class=3D"gmail_extra"><span style=3D"white-space:pre-wra=
p">	</span>We may have to mention how long the dialog must be kept alive, b=
efore=A0</div><div class=3D"gmail_extra"><span style=3D"white-space:pre-wra=
p">	</span>being cleaned up. Should we delete the dialog immediately after =
200-SUB=A0</div>

<div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>or =
wait till NOTIFY with &quot;Subscription-State&quot;=3D&quot;terminated&quo=
t;. If we always=A0</div><div class=3D"gmail_extra"><span style=3D"white-sp=
ace:pre-wrap">	</span>wait till Last Notify for dialog termination, then 20=
0-SUB don&#39;t have any=A0</div>

<div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>imp=
act on the dialog termination. Atleast some normative text is of=A0</div><d=
iv class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>neces=
sity, for better understanding of the topic.</div>

<div class=3D"gmail_extra">------------------------------------------------=
-----</div><div class=3D"gmail_extra">-------------------------------------=
----------------</div><div class=3D"gmail_extra">3.1.1. =A0Subscription Dur=
ation</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">CurrentText=
:</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">=A0 =
=A0SUBSCRIBE requests SHOULD contain an &quot;Expires&quot; header field (d=
efined</div>

<div class=3D"gmail_extra">=A0 =A0in [RFC3261]). =A0This expires value indi=
cates the duration of the</div><div class=3D"gmail_extra">=A0 =A0subscripti=
on. =A0In order to keep subscriptions effective beyond the</div><div class=
=3D"gmail_extra">

=A0 =A0duration communicated in the &quot;Expires&quot; header field, subsc=
ribers need</div><div class=3D"gmail_extra">=A0 =A0to refresh subscriptions=
 on a periodic basis using a new SUBSCRIBE</div><div class=3D"gmail_extra">=
=A0 =A0request on the same dialog as defined in [RFC3261].</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Comment:</d=
iv><div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>=
We can add a normative text saying, the refresh time will be</div>
<div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>nor=
mally 50% to 75% of subscription duration.</div><div class=3D"gmail_extra">=
<br></div><div class=3D"gmail_extra">--------------------------------------=
---------------</div>

<div class=3D"gmail_extra">------------------------------------------------=
-----</div><div class=3D"gmail_extra">4.1.2.1. =A0Requesting a Subscription=
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Curre=
ntText:</div>

<div class=3D"gmail_extra">=A0 =A0[RFC3261]. =A0For the sake of clarity: if=
 a SUBSCRIBE request contains</div><div class=3D"gmail_extra">=A0 =A0an &qu=
ot;Accept&quot; header field, but that field does not indicate a media</div=
><div class=3D"gmail_extra">

=A0 =A0type that the notifier is capable of generating in its NOTIFY</div><=
div class=3D"gmail_extra">=A0 =A0requests, then the proper error response i=
s 406 (Not Acceptable).</div><div class=3D"gmail_extra"><br></div><div clas=
s=3D"gmail_extra">

COMMENT:</div><div class=3D"gmail_extra"><span style=3D"white-space:pre-wra=
p">	</span>It is better to mention it as a normative text (indented).</div>=
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">
-----------------------------------------------------</div><div class=3D"gm=
ail_extra">-----------------------------------------------------</div><div =
class=3D"gmail_extra">4.1.2. =A0Creating and Maintaining Subscriptions</div=
>
<div class=3D"gmail_extra">
<br></div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">C=
urrentText:</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_e=
xtra">=A0 =A0From the subscriber&#39;s perspective, a subscription proceeds=
 according</div>

<div class=3D"gmail_extra">=A0 =A0to the following state diagram:</div><div=
 class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Comment:</div><=
div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>Here=
 it is better to mention, what happens to FSM when 200-SUBSCRIBE=A0</div>

<div class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>rec=
eived in notify_wait state (because very often this is possible).</div><div=
 class=3D"gmail_extra"><span style=3D"white-space:pre-wrap">	</span>If no c=
hange in state of FSM, then some normative text shall convey the same.</div=
>

<div class=3D"gmail_extra">=A0 =A0 =A0 =A0 It may be no-operation, but bett=
er to mention the same.</div><div class=3D"gmail_extra">-------------------=
----------------------------------</div><div class=3D"gmail_extra">--------=
---------------------------------------------</div>

<div class=3D"gmail_extra">3.1.1. =A0Subscription Duration</div><div class=
=3D"gmail_extra"><br></div><div class=3D"gmail_extra">CurrentText:</div><di=
v class=3D"gmail_extra">=A0 =A0 =A0 In addition to being a request to unsub=
scribe, a SUBSCRIBE request</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 with &quot;Expires&quot; of 0 also c=
auses a fetch of state; see</div><div class=3D"gmail_extra">=A0 =A0 =A0 Sec=
tion 4.4.3.</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_e=
xtra">Comment:</div>

<div class=3D"gmail_extra">=A0 =A0 =A0The above text has been mentioned to =
be normative text. But I feel this=A0</div><div class=3D"gmail_extra">=A0 =
=A0 =A0shall be written as a regualr text in the draft/rfc. because this is=
 the=A0</div><div class=3D"gmail_extra">

=A0 =A0 =A0expected behavior from the notifier.</div><div class=3D"gmail_ex=
tra">-----------------------------------------------------</div><div class=
=3D"gmail_extra">-----------------------------------------------------</div=
><div class=3D"gmail_extra">

3.2.1.=A0</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_ext=
ra">CurrentText:</div><div class=3D"gmail_extra">=A0 =A0This body will cont=
ain either the state of the subscribed resource or</div><div class=3D"gmail=
_extra">=A0 =A0a pointer to such state in the form of a URI (see Section 5.=
4.13).</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Replacement=
Text/Comment:</div><div class=3D"gmail_extra">=A0 =A0This body **SHALL** co=
ntain either the state of the subscribed resource or</div><div class=3D"gma=
il_extra">

=A0 =A0a pointer to such state in the form of a **URL** (see Section 5.4.13=
).</div><div class=3D"gmail_extra">----------------------------------------=
-------------</div><div class=3D"gmail_extra">-----------------------------=
------------------------</div>

<div class=3D"gmail_extra">4.2.1. =A0Subscription Establishment and Mainten=
ance</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">C=
urrentText:</div><div class=3D"gmail_extra">=A0 =A0 =A0 inability to signal=
 the success of a REFER request while signaling</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 a problem with the subscription; and=
 difficulty performing one</div><div class=3D"gmail_extra">=A0 =A0 =A0 acti=
on without the other. =A0Additionally, the proper exchange of</div><div cla=
ss=3D"gmail_extra">

<br></div><div class=3D"gmail_extra">ReplacementText:</div><div class=3D"gm=
ail_extra">=A0 =A0 =A0 inability to signal the success of a REFER request w=
hile signaling</div><div class=3D"gmail_extra">=A0 =A0 =A0 a problem with t=
he subscription; and difficulty **in** performing one</div>

<div class=3D"gmail_extra">=A0 =A0 =A0 action without the other. =A0Additio=
nally, the proper exchange of</div><div class=3D"gmail_extra"><br></div><di=
v class=3D"gmail_extra">---------------------------------------------------=
--</div><div>

<br></div></div><div class=3D"gmail_extra">Thanks,</div><div class=3D"gmail=
_extra">Nataraju A B<br><br><div class=3D"gmail_quote">---------- Forwarded=
 message ----------<br>From: <b class=3D"gmail_sendername">Nataraju A.B</b>=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"=
_blank">nataraju.sip@gmail.com</a>&gt;</span><br>

Date: Fri, Apr 13, 2012 at 10:53 PM<br>Subject: draft-ietf-sipcore-rfc3265b=
is-08.txt - Review comments (Nataraju A B)<br>To: Adam Roach &lt;<a href=3D=
"mailto:adam@nostrum.com" target=3D"_blank">adam@nostrum.com</a>&gt;, <a hr=
ef=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><br>

<br><br><font color=3D"#000099">Adam,</font><div><font color=3D"#000099"><b=
r></font></div><div><font color=3D"#000099">Please find some editorial comm=
ents on the draft &quot;draft=A0draft-ietf-sipcore-rfc3265bis-08.txt&quot;<=
br clear=3D"all">


</font><div><font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">For simplicity, I have mentioned =
both the current text and proposed text.</font></div><div><font color=3D"#0=
00099"><br></font></div><div><font color=3D"#000099">The comments / correct=
ive text can be found with prefix and suffix =A0--- **</font></div>




<div><font color=3D"#000099"><br></font></div><div><div><font color=3D"#000=
099">-----------------------------------------------------</font></div><div=
><font color=3D"#000099">1.1. =A0Overview of Operation</font></div><div><fo=
nt color=3D"#000099">CurrentText:</font></div>



<div><font color=3D"#000099">=A0 =A0The general concept is that entities in=
 the network can subscribe to</font></div>
<div><font color=3D"#000099">=A0 =A0resource or call state for various reso=
urces or calls in the network,</font></div><div><font color=3D"#000099">=A0=
 =A0and those entities (or entities acting on their behalf) can send</font>=
</div><div>



<font color=3D"#000099">=A0 =A0notifications when those states change.</fon=
t></div><div><font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">ReplacementText:</font></div><div=
><font color=3D"#000099">=A0 =A0The general concept is that entities in the=
 network can subscribe to</font></div><div><font color=3D"#000099">=A0 =A0r=
esource or call state **at / of** various resources or calls in the network=
,</font></div>



<div><font color=3D"#000099">=A0 =A0and those entities (or entities acting =
on their behalf) **who** can send</font></div>
<div><font color=3D"#000099">=A0 =A0notifications when those states change.=
</font></div><div><font color=3D"#000099">---------------------------------=
--------------------</font></div><div><font color=3D"#000099"><br></font></=
div><div>



<font color=3D"#000099">1.1. =A0Overview of Operation</font></div><div><fon=
t color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentTe=
xt:</font></div><div><font color=3D"#000099">=A0 =A0Subscriptions are expir=
ed and must be refreshed by subsequent</font></div>




<div><font color=3D"#000099">=A0 =A0SUBSCRIBE requests.</font></div><div><f=
ont color=3D"#000099"><br></font></div><div><font color=3D"#000099">Replace=
mentText:</font></div><div><font color=3D"#000099">=A0 =A0Subscriptions are=
 expired and must be refreshed by subsequent</font></div>



<div><font color=3D"#000099">=A0 =A0SUBSCRIBE requests **before expiration*=
*.</font></div><div><font color=3D"#000099">-------------------------------=
----------------------</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
2. =A0Definitions</font></div><div><font color=3D"#000099"><br></font></div=
><div><font color=3D"#000099">CurrentText:</font></div><div><font color=3D"=
#000099">=A0 =A0Subscriber: =A0A subscriber is a user agent which receives =
NOTIFY</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 requests from notifiers; these NOT=
IFY requests contain information</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 about the state of a resource in w=
hich the subscriber is</font></div><div><font color=3D"#000099">=A0 =A0 =A0=
 interested. =A0Subscribers typically also generate SUBSCRIBE</font></div><=
div><font color=3D"#000099">=A0 =A0 =A0 requests and send them to notifiers=
 to create subscriptions.</font></div>




<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
Comment:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>After this information, we can add a normative text whi=
ch indicates=A0</font></div>



<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>w=
hen a SIP entity might get NOTIFY messages without actually=A0</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>s=
ending SUBSCRIBE request. For example, REFER request might end up with=A0</=
font></div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap=
">	</span>implicit subscription. Also mention any other requests which may =
lead to=A0</font></div>




<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>i=
mplicit subscriptions (other than REFER).</font></div><div><font color=3D"#=
000099">-----------------------------------------------------</font></div><=
div>



<font color=3D"#000099">2. =A0Definitions</font></div><div><font color=3D"#=
000099"><br></font></div><div><font color=3D"#000099">
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0Subscription: =
=A0A subscription is a set of application state</font></div><div><font colo=
r=3D"#000099">=A0 =A0 =A0 associated with a dialog. =A0This application sta=
te includes a</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 pointer to the associated dialog, =
the event package name, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 possibly an identification token. =
=A0Event packages will define</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 additional subscription state information. =A0By definition,</font>=
</div><div><font color=3D"#000099">=A0 =A0 =A0 subscriptions exist in both =
a subscriber and a notifier.</font></div>




<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0Subscripti=
on: =A0A subscription is a set of application **states**</font></div><div><=
font color=3D"#000099">=A0 =A0 =A0 associated with a dialog. =A0**The** app=
lication state includes a</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 pointer to the associated dialog, =
the event package name, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 possibly an identification token. =
=A0Event packages will define</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 additional subscription state information. =A0By definition,</font>=
</div><div><font color=3D"#000099">=A0 =A0 =A0 subscriptions exist in both =
a subscriber and a notifier.</font></div>




<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">3.1.1. =A0Subscription Dur=
ation</font></div><div><font color=3D"#000099"><br></font></div><div><font =
color=3D"#000099">CurrentText:</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0subscription. =A0In order to keep subscriptions effective beyond the=
</font></div>
<div><font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expi=
res&quot; header field, subscribers need</font></div><div><font color=3D"#0=
00099">=A0 =A0to refresh subscriptions on a periodic basis using a new SUBS=
CRIBE</font></div>



<div><font color=3D"#000099">=A0 =A0request on the same dialog as defined i=
n [RFC3261].</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText</font></div><div><font color=3D"#000099">3.1.1. =A0Subscrip=
tion Duration</font></div><div><font color=3D"#000099"><br></font></div><di=
v><font color=3D"#000099">=A0 =A0subscription. =A0In order to keep subscrip=
tions effective beyond the</font></div>



<div><font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expi=
res&quot; header field, subscribers need</font></div>
<div><font color=3D"#000099">=A0 =A0to refresh subscriptions **before expir=
ation** on a periodic basis using a new SUBSCRIBE</font></div><div><font co=
lor=3D"#000099">=A0 =A0request on the same dialog as defined in [RFC3261].<=
/font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">3.1.1. =A0Subscription Duration</font></div><d=
iv><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Cu=
rrentText:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 In addition=
 to being a request to unsubscribe, a SUBSCRIBE request</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 with &quot;Expires&quot; of 0 also=
 causes a fetch of state; see</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 Section 4.4.3.</font></div><div><f=
ont color=3D"#000099"><br></font></div><div><font color=3D"#000099">Comment=
:</font></div><div><font color=3D"#000099">=A0 =A0 =A0The above text has be=
en mentioned to be normative text. But I feel this=A0</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0shall be written as a regualr text =
in the draft/rfc. because this is the=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0expected behavior from the notifier=
.</font></div><div><font color=3D"#000099">--------------------------------=
---------------------</font></div><div><font color=3D"#000099">3.1.2. =A0Id=
entification of Subscribed Events and Event Classes</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div>
<div><font color=3D"#000099">=A0 =A0Subscribers MUST include exactly one &q=
uot;Event&quot; header field in</font></div><div><font color=3D"#000099">=
=A0 =A0SUBSCRIBE requests, indicating to which event or class of events the=
y</font></div>



<div><font color=3D"#000099">=A0 =A0are subscribing. =A0The &quot;Event&quo=
t; header field will contain a token which</font></div>
<div><font color=3D"#000099">=A0 =A0indicates the type of state for which a=
 subscription is being</font></div><div><font color=3D"#000099">=A0 =A0requ=
ested. =A0This token will be registered with the IANA and will</font></div>=
<div><font color=3D"#000099">=A0 =A0correspond to an event package which fu=
rther describes the semantics</font></div>




<div><font color=3D"#000099">=A0 =A0of the event or event class.</font></di=
v><div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099=
"><br></font></div><div><font color=3D"#000099">ReplacementText:</font></di=
v><div>



<font color=3D"#000099">=A0 =A0Subscribers MUST include exactly one &quot;E=
vent&quot; header field in</font></div><div><font color=3D"#000099">=A0 =A0=
SUBSCRIBE requests, indicating to which event or class of events they</font=
></div>



<div><font color=3D"#000099">=A0 =A0are subscribing. =A0The &quot;Event&quo=
t; header field **MUST** contain a token which</font></div><div><font color=
=3D"#000099">=A0 =A0indicates the type of state for which a subscription is=
 being</font></div>



<div><font color=3D"#000099">=A0 =A0requested. =A0This token will be regist=
ered with the IANA and will</font></div>
<div><font color=3D"#000099">=A0 =A0correspond to an event package which fu=
rther describes the semantics</font></div><div><font color=3D"#000099">=A0 =
=A0of the event or event class.</font></div><div><font color=3D"#000099">--=
---------------------------------------------------</font></div>



<div><font color=3D"#000099">3.1. =A0SUBSCRIBE</font></div><div><font color=
=3D"#000099"><br>
</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0The SUBSCRIBE method is used to request current=
 state and state</font></div><div><font color=3D"#000099">=A0 =A0updates fr=
om a remote node. =A0SUBSCRIBE requests are target refresh</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
<br></font></div><div><font color=3D"#000099">
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0The SUBSCR=
IBE method is used to request current state and **subsequent** state</font>=
</div><div><font color=3D"#000099">=A0 =A0updates from a remote node. =A0SU=
BSCRIBE requests are target refresh</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
<br></font></div>
<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">3.1.1. =A0Subscription Dur=
ation</font></div><div><font color=3D"#000099"><br></font></div><div><font =
color=3D"#000099">CurrentText:</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0SUBSCRIBE requests SHOULD contain an &quot;Expires&quot; header fiel=
d (defined</font></div>
<div><font color=3D"#000099">=A0 =A0in [RFC3261]). =A0This expires value in=
dicates the duration of the</font></div><div><font color=3D"#000099">=A0 =
=A0subscription. =A0In order to keep subscriptions effective beyond the</fo=
nt></div><div>



<font color=3D"#000099">=A0 =A0duration communicated in the &quot;Expires&q=
uot; header field, subscribers need</font></div>
<div><font color=3D"#000099">=A0 =A0to refresh subscriptions on a periodic =
basis using a new SUBSCRIBE</font></div><div><font color=3D"#000099">=A0 =
=A0request on the same dialog as defined in [RFC3261].</font></div><div><fo=
nt color=3D"#000099"><br>



</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>We can add a =
normative text saying, the refresh time will be</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>n=
ormally 50% to 75% of subscription duration.</font></div><div><font color=
=3D"#000099"><br></font></div><div><font color=3D"#000099">----------------=
-------------------------------------</font></div>



<div><font color=3D"#000099">3.2.1.=A0</font></div><div><font color=3D"#000=
099"><br>
</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0This body will contain either the state of the =
subscribed resource or</font></div><div><font color=3D"#000099">=A0 =A0a po=
inter to such state in the form of a URI (see Section 5.4.13).</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText/Comment:</font></div>
<div><font color=3D"#000099">=A0 =A0This body **SHALL** contain either the =
state of the subscribed resource or</font></div><div><font color=3D"#000099=
">=A0 =A0a pointer to such state in the form of a **URL** (see Section 5.4.=
13).</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.1.2. =A0Creating and Maintaining Subscriptio=
ns</font></div><div><font color=3D"#000099"><br></font></div><div><font col=
or=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentText:</=
font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0From the subscriber&#39;s perspective, a subscription proceeds accor=
ding</font></div><div><font color=3D"#000099">=A0 =A0to the following state=
 diagram:</font></div>




<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
Comment:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>Here it is better to mention, what happens to FSM when =
200-SUBSCRIBE=A0</font></div>



<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>r=
eceived in notify_wait state (because very often this is possible).</font><=
/div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>I=
f no change in state of FSM, then some normative text shall convey the same=
.</font></div><div><font color=3D"#000099">=A0 =A0 =A0 =A0 It may be no-ope=
ration, but better to mention the same.</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">4.1.2.1. =A0Requesting a S=
ubscription</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0When a subscri=
ber wishes to subscribe to a particular state for a</font></div><div><font =
color=3D"#000099">=A0 =A0resource, it forms a SUBSCRIBE request. =A0If the =
initial SUBSCRIBE</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div>
<div><font color=3D"#000099">=A0 =A0When a subscriber wishes to subscribe t=
o a particular state **of** a</font></div><div><font color=3D"#000099">=A0 =
=A0resource, it forms a SUBSCRIBE request **according to sec 3.1 and send i=
t across to notifier**. =A0If the initial SUBSCRIBE</font></div>



<div>
<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">-----=
------------------------------------------------</font></div><div><font col=
or=3D"#000099">4.1.2.1. =A0Requesting a Subscription</font></div><div><font=
 color=3D"#000099"><br>



</font></div><div><font color=3D"#000099">CurrentText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0[RFC3261]. =A0For the sake of clarity: if a SUB=
SCRIBE request contains</font></div>
<div><font color=3D"#000099">=A0 =A0an &quot;Accept&quot; header field, but=
 that field does not indicate a media</font></div><div><font color=3D"#0000=
99">=A0 =A0type that the notifier is capable of generating in its NOTIFY</f=
ont></div>



<div><font color=3D"#000099">=A0 =A0requests, then the proper error respons=
e is 406 (Not Acceptable).</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
COMMENT:</font></div><div><font color=3D"#000099"><span style=3D"white-spac=
e:pre-wrap">	</span>It is better to mention it as a normative text (indente=
d).</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
-----------------------------------------------------</font></div>
<div><font color=3D"#000099">4.1.3. =A0Receiving and Processing State Infor=
mation</font></div><div><font color=3D"#000099"><br></font></div><div><font=
 color=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099"><b=
r></font></div>



<div><font color=3D"#000099">=A0 =A0If the &quot;Subscription-State&quot; v=
alue is &quot;terminated&quot;, the subscriber</font></div><div><font color=
=3D"#000099">=A0 =A0MUST consider the subscription terminated. =A0The &quot=
;expires&quot; parameter</font></div>




<div><font color=3D"#000099">=A0 =A0has no semantics for &quot;terminated&q=
uot; -- notifiers SHOULD NOT include an</font></div><div><font color=3D"#00=
0099"><br></font></div><div><font color=3D"#000099">ReplacementText:</font>=
</div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If the &quot;Subscription-State&quot; value is &quot;terminated&quot=
;, the subscriber</font></div>
<div><font color=3D"#000099">=A0 =A0MUST consider the subscription terminat=
ed. =A0The &quot;expires&quot; parameter</font></div><div><font color=3D"#0=
00099">=A0 =A0has no semantics for &quot;terminated&quot; **state**. Notifi=
ers SHOULD NOT include an</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0deactivated: =
=A0The subscription has been terminated, but the</font></div><div><font col=
or=3D"#000099">=A0 =A0 =A0 subscriber SHOULD retry immediately with a new s=
ubscription. =A0One</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div>
<div><font color=3D"#000099">=A0 =A0deactivated: =A0The subscription has be=
en terminated, but the</font></div><div><font color=3D"#000099">=A0 =A0 =A0=
 subscriber SHOULD retry with a new subscription. =A0One</font></div><div><=
font color=3D"#000099"><br>



</font></div><div><font color=3D"#000099">=A0 =A0Is It mandatory to resubsc=
ribe immediately. if not remove the word &quot;immediately&quot;.</font></d=
iv>
<div><font color=3D"#000099">=A0 =A0If it is delayed, what are the conseque=
nces ????</font></div><div><font color=3D"#000099"><br></font></div><div><f=
ont color=3D"#000099">-----------------------------------------------------=
</font></div>



<div><font color=3D"#000099">4.2.1. =A0Subscription Establishment and Maint=
enance</font></div><div><font color=3D"#000099"><br></font></div><div><font=
 color=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099">=
=A0 =A0 =A0 inability to signal the success of a REFER request while signal=
ing</font></div>




<div><font color=3D"#000099">=A0 =A0 =A0 a problem with the subscription; a=
nd difficulty performing one</font></div><div><font color=3D"#000099">=A0 =
=A0 =A0 action without the other. =A0Additionally, the proper exchange of</=
font></div><div>



<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Repla=
cementText:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 inability =
to signal the success of a REFER request while signaling</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 a problem with the subscription; a=
nd difficulty **in** performing one</font></div><div><font color=3D"#000099=
">=A0 =A0 =A0 action without the other. =A0Additionally, the proper exchang=
e of</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
-----------------------------------------------------</font></div>
<div><font color=3D"#000099">4.2.1.4. =A0Refreshing of Subscriptions</font>=
</div><div><font color=3D"#000099"><br></font></div><div><font color=3D"#00=
0099">CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0If no re=
fresh for a notification address is received before its</font></div>



<div><font color=3D"#000099">=A0 =A0expiration time, the subscription is re=
moved. =A0When removing a</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0If no refr=
esh for a notification is received before its</font></div><div><font color=
=3D"#000099">=A0 =A0expiration time, the subscription is removed. =A0When r=
emoving a</font></div>



<div><span style=3D"white-space:pre-wrap"><font color=3D"#000099">	</font><=
/span></div>
<div><font color=3D"#000099">Comment:<span style=3D"white-space:pre-wrap">	=
</span>Word &quot;address&quot; needs to be removed.</font></div><div><font=
 color=3D"#000099">-----------------------------------------------------</f=
ont></div>



<div><font color=3D"#000099">4.3. =A0Proxy Behavior</font></div><div>
<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">Curre=
ntText:</font></div><div><font color=3D"#000099">=A0 =A0Proxies that did no=
t add a Record-Route header field to the initial</font></div><div><font col=
or=3D"#000099">=A0 =A0SUBSCRIBE request MUST NOT add a Record-Route header =
field to any of</font></div>



<div><font color=3D"#000099">=A0 =A0the associated NOTIFY requests.</font><=
/div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
ReplacementText:</font></div><div><font color=3D"#000099">=A0 =A0Proxies th=
at did not add a Record-Route header field to the initial</font></div><div>=
<font color=3D"#000099">=A0 =A0SUBSCRIBE request MUST NOT add a Record-Rout=
e header field to any of</font></div>



<div><font color=3D"#000099">=A0 =A0the associated NOTIFY requests **or any=
 SUBSCRIBE refresh requests**.</font></div>
<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">4.4.1. =A0Dialog Creation =
and Termination</font></div><div><font color=3D"#000099"><br></font></div><=
div>



<font color=3D"#000099">CurrentText:</font></div><div><font color=3D"#00009=
9"><br></font></div><div><font color=3D"#000099">=A0 =A0 =A0 A subscriber m=
ay send a SUBSCRIBE request with an &quot;Expires&quot; header</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 field of 0 in order to trigger the=
 sending of such a NOTIFY</font></div><div><font color=3D"#000099">=A0 =A0 =
=A0 request; however, for the purposes of subscription and dialog</font></d=
iv><div><font color=3D"#000099">=A0 =A0 =A0 lifetime, the subscription is n=
ot considered terminated until the</font></div>




<div><font color=3D"#000099">=A0 =A0 =A0 NOTIFY transaction with a &quot;Su=
bscription-State&quot; of &quot;terminated&quot;</font></div><div><font col=
or=3D"#000099">=A0 =A0 =A0 completes.</font></div><div><font color=3D"#0000=
99"><br></font></div>



<div><font color=3D"#000099">Comment:</font></div><div><font color=3D"#0000=
99"><span style=3D"white-space:pre-wrap">	</span>We may have to mention how=
 long the dialog must be kept alive, before=A0</font></div>
<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>b=
eing cleaned up. Do we delete the dialog immediately after 200-SUB=A0</font=
></div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	<=
/span>or wait till NOTIFY with &quot;Subscription-State&quot;=3D&quot;termi=
nated&quot;. If we always=A0</font></div>




<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>w=
ait till Last Notify for dialog termination, then 200-SUB don&#39;t have an=
y=A0</font></div><div><font color=3D"#000099"><span style=3D"white-space:pr=
e-wrap">	</span>impact on the dialog termination. Atleast some normative te=
xt is of=A0</font></div>




<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>n=
ecessity, for better understanding of the topic.</font></div><div><font col=
or=3D"#000099">-----------------------------------------------------</font>=
</div>



<div><font color=3D"#000099">4.4.3. =A0Polling Resource State</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0Note that the NOTIFY requests triggered by =
SUBSCRIBE requests with</font></div>



<div><font color=3D"#000099">=A0 =A0&quot;Expires&quot; header fields of 0 =
will contain a &quot;Subscription-State&quot;</font></div>
<div><font color=3D"#000099">=A0 =A0value of &quot;terminated&quot;, and a =
&quot;reason&quot; parameter of &quot;timeout&quot;.</font></div><div><font=
 color=3D"#000099"><br></font></div><div><font color=3D"#000099">Replacemen=
tText:</font></div>



<div><font color=3D"#000099">=A0 =A0Note that the NOTIFY requests triggered=
 by SUBSCRIBE requests with</font></div>
<div><font color=3D"#000099">=A0 =A0&quot;Expires&quot; header fields of 0 =
**shall** contain a &quot;Subscription-State&quot;</font></div><div><font c=
olor=3D"#000099">=A0 =A0value of &quot;terminated&quot;, and a &quot;reason=
&quot; parameter of &quot;timeout&quot;.</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentT=
ext:</font></div><div><font color=3D"#000099">=A0 =A0 =A0 for a dialog (e.g=
., route set handling). =A0In particular, multiple</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 subscriptions within a dialog are =
expire independently, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 require independent subscription r=
efreshes.</font></div><div><font color=3D"#000099"><br></font></div><div><f=
ont color=3D"#000099">ReplacementText:</font></div><div><font color=3D"#000=
099">=A0 =A0 =A0 for a dialog (e.g., route set handling). =A0In particular,=
 multiple</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 subscriptions within a dialog **op=
erates** independently, and</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 require independent subscription r=
efreshes.</font></div><div><font color=3D"#000099">------------------------=
-----------------------------</font></div><div><font color=3D"#000099">4.5.=
2. =A0Sharing Dialogs</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">=A0 =A0o =A0Subscribe=
rs MAY include an &quot;id&quot; parameter in SUBSCRIBE request</font></div=
>
<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field to =
allow differentiation between multiple</font></div><div><font color=3D"#000=
099"><br></font></div><div><font color=3D"#000099">ReplacementText:</font><=
/div><div>



<font color=3D"#000099">=A0 =A0o =A0Subscribers MAY include an &quot;id&quo=
t; parameter in SUBSCRIBE **request&#39;s**</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field to =
allow differentiation between multiple</font></div><div><font color=3D"#000=
099"><span style=3D"white-space:pre-wrap">				</span>OR</font></div><div><f=
ont color=3D"#000099">=A0 =A0o =A0Subscribers MAY include an &quot;id&quot;=
 parameter in **&quot;Event&quot; header field</font></div>




<div><font color=3D"#000099">=A0 =A0 =A0 of SUBSCRIBE request** to allow di=
fferentiation between multiple</font></div><div><font color=3D"#000099">---=
--------------------------------------------------</font></div><div><font c=
olor=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099">
=A0 =A0o =A0When a subscriber refreshes a the subscription timer, the</font=
></div><div><font color=3D"#000099">=A0 =A0 =A0 SUBSCRIBE request MUST cont=
ain the same &quot;Event&quot; header field &quot;id&quot;</font></div><div=
><font color=3D"#000099">=A0 =A0 =A0 parameter as was present in the SUBSCR=
IBE request that created the</font></div>




<div><font color=3D"#000099">=A0 =A0 =A0 subscription. =A0(Otherwise, the n=
otifier will interpret the</font></div><div><font color=3D"#000099"><br></f=
ont></div><div><font color=3D"#000099">ReplacementText:</font></div><div><f=
ont color=3D"#000099">=A0 =A0o =A0When a subscriber refreshes a the subscri=
ption timer, the</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 SUBSCRIBE request MUST contain the=
 same **&quot;id&quot; parameter in &quot;Event&quot;=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 header field, which** was present =
in the SUBSCRIBE request that created the</font></div><div><font color=3D"#=
000099">=A0 =A0 =A0 subscription. =A0(Otherwise, the notifier will interpre=
t the</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">4.5.2. =A0Sharing Dialogs</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099">CurrentT=
ext:</font></div><div><font color=3D"#000099"><br></font></div><div><font c=
olor=3D"#000099">=A0 =A0o =A0When a subscription is created in the notifier=
, it stores any</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 &quot;Event&quot; header field &qu=
ot;id&quot; parameter as part of the subscription</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 information (along with the event =
package name).</font></div><div><font color=3D"#000099"><br></font></div><d=
iv><font color=3D"#000099">ReplacementText:</font></div><div><font color=3D=
"#000099">=A0 =A0o =A0When a subscription is created in the notifier, it st=
ores **the**</font></div>



<div><font color=3D"#000099">=A0 =A0 =A0 **&quot;id&quot; parameter in &quo=
t;Event&quot; header field** as part of the subscription</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 information (along with the event =
package name).</font></div><div><font color=3D"#000099">-------------------=
----------------------------------</font></div><div><font color=3D"#000099"=
>4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transactions</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0Neither SUBSCRIBE nor NOTIFY requests can be canceled. =A0If a UAS</=
font></div><div><font color=3D"#000099">=A0 =A0receives a CANCEL request th=
at matches a known SUBSCRIBE or NOTIFY</font></div>



<div><font color=3D"#000099">=A0 =A0transaction, it MUST respond to the CAN=
CEL request, but otherwise</font></div>
<div><font color=3D"#000099">=A0 =A0ignore it. =A0In particular, the CANCEL=
 request MUST NOT affect</font></div><div><font color=3D"#000099">=A0 =A0pr=
ocessing of the SUBSCRIBE or NOTIFY request in any way.</font></div><div><f=
ont color=3D"#000099"><br>



</font></div><div><font color=3D"#000099">ReplacementText/Comment:</font></=
div><div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</sp=
an>What should be the response code for the CANCEL&#39;s reply ???</font></=
div>




<div><font color=3D"#000099"><span style=3D"white-space:pre-wrap">	</span>P=
referably 200-OK shall be used.</font></div><div><font color=3D"#000099">--=
---------------------------------------------------</font></div><div><font =
color=3D"#000099">4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transact=
ions</font></div>




<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0UACs SHOULD NOT send CANCEL requests for SU=
BSCRIBE or NOTIFY</font></div>



<div><font color=3D"#000099">=A0 =A0transactions.</font></div><div><font co=
lor=3D"#000099"><br></font></div><div><font color=3D"#000099">ReplacementTe=
xt:</font></div><div><font color=3D"#000099">=A0 =A0UACs **MUST** NOT send =
CANCEL requests for SUBSCRIBE or NOTIFY</font></div>




<div><font color=3D"#000099">=A0 =A0transactions. **UASs SHOULD send 200OK =
for CANCEL within NOTIFY=A0</font></div><div><font color=3D"#000099">=A0 =
=A0or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**</font></d=
iv><div><font color=3D"#000099">-------------------------------------------=
----------</font></div>



<div><font color=3D"#000099">5.3.2. =A0State Deltas</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">=A0 =A0In the case that the state information to b=
e conveyed is large, the</font></div>



<div><font color=3D"#000099">=A0 =A0event package may choose to detail a sc=
heme by which NOTIFY requests</font></div><div><font color=3D"#000099">
=A0 =A0contain state deltas instead of complete state.</font></div><div><fo=
nt color=3D"#000099"><br></font></div><div><font color=3D"#000099"><br></fo=
nt></div><div><font color=3D"#000099">ReplacementText:</font></div><div><fo=
nt color=3D"#000099">=A0 =A0In the case that the state information to be co=
nveyed is large, the</font></div>



<div><font color=3D"#000099">=A0 =A0event package may choose a scheme by wh=
ich NOTIFY requests</font></div>
<div><font color=3D"#000099">=A0 =A0contain state deltas instead of complet=
e state.</font></div><div><font color=3D"#000099"><br></font></div><div><fo=
nt color=3D"#000099">comment:<span style=3D"white-space:pre-wrap">	</span>d=
eleted words &quot;to detail&quot; in 2nd line.</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">
-----------------------------------------------------</font></div><div><fon=
t color=3D"#000099">5.3.2. =A0State Deltas</font></div><div><font color=3D"=
#000099"><br></font></div><div><font color=3D"#000099">CurrentText:</font><=
/div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If a NOTIFY request arrives that has a version number that is</font>=
</div><div><font color=3D"#000099">
=A0 =A0incremented by more than one, the subscriber knows that a state delt=
a</font></div><div><font color=3D"#000099">=A0 =A0has been missed; it ignor=
es the NOTIFY request containing the state</font></div><div><font color=3D"=
#000099">=A0 =A0delta (except for the version number, which it retains to d=
etect</font></div>




<div><font color=3D"#000099">=A0 =A0message loss), and re-sends a SUBSCRIBE=
 request to force a NOTIFY</font></div><div><font color=3D"#000099">=A0 =A0=
request containing a complete state snapshot.</font></div><div><font color=
=3D"#000099"><br>



</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099">=A0 =A0From this text, it infers that if any state delta l=
ost, then subscriber=A0</font></div>
<div><font color=3D"#000099">=A0 =A0sends refresh SUBSCRIBE (re-SUBSCRIBE) =
request. This re-SUBSCRIBE request inturn lead=A0</font></div><div><font co=
lor=3D"#000099">=A0 =A0to fetching the complete state information.=A0</font=
></div><div>



<font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=A0 =
=A0It may be a good diea to differentiate the re-SUBSCRIBE triggered by dif=
ferent scenarios.</font></div>
<div><font color=3D"#000099">=A0 =A0a) natural re-SUBSCRIBE request, which =
extends the SUBSCRIBE duration. In this=A0</font></div><div><font color=3D"=
#000099">=A0 =A0 =A0 case it is <b>not </b>necessary to fetch the complete =
state information</font></div>



<div><font color=3D"#000099">=A0 =A0b) re-SUBSCRIBE triggered because of lo=
st NOTIFY request (ref sec 5.3.2.=A0</font></div>
<div><font color=3D"#000099">=A0 =A0 =A0 VERSION number in received NOTIFY =
is increased by greater than ONE).=A0</font></div><div><font color=3D"#0000=
99">=A0 =A0 =A0 In this case subscriber wants to fetch the complete state i=
nformation.</font></div>



<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
=A0 =A0If we don&#39;t differentiate these two cases, then natural re-SUBSC=
RIBE request</font></div>
<div><font color=3D"#000099">=A0 =A0unnecessarily fetch the complete state =
information, which may not be necessary / expected.=A0</font></div><div><fo=
nt color=3D"#000099">=A0 =A0But uses higher network bandwidth to fetch the =
complete state information.</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
<div><font color=3D"#000099">5.4.7. =A0Notifier generation of NOTIFY reques=
ts</font></div><div><font color=3D"#000099"><br></font></div><div><font col=
or=3D"#000099">CurrentText:</font></div><div><font color=3D"#000099">=A0 =
=A0This section may optionally describe the behavior used to process the</f=
ont></div>



<div><font color=3D"#000099">=A0 =A0subsequent response.</font></div><div><=
font color=3D"#000099"><br>
</font></div><div><font color=3D"#000099">Comment:</font></div><div><font c=
olor=3D"#000099">=A0 =A0In this section 5.4.7, the above statement is not c=
lear, what is really meant to be.</font></div><div><font color=3D"#000099">=
<br></font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div><div><font color=3D"#000099">5.4.13. =A0Use of URIs to =
Retrieve State</font></div>
<div><font color=3D"#000099"><br></font></div><div><font color=3D"#000099">=
CurrentText:</font></div><div><font color=3D"#000099"><br></font></div><div=
><font color=3D"#000099">ReplacementText:</font></div><div><font color=3D"#=
000099">=A0 =A0In this context, I feel Use of &quot;URLs&quot; is more appr=
opriate than &quot;URIs&quot;.=A0</font></div>



<div><font color=3D"#000099">----------------------------------------------=
-------</font></div>
</div><div><font color=3D"#000099"><br></font></div><div><font color=3D"#00=
0099"><br></font></div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace">Thanks,</font></font><div><font face=3D"&#39;courier ne=
w&#39;, monospace" color=3D"#000099">Nataraju A.B.</font></div>



<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">E-ID=
 : <a href=3D"mailto:nataraju.ab@gmail.com" target=3D"_blank">nataraju.ab@g=
mail.com</a> / <a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">=
nataraju.sip@gmail.com</a></font></div>


<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">



+91-98455-95744
</font></div>
</div><br><br clear=3D"all"><div><br></div>-- <br><font color=3D"#000099"><=
font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Thanks,</font></f=
ont><div><font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monos=
pace" size=3D"1">Nataraju A.B.</font></font></div>

<br>
</div>

--f46d0401fde7b61d7b04be517da3--

From pravindran@sonusnet.com  Sun Apr 22 22:32:27 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
Thread-Index: AQHNIDOCuN/qLU9tjk656lTA6m0G85am5VEAgAD8KaA=
Date: Mon, 23 Apr 2012 05:32:16 +0000
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <CA+rAfUM0MCfEd-Rj5aykeZ24=Kcgxk6GQzi4jFPc8P6EUQuoxg@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227F98@inba-mail01.sonusnet.com> <CALiegf=4yKSAgSRHeazcos7k6M87Vn=o0RfOizeOPEkxG_H-8A@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E227FFD@inba-mail01.sonusnet.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com> <FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228098@inba-mail01.sonusnet.com> <CALiegf=q1HE4bn1dgDz7RqKFe3NCDxr3vOWtR5DTevWAo0rK6A@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C42DF5E46@ESESSCMS0356.eemea.ericsson.se> <4F8ECB82.7070805@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228C1B@inba-mail01.sonusnet.com> <4F9460AC.2080605@alum.mit.edu>
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Paul,

Could you please explain your view of why SIPS URI is a mistake for TLS tra=
nsport. I'm interested in knowing as apart from RFC 3261, Sec 3.1.4 of RFC =
5630 (recent RFC) also recommends the usage of SIPS URI with TCP as a trans=
port for TLS over TCP (http://tools.ietf.org/html/rfc5630#page-5) and there=
 is an explicit statement in the same para

"This specification does not make use of the transport=3Dtls parameter."

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Paul Kyzivat
>Sent: Monday, April 23, 2012 1:19 AM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>Partha,
>
>IMO the main thing we learned about sips is that it was a bad idea.
>I don't support making that mistake again.
>
>	Thanks,
>	Paul
>
>On 4/21/12 10:55 PM, Ravindran, Parthasarathi wrote:
>> Sal&  others,
>>
>> After reading lot of mails and thinking further, I agree with you that
>Websocket is an application protocol that is used as an transport for
>SIP. TLS is one another application protocol on top of TCP which was
>considered as transport (transport=3Dtls) earlier and deprecated as part
>of RFC 3261 in Sec 26.2.2. TLS and Websocket has some unique behavior
>like hop-by-hop protocol rather than end-to-end.
>>
>> TLS is adopted within SIP using new URI namely SIPS with
>transport=3Dtcp. In the same way, I'm proposing SIPWS which implies
>websocket over TCP, SIPWSS which implies websocket over TLS over TCP.
>Please let me know your opinion on the same.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>>> Behalf Of Salvatore Loreto
>>> Sent: Wednesday, April 18, 2012 7:41 PM
>>> To: sipcore@ietf.org
>>> Subject: Re: [sipcore] Stateless or Transaction stateful proxy
>>> possible in Websocket transport? [RE: Call for Adoption:
>>> draft-ibc-sipcore-sip- websocket-02]
>>>
>>> On 4/18/12 4:40 PM, Christer Holmberg wrote:
>>>> Hi,
>>>>
>>>> I may have misunderstood something in the conversation, but my
>>> suggestion would be to consider *NOT* mandating SIP WebSocket clients
>>> to implement SIP UDP and TCP.
>>>>
>>>> Especially if I write a JavaScript SIP browser app, I very likely
>>> would want to support only WebSocket.
>>>>
>>>> Also, I don't think we can directly compare with SCTP, since the
>>> typical use-case for WebSocket is browser apps (eventhough a native
>>> app can of course also use WS).
>>>
>>> I think that in general we can not compare SCTP to WebSocket at all
>>> because the first is a pure Transport Protocol, the latter is an
>>> application protocol that is used as transport.
>>>
>>>  From a pure WebSocket prospective, what you are going to define is
>>> SIP as a websocket sub-protocol that is transported on top of.
>>>
>>> The important thing to standardize from a SIPCore prospective is the
>>> interaction/translations that happens between the WebSocket server
>>> and the SIP proxy that are two different logical entities.
>>>
>>> my two cents
>>> Sal
>>>
>>> --
>>> Salvatore Loreto, PhD
>>> www.sloreto.com
>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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>=20
>=20
>> - how often are large sip messages universally required by a UA,
>>  rather than being required in certain contexts?
>=20
> A SIP UA typically just registers using a single transport and, in
> common scenarios, it is just reachable through the proxy it is
> connected to (i.e. Outbound scenarios). Therefore the UA would prefer
> to use TCP rather than UDP and then allow big SIP messages (in case
> they are needed, as for example when receiving a NOTIFY with a
> resource-list XML body).
>=20
> If a vendor or provider offers such a service, it might consider a
> waste of time to implement UDP transport.
>=20

Big payloads are often associated with presence but IIRC there is a =
vendor who could not use UDP due to big payloads and it was just using =
SDP. An imaginary example: a video streaming service built with SIP: =
several video streams for different video qualities and profiles, =
several audio streams for the different languages and several chat =
streams for the subtitles in different languages. That could get big :-)

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 23 Apr 2012 09:53:22 +0200
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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2012/4/22 Paul Kyzivat <pkyzivat@alum.mit.edu>:
>> I could imagine a "like-Skype" service using SIP protocol and
>> providing both the server infrastructure and the SIP client (i.e. a
>> SIP softphone). Such a provider could be interested in secure
>> communications so its SIP client would just implement TLS-TCP
>> transport for signaling and SRTP (with DTLS or SDES) for media. The
>> provider does not want and does not accept that their SIP clients are
>> reachable via UDP, neither it accepts that their SIP clients
>> communicate with SIP entities other than their "official" SIP
>> proxies/servers.
>
>
> A service such as you describe, where all the pieces are provided by a
> single provider, is a proprietary solution, even if it is built using pub=
lic
> standards.
>
> The more interesting case is when the service is more open, allowing
> differing clients, and maybe differing servers as well. That allows
> tradeoffs to be made.
>
> For instance, maybe somebody wants to make a client for your service that
> can be used by prisoners in a prison. And then maybe there is a law
> forbidding encryption from being used. (And don't get all over me about t=
he
> details of the example - it is just the first one that came to mind.)

The example is good :)

But it is still decision of the service provider whether to offer a
transport other than a secure one. This is, the provider could let
clients to use their own SIP clients but still require TLS-TCP as
unique transport. But in that case we are indeed talking about
service-agnostic SIP UA's that won't *use* some transports (UDP and
insecure TCP) at deployment time. So there was no reason to *build*
those SIP UA's with no support for those transports. Therefore I agree
with you.





>> 1) If a SIP phone implements both UDP and TCP, but allows the user to
>> disable UDP (so it no longer listens into a UDP socket), is that phone
>> "violating" RFC 3261 or not?
>
>
> This is a valid question to ask - 3261 isn't clear on what it means by
> "implement".
>
> IMO its currently mandatory to implement but not mandatory to use, so its
> valid to allow disabling it.
>
>
>> 2) If a SIP domain announces DNS NAPTR records just including SCTP
>> transport or just SCTP and TCP, is it "violating" RFC 3261? Note that
>> RFC 3263 states that, when NAPTR records are present, those are the
>> options to reach the server.
>
>
> Again I am inclined to think mandatory to implement, optional to use. So =
as
> long as the NAPTR config is a deployment-time decision its ok.

After rethinking about those two questions I also consider that these
cases don't violate RFC 3261. And it is unlikely to mandate people to
*enable* all the mandatory-to-implement transports in any scenario.





>> - A scenario in which *direct* interoperability with other SIP
>> transports won't take place (i.e. a SIP stack in JavaScript using
>> WebSocket transport, and whose SIP outbound proxy provides
>> interoperability with other SIP transports when required).
>
>
> ISTM that this is a good example of why relaxing the requirements in a
> careful way could still allow the sorts of interop we need. E.g. relax th=
e
> requirements on UAs that connect using outbound.

And I might add "...and the Outbound proxy/registrar server(s)
provides interoperability with UDP and TCP transports".



> I'm generally in favor of general approaches to things rather than narrow
> ones. But I can also see why people might not want to expend the time and
> effort to sort out all the issues of the general issue, and so do the nar=
row
> way for now. And certainly doing that doesn't preclude doing something
> broader later.

If in the future there was interest on the broad way (the separate
document updating RFC 3261) I would like to help.


Thanks a lot.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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Hi Paul,

[snip]

>>=20
>> Also let me some questions coming to my mind:
>>=20
>> 1) If a SIP phone implements both UDP and TCP, but allows the user to
>> disable UDP (so it no longer listens into a UDP socket), is that =
phone
>> "violating" RFC 3261 or not?
>=20
> This is a valid question to ask - 3261 isn't clear on what it means by =
"implement".
>=20
> IMO its currently mandatory to implement but not mandatory to use, so =
its valid to allow disabling it.
>=20

That's my reading as well. Assuming that, how is disabling a transport =
different from not implementing it? One can claim to implement transport =
X but never allow you to use it. Is RFC-compliance the only difference?


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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2012/4/23 Sa=C3=BAl Ibarra Corretg=C3=A9 <saul@ag-projects.com>:
> That's my reading as well. Assuming that, how is disabling a transport di=
fferent from not implementing it? One can claim to implement transport X bu=
t never allow you to use it. Is RFC-compliance the only difference?

If you buy a phone of vendor XXX which announces UDP transport in its
specifications but there is no way to use it, then the vendor lied you
:)

IMHO the point here is about devices not designed for specific
environments, but for "any" environment, so theorically it's good that
they MUST implement basic SIP transports as a requirement for
announcing a "SIP compliant" label. Whether certain transports are
later used or not is a deployment decision which depends on each
scenario requirements and topology.

Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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I am pretty sure we don't want to any new URI schemes for SIP we will howev=
er have to think about sow SIPS applies to websockets if at all.

Regards
Andy



> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: 22 April 2012 20:49
> To: sipcore@ietf.org
> Subject: Re: [sipcore] Websocket is a new transport or new URI?
>=20
> Partha,
>=20
> IMO the main thing we learned about sips is that it was a bad idea.
> I don't support making that mistake again.
>=20
> 	Thanks,
> 	Paul
>=20
> On 4/21/12 10:55 PM, Ravindran, Parthasarathi wrote:
> > Sal&  others,
> >
> > After reading lot of mails and thinking further, I agree with you
> that Websocket is an application protocol that is used as an transport
> for SIP. TLS is one another application protocol on top of TCP which
> was considered as transport (transport=3Dtls) earlier and deprecated as
> part of RFC 3261 in Sec 26.2.2. TLS and Websocket has some unique
> behavior like hop-by-hop protocol rather than end-to-end.
> >
> > TLS is adopted within SIP using new URI namely SIPS with
> transport=3Dtcp. In the same way, I'm proposing SIPWS which implies
> websocket over TCP, SIPWSS which implies websocket over TLS over TCP.
> Please let me know your opinion on the same.
> >
> > Thanks
> > Partha
> >
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> >> Behalf Of Salvatore Loreto
> >> Sent: Wednesday, April 18, 2012 7:41 PM
> >> To: sipcore@ietf.org
> >> Subject: Re: [sipcore] Stateless or Transaction stateful proxy
> possible
> >> in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-
> sip-
> >> websocket-02]
> >>
> >> On 4/18/12 4:40 PM, Christer Holmberg wrote:
> >>> Hi,
> >>>
> >>> I may have misunderstood something in the conversation, but my
> >> suggestion would be to consider *NOT* mandating SIP WebSocket
> clients to
> >> implement SIP UDP and TCP.
> >>>
> >>> Especially if I write a JavaScript SIP browser app, I very likely
> >> would want to support only WebSocket.
> >>>
> >>> Also, I don't think we can directly compare with SCTP, since the
> >> typical use-case for WebSocket is browser apps (eventhough a native
> app
> >> can of course also use WS).
> >>
> >> I think that in general we can not compare SCTP to WebSocket at all
> >> because the first is a pure Transport Protocol, the latter is an
> >> application protocol that is used as transport.
> >>
> >>  From a pure WebSocket prospective, what you are going to define is
> SIP
> >> as a websocket sub-protocol that is transported on top of.
> >>
> >> The important thing to standardize from a SIPCore prospective is the
> >> interaction/translations that happens between the WebSocket server
> and
> >> the SIP proxy that are two different logical entities.
> >>
> >> my two cents
> >> Sal
> >>
> >> --
> >> Salvatore Loreto, PhD
> >> www.sloreto.com
> >>
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From pkyzivat@alum.mit.edu  Mon Apr 23 07:02:23 2012
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Subject: Re: [sipcore] comments on draft-ietf-sipcore-rfc3265bis-08.txt
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Nataraju,

Your comments would have been much more helpful if received some months 
ago when the document went through WGLC. This document is now *very* far 
down the process to publication, having already gone through IESG last 
call, IANA review, etc. and is now in final IESG Evaluation. Changes at 
this point are very disruptive.

If you feel your suggestions are essential then you will have to go to 
extremes to pull this document back. I don't see anything below that 
convinces me to do that. AFAICT the comments below are all minor 
editorial except for the first, which suggests changing the mechanism.

	Thanks,
	Paul (as co-chair)

On 4/23/12 12:58 AM, Nataraju A.B wrote:
> All,
>
> Some logical comments on the draft-ietf-sipcore-rfc3265bis-08.txt.
>
> -----------------------------------------------------
> 5.3.2.  State Deltas
>
> CurrentText:
>
>     If a NOTIFY request arrives that has a version number that is
>     incremented by more than one, the subscriber knows that a state delta
>     has been missed; it ignores the NOTIFY request containing the state
>     delta (except for the version number, which it retains to detect
>     message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
>     request containing a complete state snapshot.
>
> Comment:
>     From this text, it infers that if any state delta lost, then subscriber
>     sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
> request inturn lead
>     to fetching the complete state information.
>
>     It may be a good diea to differentiate the re-SUBSCRIBE triggered by
> different scenarios.
>     a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
> duration. In this
>        case it is not necessary to fetch the complete state information
>     b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec
> 5.3.2.
>        VERSION number in received NOTIFY is increased by greater than ONE).
>        In this case subscriber wants to fetch the complete state
> information.
>
>     If we don't differentiate these two cases, then natural re-SUBSCRIBE
> request
>     unnecessarily fetch the complete state information, which may not be
> necessary / expected.
>     But uses higher network bandwidth to fetch the complete state
> information.
> -----------------------------------------------------
> -----------------------------------------------------
> 4.4.1.  Dialog Creation and Termination
>
> CurrentText:
>
>        A subscriber may send a SUBSCRIBE request with an "Expires" header
>        field of 0 in order to trigger the sending of such a NOTIFY
>        request; however, for the purposes of subscription and dialog
>        lifetime, the subscription is not considered terminated until the
>        NOTIFY transaction with a "Subscription-State" of "terminated"
>        completes.
>
> Comment:
> We may have to mention how long the dialog must be kept alive, before
> being cleaned up. Should we delete the dialog immediately after 200-SUB
> or wait till NOTIFY with "Subscription-State"="terminated". If we always
> wait till Last Notify for dialog termination, then 200-SUB don't have any
> impact on the dialog termination. Atleast some normative text is of
> necessity, for better understanding of the topic.
> -----------------------------------------------------
> -----------------------------------------------------
> 3.1.1.  Subscription Duration
>
> CurrentText:
>
>     SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
>     in [RFC3261]).  This expires value indicates the duration of the
>     subscription.  In order to keep subscriptions effective beyond the
>     duration communicated in the "Expires" header field, subscribers need
>     to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>     request on the same dialog as defined in [RFC3261].
>
> Comment:
> We can add a normative text saying, the refresh time will be
> normally 50% to 75% of subscription duration.
>
> -----------------------------------------------------
> -----------------------------------------------------
> 4.1.2.1.  Requesting a Subscription
>
> CurrentText:
>     [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
>     an "Accept" header field, but that field does not indicate a media
>     type that the notifier is capable of generating in its NOTIFY
>     requests, then the proper error response is 406 (Not Acceptable).
>
> COMMENT:
> It is better to mention it as a normative text (indented).
>
> -----------------------------------------------------
> -----------------------------------------------------
> 4.1.2.  Creating and Maintaining Subscriptions
>
>
> CurrentText:
>
>     From the subscriber's perspective, a subscription proceeds according
>     to the following state diagram:
>
> Comment:
> Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
> received in notify_wait state (because very often this is possible).
> If no change in state of FSM, then some normative text shall convey the
> same.
>          It may be no-operation, but better to mention the same.
> -----------------------------------------------------
> -----------------------------------------------------
> 3.1.1.  Subscription Duration
>
> CurrentText:
>        In addition to being a request to unsubscribe, a SUBSCRIBE request
>        with "Expires" of 0 also causes a fetch of state; see
>        Section 4.4.3.
>
> Comment:
>       The above text has been mentioned to be normative text. But I feel
> this
>       shall be written as a regualr text in the draft/rfc. because this
> is the
>       expected behavior from the notifier.
> -----------------------------------------------------
> -----------------------------------------------------
> 3.2.1.
>
> CurrentText:
>     This body will contain either the state of the subscribed resource or
>     a pointer to such state in the form of a URI (see Section 5.4.13).
>
> ReplacementText/Comment:
>     This body **SHALL** contain either the state of the subscribed
> resource or
>     a pointer to such state in the form of a **URL** (see Section 5.4.13).
> -----------------------------------------------------
> -----------------------------------------------------
> 4.2.1.  Subscription Establishment and Maintenance
>
> CurrentText:
>        inability to signal the success of a REFER request while signaling
>        a problem with the subscription; and difficulty performing one
>        action without the other.  Additionally, the proper exchange of
>
> ReplacementText:
>        inability to signal the success of a REFER request while signaling
>        a problem with the subscription; and difficulty **in** performing one
>        action without the other.  Additionally, the proper exchange of
>
> -----------------------------------------------------
>
> Thanks,
> Nataraju A B
>
> ---------- Forwarded message ----------
> From: *Nataraju A.B* <nataraju.sip@gmail.com
> <mailto:nataraju.sip@gmail.com>>
> Date: Fri, Apr 13, 2012 at 10:53 PM
> Subject: draft-ietf-sipcore-rfc3265bis-08.txt - Review comments
> (Nataraju A B)
> To: Adam Roach <adam@nostrum.com <mailto:adam@nostrum.com>>,
> sipcore@ietf.org <mailto:sipcore@ietf.org>
>
>
> Adam,
>
> Please find some editorial comments on the draft
> "draft draft-ietf-sipcore-rfc3265bis-08.txt"
>
> For simplicity, I have mentioned both the current text and proposed text.
>
> The comments / corrective text can be found with prefix and suffix  --- **
>
> -----------------------------------------------------
> 1.1.  Overview of Operation
> CurrentText:
>     The general concept is that entities in the network can subscribe to
>     resource or call state for various resources or calls in the network,
>     and those entities (or entities acting on their behalf) can send
>     notifications when those states change.
>
> ReplacementText:
>     The general concept is that entities in the network can subscribe to
>     resource or call state **at / of** various resources or calls in the
> network,
>     and those entities (or entities acting on their behalf) **who** can send
>     notifications when those states change.
> -----------------------------------------------------
>
> 1.1.  Overview of Operation
>
> CurrentText:
>     Subscriptions are expired and must be refreshed by subsequent
>     SUBSCRIBE requests.
>
> ReplacementText:
>     Subscriptions are expired and must be refreshed by subsequent
>     SUBSCRIBE requests **before expiration**.
> -----------------------------------------------------
>
> 2.  Definitions
>
> CurrentText:
>     Subscriber:  A subscriber is a user agent which receives NOTIFY
>        requests from notifiers; these NOTIFY requests contain information
>        about the state of a resource in which the subscriber is
>        interested.  Subscribers typically also generate SUBSCRIBE
>        requests and send them to notifiers to create subscriptions.
>
> Comment:
> After this information, we can add a normative text which indicates
> when a SIP entity might get NOTIFY messages without actually
> sending SUBSCRIBE request. For example, REFER request might end up with
> implicit subscription. Also mention any other requests which may lead to
> implicit subscriptions (other than REFER).
> -----------------------------------------------------
> 2.  Definitions
>
> CurrentText:
>     Subscription:  A subscription is a set of application state
>        associated with a dialog.  This application state includes a
>        pointer to the associated dialog, the event package name, and
>        possibly an identification token.  Event packages will define
>        additional subscription state information.  By definition,
>        subscriptions exist in both a subscriber and a notifier.
>
> ReplacementText:
>     Subscription:  A subscription is a set of application **states**
>        associated with a dialog.  **The** application state includes a
>        pointer to the associated dialog, the event package name, and
>        possibly an identification token.  Event packages will define
>        additional subscription state information.  By definition,
>        subscriptions exist in both a subscriber and a notifier.
> -----------------------------------------------------
> 3.1.1.  Subscription Duration
>
> CurrentText:
>
>     subscription.  In order to keep subscriptions effective beyond the
>     duration communicated in the "Expires" header field, subscribers need
>     to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>     request on the same dialog as defined in [RFC3261].
>
> ReplacementText
> 3.1.1.  Subscription Duration
>
>     subscription.  In order to keep subscriptions effective beyond the
>     duration communicated in the "Expires" header field, subscribers need
>     to refresh subscriptions **before expiration** on a periodic basis
> using a new SUBSCRIBE
>     request on the same dialog as defined in [RFC3261].
> -----------------------------------------------------
> 3.1.1.  Subscription Duration
>
> CurrentText:
>        In addition to being a request to unsubscribe, a SUBSCRIBE request
>        with "Expires" of 0 also causes a fetch of state; see
>        Section 4.4.3.
>
> Comment:
>       The above text has been mentioned to be normative text. But I feel
> this
>       shall be written as a regualr text in the draft/rfc. because this
> is the
>       expected behavior from the notifier.
> -----------------------------------------------------
> 3.1.2.  Identification of Subscribed Events and Event Classes
>
> CurrentText:
>     Subscribers MUST include exactly one "Event" header field in
>     SUBSCRIBE requests, indicating to which event or class of events they
>     are subscribing.  The "Event" header field will contain a token which
>     indicates the type of state for which a subscription is being
>     requested.  This token will be registered with the IANA and will
>     correspond to an event package which further describes the semantics
>     of the event or event class.
>
>
> ReplacementText:
>     Subscribers MUST include exactly one "Event" header field in
>     SUBSCRIBE requests, indicating to which event or class of events they
>     are subscribing.  The "Event" header field **MUST** contain a token
> which
>     indicates the type of state for which a subscription is being
>     requested.  This token will be registered with the IANA and will
>     correspond to an event package which further describes the semantics
>     of the event or event class.
> -----------------------------------------------------
> 3.1.  SUBSCRIBE
>
> CurrentText:
>     The SUBSCRIBE method is used to request current state and state
>     updates from a remote node.  SUBSCRIBE requests are target refresh
>
>
> ReplacementText:
>     The SUBSCRIBE method is used to request current state and
> **subsequent** state
>     updates from a remote node.  SUBSCRIBE requests are target refresh
>
>
> -----------------------------------------------------
> 3.1.1.  Subscription Duration
>
> CurrentText:
>
>     SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
>     in [RFC3261]).  This expires value indicates the duration of the
>     subscription.  In order to keep subscriptions effective beyond the
>     duration communicated in the "Expires" header field, subscribers need
>     to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>     request on the same dialog as defined in [RFC3261].
>
> Comment:
> We can add a normative text saying, the refresh time will be
> normally 50% to 75% of subscription duration.
>
> -----------------------------------------------------
> 3.2.1.
>
> CurrentText:
>     This body will contain either the state of the subscribed resource or
>     a pointer to such state in the form of a URI (see Section 5.4.13).
>
> ReplacementText/Comment:
>     This body **SHALL** contain either the state of the subscribed
> resource or
>     a pointer to such state in the form of a **URL** (see Section 5.4.13).
> -----------------------------------------------------
> 4.1.2.  Creating and Maintaining Subscriptions
>
>
> CurrentText:
>
>     From the subscriber's perspective, a subscription proceeds according
>     to the following state diagram:
>
> Comment:
> Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
> received in notify_wait state (because very often this is possible).
> If no change in state of FSM, then some normative text shall convey the
> same.
>          It may be no-operation, but better to mention the same.
> -----------------------------------------------------
> 4.1.2.1.  Requesting a Subscription
>
> CurrentText:
>     When a subscriber wishes to subscribe to a particular state for a
>     resource, it forms a SUBSCRIBE request.  If the initial SUBSCRIBE
>
> ReplacementText:
>     When a subscriber wishes to subscribe to a particular state **of** a
>     resource, it forms a SUBSCRIBE request **according to sec 3.1 and
> send it across to notifier**.  If the initial SUBSCRIBE
>
> -----------------------------------------------------
> 4.1.2.1.  Requesting a Subscription
>
> CurrentText:
>     [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
>     an "Accept" header field, but that field does not indicate a media
>     type that the notifier is capable of generating in its NOTIFY
>     requests, then the proper error response is 406 (Not Acceptable).
>
> COMMENT:
> It is better to mention it as a normative text (indented).
>
> -----------------------------------------------------
> 4.1.3.  Receiving and Processing State Information
>
> CurrentText:
>
>     If the "Subscription-State" value is "terminated", the subscriber
>     MUST consider the subscription terminated.  The "expires" parameter
>     has no semantics for "terminated" -- notifiers SHOULD NOT include an
>
> ReplacementText:
>
>     If the "Subscription-State" value is "terminated", the subscriber
>     MUST consider the subscription terminated.  The "expires" parameter
>     has no semantics for "terminated" **state**. Notifiers SHOULD NOT
> include an
> -----------------------------------------------------
>
> CurrentText:
>     deactivated:  The subscription has been terminated, but the
>        subscriber SHOULD retry immediately with a new subscription.  One
>
> ReplacementText:
>     deactivated:  The subscription has been terminated, but the
>        subscriber SHOULD retry with a new subscription.  One
>
>     Is It mandatory to resubscribe immediately. if not remove the word
> "immediately".
>     If it is delayed, what are the consequences ????
>
> -----------------------------------------------------
> 4.2.1.  Subscription Establishment and Maintenance
>
> CurrentText:
>        inability to signal the success of a REFER request while signaling
>        a problem with the subscription; and difficulty performing one
>        action without the other.  Additionally, the proper exchange of
>
> ReplacementText:
>        inability to signal the success of a REFER request while signaling
>        a problem with the subscription; and difficulty **in** performing one
>        action without the other.  Additionally, the proper exchange of
>
> -----------------------------------------------------
> 4.2.1.4.  Refreshing of Subscriptions
>
> CurrentText:
>     If no refresh for a notification address is received before its
>     expiration time, the subscription is removed.  When removing a
>
> ReplacementText:
>     If no refresh for a notification is received before its
>     expiration time, the subscription is removed.  When removing a
> Comment:Word "address" needs to be removed.
> -----------------------------------------------------
> 4.3.  Proxy Behavior
>
> CurrentText:
>     Proxies that did not add a Record-Route header field to the initial
>     SUBSCRIBE request MUST NOT add a Record-Route header field to any of
>     the associated NOTIFY requests.
>
> ReplacementText:
>     Proxies that did not add a Record-Route header field to the initial
>     SUBSCRIBE request MUST NOT add a Record-Route header field to any of
>     the associated NOTIFY requests **or any SUBSCRIBE refresh requests**.
> -----------------------------------------------------
> 4.4.1.  Dialog Creation and Termination
>
> CurrentText:
>
>        A subscriber may send a SUBSCRIBE request with an "Expires" header
>        field of 0 in order to trigger the sending of such a NOTIFY
>        request; however, for the purposes of subscription and dialog
>        lifetime, the subscription is not considered terminated until the
>        NOTIFY transaction with a "Subscription-State" of "terminated"
>        completes.
>
> Comment:
> We may have to mention how long the dialog must be kept alive, before
> being cleaned up. Do we delete the dialog immediately after 200-SUB
> or wait till NOTIFY with "Subscription-State"="terminated". If we always
> wait till Last Notify for dialog termination, then 200-SUB don't have any
> impact on the dialog termination. Atleast some normative text is of
> necessity, for better understanding of the topic.
> -----------------------------------------------------
> 4.4.3.  Polling Resource State
>
> CurrentText:
>
>     Note that the NOTIFY requests triggered by SUBSCRIBE requests with
> "Expires" header fields of 0 will contain a "Subscription-State"
>     value of "terminated", and a "reason" parameter of "timeout".
>
> ReplacementText:
>     Note that the NOTIFY requests triggered by SUBSCRIBE requests with
> "Expires" header fields of 0 **shall** contain a "Subscription-State"
>     value of "terminated", and a "reason" parameter of "timeout".
> -----------------------------------------------------
> 4.5.2.  Sharing Dialogs
>
> CurrentText:
>        for a dialog (e.g., route set handling).  In particular, multiple
>        subscriptions within a dialog are expire independently, and
>        require independent subscription refreshes.
>
> ReplacementText:
>        for a dialog (e.g., route set handling).  In particular, multiple
>        subscriptions within a dialog **operates** independently, and
>        require independent subscription refreshes.
> -----------------------------------------------------
> 4.5.2.  Sharing Dialogs
>
> CurrentText:
>     o  Subscribers MAY include an "id" parameter in SUBSCRIBE request
> "Event" header field to allow differentiation between multiple
>
> ReplacementText:
>     o  Subscribers MAY include an "id" parameter in SUBSCRIBE **request's**
> "Event" header field to allow differentiation between multiple
> OR
>     o  Subscribers MAY include an "id" parameter in **"Event" header field
>        of SUBSCRIBE request** to allow differentiation between multiple
> -----------------------------------------------------
> 4.5.2.  Sharing Dialogs
>
> CurrentText:
>     o  When a subscriber refreshes a the subscription timer, the
>        SUBSCRIBE request MUST contain the same "Event" header field "id"
>        parameter as was present in the SUBSCRIBE request that created the
>        subscription.  (Otherwise, the notifier will interpret the
>
> ReplacementText:
>     o  When a subscriber refreshes a the subscription timer, the
>        SUBSCRIBE request MUST contain the same **"id" parameter in "Event"
>        header field, which** was present in the SUBSCRIBE request that
> created the
>        subscription.  (Otherwise, the notifier will interpret the
> -----------------------------------------------------
> 4.5.2.  Sharing Dialogs
>
> CurrentText:
>
>     o  When a subscription is created in the notifier, it stores any
> "Event" header field "id" parameter as part of the subscription
>        information (along with the event package name).
>
> ReplacementText:
>     o  When a subscription is created in the notifier, it stores **the**
>        **"id" parameter in "Event" header field** as part of the
> subscription
>        information (along with the event package name).
> -----------------------------------------------------
> 4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>
> CurrentText:
>
>     Neither SUBSCRIBE nor NOTIFY requests can be canceled.  If a UAS
>     receives a CANCEL request that matches a known SUBSCRIBE or NOTIFY
>     transaction, it MUST respond to the CANCEL request, but otherwise
>     ignore it.  In particular, the CANCEL request MUST NOT affect
>     processing of the SUBSCRIBE or NOTIFY request in any way.
>
> ReplacementText/Comment:
> What should be the response code for the CANCEL's reply ???
> Preferably 200-OK shall be used.
> -----------------------------------------------------
> 4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>
> CurrentText:
>
>     UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>     transactions.
>
> ReplacementText:
>     UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>     transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY
>     or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**
> -----------------------------------------------------
> 5.3.2.  State Deltas
>
> CurrentText:
>
>     In the case that the state information to be conveyed is large, the
>     event package may choose to detail a scheme by which NOTIFY requests
>     contain state deltas instead of complete state.
>
>
> ReplacementText:
>     In the case that the state information to be conveyed is large, the
>     event package may choose a scheme by which NOTIFY requests
>     contain state deltas instead of complete state.
>
> comment:deleted words "to detail" in 2nd line.
>
> -----------------------------------------------------
> 5.3.2.  State Deltas
>
> CurrentText:
>
>     If a NOTIFY request arrives that has a version number that is
>     incremented by more than one, the subscriber knows that a state delta
>     has been missed; it ignores the NOTIFY request containing the state
>     delta (except for the version number, which it retains to detect
>     message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
>     request containing a complete state snapshot.
>
> Comment:
>     From this text, it infers that if any state delta lost, then subscriber
>     sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
> request inturn lead
>     to fetching the complete state information.
>
>     It may be a good diea to differentiate the re-SUBSCRIBE triggered by
> different scenarios.
>     a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
> duration. In this
>        case it is *not *necessary to fetch the complete state information
>     b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec
> 5.3.2.
>        VERSION number in received NOTIFY is increased by greater than ONE).
>        In this case subscriber wants to fetch the complete state
> information.
>
>     If we don't differentiate these two cases, then natural re-SUBSCRIBE
> request
>     unnecessarily fetch the complete state information, which may not be
> necessary / expected.
>     But uses higher network bandwidth to fetch the complete state
> information.
> -----------------------------------------------------
> 5.4.7.  Notifier generation of NOTIFY requests
>
> CurrentText:
>     This section may optionally describe the behavior used to process the
>     subsequent response.
>
> Comment:
>     In this section 5.4.7, the above statement is not clear, what is
> really meant to be.
>
> -----------------------------------------------------
> 5.4.13.  Use of URIs to Retrieve State
>
> CurrentText:
>
> ReplacementText:
>     In this context, I feel Use of "URLs" is more appropriate than "URIs".
> -----------------------------------------------------
>
>
> Thanks,
> Nataraju A.B.
> E-ID : nataraju.ab@gmail.com <mailto:nataraju.ab@gmail.com> /
> nataraju.sip@gmail.com <mailto:nataraju.sip@gmail.com>
> +91-98455-95744
>
>
>
> --
> Thanks,
> Nataraju A.B.
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From pkyzivat@alum.mit.edu  Mon Apr 23 07:11:22 2012
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/23/12 3:53 AM, IĆ±aki Baz Castillo wrote:
> 2012/4/22 Paul Kyzivat<pkyzivat@alum.mit.edu>:
>>> I could imagine a "like-Skype" service using SIP protocol and
>>> providing both the server infrastructure and the SIP client (i.e. a
>>> SIP softphone). Such a provider could be interested in secure
>>> communications so its SIP client would just implement TLS-TCP
>>> transport for signaling and SRTP (with DTLS or SDES) for media. The
>>> provider does not want and does not accept that their SIP clients are
>>> reachable via UDP, neither it accepts that their SIP clients
>>> communicate with SIP entities other than their "official" SIP
>>> proxies/servers.
>>
>>
>> A service such as you describe, where all the pieces are provided by a
>> single provider, is a proprietary solution, even if it is built using public
>> standards.
>>
>> The more interesting case is when the service is more open, allowing
>> differing clients, and maybe differing servers as well. That allows
>> tradeoffs to be made.
>>
>> For instance, maybe somebody wants to make a client for your service that
>> can be used by prisoners in a prison. And then maybe there is a law
>> forbidding encryption from being used. (And don't get all over me about the
>> details of the example - it is just the first one that came to mind.)
>
> The example is good :)
>
> But it is still decision of the service provider whether to offer a
> transport other than a secure one. This is, the provider could let
> clients to use their own SIP clients but still require TLS-TCP as
> unique transport. But in that case we are indeed talking about
> service-agnostic SIP UA's that won't *use* some transports (UDP and
> insecure TCP) at deployment time. So there was no reason to *build*
> those SIP UA's with no support for those transports. Therefore I agree
> with you.

That's good! :)

>>> 1) If a SIP phone implements both UDP and TCP, but allows the user to
>>> disable UDP (so it no longer listens into a UDP socket), is that phone
>>> "violating" RFC 3261 or not?
>>
>>
>> This is a valid question to ask - 3261 isn't clear on what it means by
>> "implement".
>>
>> IMO its currently mandatory to implement but not mandatory to use, so its
>> valid to allow disabling it.
>>
>>
>>> 2) If a SIP domain announces DNS NAPTR records just including SCTP
>>> transport or just SCTP and TCP, is it "violating" RFC 3261? Note that
>>> RFC 3263 states that, when NAPTR records are present, those are the
>>> options to reach the server.
>>
>>
>> Again I am inclined to think mandatory to implement, optional to use. So as
>> long as the NAPTR config is a deployment-time decision its ok.
>
> After rethinking about those two questions I also consider that these
> cases don't violate RFC 3261. And it is unlikely to mandate people to
> *enable* all the mandatory-to-implement transports in any scenario.
>
>
>
>
>
>>> - A scenario in which *direct* interoperability with other SIP
>>> transports won't take place (i.e. a SIP stack in JavaScript using
>>> WebSocket transport, and whose SIP outbound proxy provides
>>> interoperability with other SIP transports when required).
>>
>>
>> ISTM that this is a good example of why relaxing the requirements in a
>> careful way could still allow the sorts of interop we need. E.g. relax the
>> requirements on UAs that connect using outbound.
>
> And I might add "...and the Outbound proxy/registrar server(s)
> provides interoperability with UDP and TCP transports".

Yes. That is what I meant - relax requirement on the UA that supports 
outbound, but leave the requirement in place for the proxy/registrar.

>> I'm generally in favor of general approaches to things rather than narrow
>> ones. But I can also see why people might not want to expend the time and
>> effort to sort out all the issues of the general issue, and so do the narrow
>> way for now. And certainly doing that doesn't preclude doing something
>> broader later.
>
> If in the future there was interest on the broad way (the separate
> document updating RFC 3261) I would like to help.

Thanks. I'll keep that in mind.

	Thanks,
	Paul

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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/23/12 3:54 AM, Saśl Ibarra Corretgé wrote:
> Hi Paul,
>
> [snip]
>
>>>
>>> Also let me some questions coming to my mind:
>>>
>>> 1) If a SIP phone implements both UDP and TCP, but allows the user to
>>> disable UDP (so it no longer listens into a UDP socket), is that phone
>>> "violating" RFC 3261 or not?
>>
>> This is a valid question to ask - 3261 isn't clear on what it means by "implement".
>>
>> IMO its currently mandatory to implement but not mandatory to use, so its valid to allow disabling it.
>>
>
> That's my reading as well. Assuming that, how is disabling a transport different from not implementing it? One can claim to implement transport X but never allow you to use it. Is RFC-compliance the only difference?

The difference is that the decision about use is put into the hands of 
the person deploying the equipment rather than the people writing the 
code that goes into the equipment. Then, if an interop problem arises 
due to the choices made during deployment, they can be fixed during 
deployment merely by reconfiguration. The conformance of the equipment 
the deployer already bought makes it possible to solve the problem 
without buying new equipment.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Mon Apr 23 07:29:35 2012
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/23/12 3:48 AM, Saśl Ibarra Corretgé wrote:
>>
>>
>>> - how often are large sip messages universally required by a UA,
>>>   rather than being required in certain contexts?

Note that I am largely playing devil's advocate on this particular issue.

>> A SIP UA typically just registers using a single transport and, in
>> common scenarios, it is just reachable through the proxy it is
>> connected to (i.e. Outbound scenarios). Therefore the UA would prefer
>> to use TCP rather than UDP and then allow big SIP messages (in case
>> they are needed, as for example when receiving a NOTIFY with a
>> resource-list XML body).
>>
>> If a vendor or provider offers such a service, it might consider a
>> waste of time to implement UDP transport.
>>
>
> Big payloads are often associated with presence but IIRC there is a vendor who could not use UDP due to big payloads and it was just using SDP. An imaginary example: a video streaming service built with SIP: several video streams for different video qualities and profiles, several audio streams for the different languages and several chat streams for the subtitles in different languages. That could get big :-)

The usual argument, which you seem to be making again, is that some 
vendor decides it only needs to support UDP because it doesn't 
anticipate being deployed in any situations where messages will exceed 
the PDU size. And the usual counter-argument is that it is nearly 
impossible to guarantee that this will always be the case, and hence the 
implementation MUST be prepared to use TCP.

Here we are arguing the flip side - someone arguing that it NEEDS TCP in 
all its deployments because it will have *some* big messages in every 
deployment, and hence sees no reason to implement UDP. I'm just arguing 
that some deployer might have a usage in mind where UDP *would* work all 
the time, and hence decide to deploy with only UDP configured.

Its a weak argument. But I can imagine someone making it. E.g. somebody 
might do so in order to optimize a benchmark.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Mon Apr 23 07:37:39 2012
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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On 4/23/12 1:32 AM, Ravindran, Parthasarathi wrote:
> Paul,
>
> Could you please explain your view of why SIPS URI is a mistake for TLS transport.

My view is that the *intent* of sips was to provide the sort of 
"guarantee" of security that https provides. But sips doesn't provide 
that guarantee, because it isn't e2e and a variety of other things. The 
end result is that with sips you have very little more confidence that 
you are securely talking to who you think than if you use best effort by 
using TLS on your first hop. In essence, sips provides a false sense of 
security to the naive, but little more than that.

	Thanks,
	Paul

> I'm interested in knowing as apart from RFC 3261, Sec 3.1.4 of RFC 5630 (recent RFC) also recommends the usage of SIPS URI with TCP as a transport for TLS over TCP (http://tools.ietf.org/html/rfc5630#page-5) and there is an explicit statement in the same para
>
> "This specification does not make use of the transport=tls parameter."
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Paul Kyzivat
>> Sent: Monday, April 23, 2012 1:19 AM
>> To: sipcore@ietf.org
>> Subject: Re: [sipcore] Websocket is a new transport or new URI?
>>
>> Partha,
>>
>> IMO the main thing we learned about sips is that it was a bad idea.
>> I don't support making that mistake again.
>>
>> 	Thanks,
>> 	Paul
>>
>> On 4/21/12 10:55 PM, Ravindran, Parthasarathi wrote:
>>> Sal&   others,
>>>
>>> After reading lot of mails and thinking further, I agree with you that
>> Websocket is an application protocol that is used as an transport for
>> SIP. TLS is one another application protocol on top of TCP which was
>> considered as transport (transport=tls) earlier and deprecated as part
>> of RFC 3261 in Sec 26.2.2. TLS and Websocket has some unique behavior
>> like hop-by-hop protocol rather than end-to-end.
>>>
>>> TLS is adopted within SIP using new URI namely SIPS with
>> transport=tcp. In the same way, I'm proposing SIPWS which implies
>> websocket over TCP, SIPWSS which implies websocket over TLS over TCP.
>> Please let me know your opinion on the same.
>>>
>>> Thanks
>>> Partha
>>>
>>>> -----Original Message-----
>>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>>>> Behalf Of Salvatore Loreto
>>>> Sent: Wednesday, April 18, 2012 7:41 PM
>>>> To: sipcore@ietf.org
>>>> Subject: Re: [sipcore] Stateless or Transaction stateful proxy
>>>> possible in Websocket transport? [RE: Call for Adoption:
>>>> draft-ibc-sipcore-sip- websocket-02]
>>>>
>>>> On 4/18/12 4:40 PM, Christer Holmberg wrote:
>>>>> Hi,
>>>>>
>>>>> I may have misunderstood something in the conversation, but my
>>>> suggestion would be to consider *NOT* mandating SIP WebSocket clients
>>>> to implement SIP UDP and TCP.
>>>>>
>>>>> Especially if I write a JavaScript SIP browser app, I very likely
>>>> would want to support only WebSocket.
>>>>>
>>>>> Also, I don't think we can directly compare with SCTP, since the
>>>> typical use-case for WebSocket is browser apps (eventhough a native
>>>> app can of course also use WS).
>>>>
>>>> I think that in general we can not compare SCTP to WebSocket at all
>>>> because the first is a pure Transport Protocol, the latter is an
>>>> application protocol that is used as transport.
>>>>
>>>>   From a pure WebSocket prospective, what you are going to define is
>>>> SIP as a websocket sub-protocol that is transported on top of.
>>>>
>>>> The important thing to standardize from a SIPCore prospective is the
>>>> interaction/translations that happens between the WebSocket server
>>>> and the SIP proxy that are two different logical entities.
>>>>
>>>> my two cents
>>>> Sal
>>>>
>>>> --
>>>> Salvatore Loreto, PhD
>>>> www.sloreto.com
>>>>
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>
>>
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>


From saul@ag-projects.com  Mon Apr 23 07:56:30 2012
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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>=20
> The difference is that the decision about use is put into the hands of =
the person deploying the equipment rather than the people writing the =
code that goes into the equipment. Then, if an interop problem arises =
due to the choices made during deployment, they can be fixed during =
deployment merely by reconfiguration. The conformance of the equipment =
the deployer already bought makes it possible to solve the problem =
without buying new equipment.
>=20

Thanks for clarifying.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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Hi Paul,

>>=20
>> Big payloads are often associated with presence but IIRC there is a =
vendor who could not use UDP due to big payloads and it was just using =
SDP. An imaginary example: a video streaming service built with SIP: =
several video streams for different video qualities and profiles, =
several audio streams for the different languages and several chat =
streams for the subtitles in different languages. That could get big :-)
>=20
> The usual argument, which you seem to be making again, is that some =
vendor decides it only needs to support UDP because it doesn't =
anticipate being deployed in any situations where messages will exceed =
the PDU size. And the usual counter-argument is that it is nearly =
impossible to guarantee that this will always be the case, and hence the =
implementation MUST be prepared to use TCP.
>=20

I was trying to expose an example of the opposite :-) This vendor had to =
disable UDP support because their SDPs were really huge.

> Here we are arguing the flip side - someone arguing that it NEEDS TCP =
in all its deployments because it will have *some* big messages in every =
deployment, and hence sees no reason to implement UDP. I'm just arguing =
that some deployer might have a usage in mind where UDP *would* work all =
the time, and hence decide to deploy with only UDP configured.
>=20
> Its a weak argument. But I can imagine someone making it. E.g. =
somebody might do so in order to optimize a benchmark.
>=20

Sure, UDP-only deployments might be possible.

It seems like allowing for implementors to disable transports completely =
(or not even implement them) could open the can of worms, but at the =
same time one could find that limiting in certain environments. I don't =
even have a clear opinion on what would be best here :-S


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From pkyzivat@alum.mit.edu  Mon Apr 23 08:19:31 2012
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Cc: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, sipcore@ietf.org
Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/23/12 11:10 AM, Saśl Ibarra Corretgé wrote:
> Hi Paul,
>
>>>
>>> Big payloads are often associated with presence but IIRC there is a vendor who could not use UDP due to big payloads and it was just using SDP. An imaginary example: a video streaming service built with SIP: several video streams for different video qualities and profiles, several audio streams for the different languages and several chat streams for the subtitles in different languages. That could get big :-)
>>
>> The usual argument, which you seem to be making again, is that some vendor decides it only needs to support UDP because it doesn't anticipate being deployed in any situations where messages will exceed the PDU size. And the usual counter-argument is that it is nearly impossible to guarantee that this will always be the case, and hence the implementation MUST be prepared to use TCP.
>>
>
> I was trying to expose an example of the opposite :-) This vendor had to disable UDP support because their SDPs were really huge.

It should not be necessary to disable UDP because messages sometimes get 
big. As long as both are supported the proper one can be used based on 
message size. Of course there are some cases where that won't work - 
e.g. an offerless invite sent via UDP, leading to the need to send an 
offer in a reply.

I suppose that in practice restricting use to TCP would avoid a number 
of nasty issues.

>> Here we are arguing the flip side - someone arguing that it NEEDS TCP in all its deployments because it will have *some* big messages in every deployment, and hence sees no reason to implement UDP. I'm just arguing that some deployer might have a usage in mind where UDP *would* work all the time, and hence decide to deploy with only UDP configured.
>>
>> Its a weak argument. But I can imagine someone making it. E.g. somebody might do so in order to optimize a benchmark.
>>
>
> Sure, UDP-only deployments might be possible.
>
> It seems like allowing for implementors to disable transports completely (or not even implement them) could open the can of worms, but at the same time one could find that limiting in certain environments. I don't even have a clear opinion on what would be best here :-S

That is why we are discussing it. :)

	Thanks,
	Paul

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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 04/23/2012 10:19 AM, Paul Kyzivat wrote:
> On 4/23/12 11:10 AM, Sa=FAl Ibarra Corretg=E9 wrote:
>> Hi Paul,
>>
>>>>
>>>> Big payloads are often associated with presence but IIRC there is a
>>>> vendor who could not use UDP due to big payloads and it was just
>>>> using SDP. An imaginary example: a video streaming service built
>>>> with SIP: several video streams for different video qualities and
>>>> profiles, several audio streams for the different languages and
>>>> several chat streams for the subtitles in different languages. That
>>>> could get big :-)
>>>
>>> The usual argument, which you seem to be making again, is that some
>>> vendor decides it only needs to support UDP because it doesn't
>>> anticipate being deployed in any situations where messages will
>>> exceed the PDU size. And the usual counter-argument is that it is
>>> nearly impossible to guarantee that this will always be the case, and
>>> hence the implementation MUST be prepared to use TCP.
>>>
>>
>> I was trying to expose an example of the opposite :-) This vendor had
>> to disable UDP support because their SDPs were really huge.
>
> It should not be necessary to disable UDP because messages sometimes ge=
t
> big. As long as both are supported the proper one can be used based on
> message size. Of course there are some cases where that won't work -
> e.g. an offerless invite sent via UDP, leading to the need to send an
> offer in a reply.

Wasn't that the purpose of RFC 3261 mandating support for both though?=20
The same could happen in a SUBSCRIBE dialog; the SUBSCRIBE request and=20
200 OK reply could be sent over UDP, but the first NOTIFY may have to be=20
sent over TCP due to its size.

I think I'm in agreement with your 'narrow' approach, because it allows=20
the relaxation to be applied only in cases where an alternate 'large=20
message' transport (other than TCP or TLS/TCP) is being made available.=20
Granted, Asterisk survived for many years only supporting UDP, and it=20
was rare that a user ran into an problem due to message sizes, but with=20
SDPs growing (ICE support is on the way) and complex presence data being=20
transferred, it's clear that some sort of large message transport needs=20
to be supported too.

--=20
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpflemin=
g
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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2012/4/23 Paul Kyzivat <pkyzivat@alum.mit.edu>:
> It should not be necessary to disable UDP because messages sometimes get
> big. As long as both are supported the proper one can be used based on
> message size. Of course there are some cases where that won't work - e.g.=
 an
> offerless invite sent via UDP, leading to the need to send an offer in a
> reply.

Hi Paul, that sounds a bit "theorical" for me. SIP phones usually
perform a single registration using a single transport and binding a
single Contact URI with same transport than the one used for sending
the REGISTER.

Indeed a SIP UA could register two URIs (one with ;transport=3Dudp and
the other one with tcp), but then it would generate useless parallel
forking for initial incoming-requests (and lot of merged requests and
482 responses from the UA). Well, the UA could set different "q"
values and give more priority to the UDP binding, so the registrar
would just use TCP if sending the request over UDP fails (internal
503). However that would not work for in-dialog requests since the
RURI already indicates the transport.

IMHO this would be more feasible in the case of proxy to proxy
communication, in which proxy A could decide (based on request size)
whether to select one or other transport regardless the priority of
the NAPTR records, or selecting TCP regarless the RURI has an IP and
no ;transport=3Dtcp param.

But honestly, in the UA side I don't expect that dynamic selection of
the transport could work in "real" scenarios (maybe I'm wrong).

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Mon, 23 Apr 2012 22:02:24 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] comments on draft-ietf-sipcore-rfc3265bis-08.txt
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Paul,

I understand, I was late into the party... :(

The comment is more of an OPTIMIZATION issue. As a minimal change, can we
add a normative text which provide some pointers for implementers to
differentiate the re-SUBSCRIBE (to extend the subscription duration)
and re-SUBSCRIBE (to fetch the complete state, due to lost NOTIFY).

This might be of great concern for lossy networks (for example, wireless
networks, ...), where all the subsequent refresh SUBSCRIBE requests would
un-necessarily eats up more bandwidth for fetching the complete state which
is not necessary. For non-lossy network this is of minor issue...

Even if we decide to handle this optimization issue, we have to analyze
thoroughly how to differentiate differentiate these requests, through some
new header / header parameter / any other way. Need more analysis...

You can take a call, based on the feasibility. Even if dropped it is
fine....

Thanks,
Nataraju A B

On Mon, Apr 23, 2012 at 7:32 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Nataraju,
>
> Your comments would have been much more helpful if received some months
> ago when the document went through WGLC. This document is now *very* far
> down the process to publication, having already gone through IESG last
> call, IANA review, etc. and is now in final IESG Evaluation. Changes at
> this point are very disruptive.
>
> If you feel your suggestions are essential then you will have to go to
> extremes to pull this document back. I don't see anything below that
> convinces me to do that. AFAICT the comments below are all minor editorial
> except for the first, which suggests changing the mechanism.
>
>        Thanks,
>        Paul (as co-chair)
>
>
> On 4/23/12 12:58 AM, Nataraju A.B wrote:
>
>> All,
>>
>> Some logical comments on the draft-ietf-sipcore-rfc3265bis-**08.txt.
>>
>> ------------------------------**-----------------------
>> 5.3.2.  State Deltas
>>
>> CurrentText:
>>
>>    If a NOTIFY request arrives that has a version number that is
>>    incremented by more than one, the subscriber knows that a state delta
>>    has been missed; it ignores the NOTIFY request containing the state
>>    delta (except for the version number, which it retains to detect
>>    message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
>>    request containing a complete state snapshot.
>>
>> Comment:
>>    From this text, it infers that if any state delta lost, then subscriber
>>    sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
>> request inturn lead
>>    to fetching the complete state information.
>>
>>    It may be a good diea to differentiate the re-SUBSCRIBE triggered by
>> different scenarios.
>>    a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
>> duration. In this
>>       case it is not necessary to fetch the complete state information
>>    b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec
>> 5.3.2.
>>       VERSION number in received NOTIFY is increased by greater than ONE).
>>       In this case subscriber wants to fetch the complete state
>> information.
>>
>>    If we don't differentiate these two cases, then natural re-SUBSCRIBE
>> request
>>    unnecessarily fetch the complete state information, which may not be
>> necessary / expected.
>>    But uses higher network bandwidth to fetch the complete state
>> information.
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 4.4.1.  Dialog Creation and Termination
>>
>> CurrentText:
>>
>>       A subscriber may send a SUBSCRIBE request with an "Expires" header
>>       field of 0 in order to trigger the sending of such a NOTIFY
>>       request; however, for the purposes of subscription and dialog
>>       lifetime, the subscription is not considered terminated until the
>>       NOTIFY transaction with a "Subscription-State" of "terminated"
>>       completes.
>>
>> Comment:
>> We may have to mention how long the dialog must be kept alive, before
>> being cleaned up. Should we delete the dialog immediately after 200-SUB
>> or wait till NOTIFY with "Subscription-State"="**terminated". If we
>> always
>> wait till Last Notify for dialog termination, then 200-SUB don't have any
>> impact on the dialog termination. Atleast some normative text is of
>> necessity, for better understanding of the topic.
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 3.1.1.  Subscription Duration
>>
>> CurrentText:
>>
>>    SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
>>    in [RFC3261]).  This expires value indicates the duration of the
>>    subscription.  In order to keep subscriptions effective beyond the
>>    duration communicated in the "Expires" header field, subscribers need
>>    to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>>    request on the same dialog as defined in [RFC3261].
>>
>> Comment:
>> We can add a normative text saying, the refresh time will be
>> normally 50% to 75% of subscription duration.
>>
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 4.1.2.1.  Requesting a Subscription
>>
>> CurrentText:
>>    [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
>>    an "Accept" header field, but that field does not indicate a media
>>    type that the notifier is capable of generating in its NOTIFY
>>    requests, then the proper error response is 406 (Not Acceptable).
>>
>> COMMENT:
>> It is better to mention it as a normative text (indented).
>>
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 4.1.2.  Creating and Maintaining Subscriptions
>>
>>
>> CurrentText:
>>
>>    From the subscriber's perspective, a subscription proceeds according
>>    to the following state diagram:
>>
>> Comment:
>> Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
>> received in notify_wait state (because very often this is possible).
>> If no change in state of FSM, then some normative text shall convey the
>> same.
>>         It may be no-operation, but better to mention the same.
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 3.1.1.  Subscription Duration
>>
>> CurrentText:
>>       In addition to being a request to unsubscribe, a SUBSCRIBE request
>>       with "Expires" of 0 also causes a fetch of state; see
>>       Section 4.4.3.
>>
>> Comment:
>>      The above text has been mentioned to be normative text. But I feel
>> this
>>      shall be written as a regualr text in the draft/rfc. because this
>> is the
>>      expected behavior from the notifier.
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 3.2.1.
>>
>> CurrentText:
>>    This body will contain either the state of the subscribed resource or
>>    a pointer to such state in the form of a URI (see Section 5.4.13).
>>
>> ReplacementText/Comment:
>>    This body **SHALL** contain either the state of the subscribed
>> resource or
>>    a pointer to such state in the form of a **URL** (see Section 5.4.13).
>> ------------------------------**-----------------------
>> ------------------------------**-----------------------
>> 4.2.1.  Subscription Establishment and Maintenance
>>
>> CurrentText:
>>       inability to signal the success of a REFER request while signaling
>>       a problem with the subscription; and difficulty performing one
>>       action without the other.  Additionally, the proper exchange of
>>
>> ReplacementText:
>>       inability to signal the success of a REFER request while signaling
>>       a problem with the subscription; and difficulty **in** performing
>> one
>>       action without the other.  Additionally, the proper exchange of
>>
>> ------------------------------**-----------------------
>>
>> Thanks,
>> Nataraju A B
>>
>> ---------- Forwarded message ----------
>> From: *Nataraju A.B* <nataraju.sip@gmail.com
>> <mailto:nataraju.sip@gmail.com**>>
>> Date: Fri, Apr 13, 2012 at 10:53 PM
>> Subject: draft-ietf-sipcore-rfc3265bis-**08.txt - Review comments
>> (Nataraju A B)
>> To: Adam Roach <adam@nostrum.com <mailto:adam@nostrum.com>>,
>> sipcore@ietf.org <mailto:sipcore@ietf.org>
>>
>>
>> Adam,
>>
>> Please find some editorial comments on the draft
>> "draft draft-ietf-sipcore-rfc3265bis-**08.txt"
>>
>> For simplicity, I have mentioned both the current text and proposed text.
>>
>> The comments / corrective text can be found with prefix and suffix  --- **
>>
>> ------------------------------**-----------------------
>> 1.1.  Overview of Operation
>> CurrentText:
>>    The general concept is that entities in the network can subscribe to
>>    resource or call state for various resources or calls in the network,
>>    and those entities (or entities acting on their behalf) can send
>>    notifications when those states change.
>>
>> ReplacementText:
>>    The general concept is that entities in the network can subscribe to
>>    resource or call state **at / of** various resources or calls in the
>> network,
>>    and those entities (or entities acting on their behalf) **who** can
>> send
>>    notifications when those states change.
>> ------------------------------**-----------------------
>>
>> 1.1.  Overview of Operation
>>
>> CurrentText:
>>    Subscriptions are expired and must be refreshed by subsequent
>>    SUBSCRIBE requests.
>>
>> ReplacementText:
>>    Subscriptions are expired and must be refreshed by subsequent
>>    SUBSCRIBE requests **before expiration**.
>> ------------------------------**-----------------------
>>
>> 2.  Definitions
>>
>> CurrentText:
>>    Subscriber:  A subscriber is a user agent which receives NOTIFY
>>       requests from notifiers; these NOTIFY requests contain information
>>       about the state of a resource in which the subscriber is
>>       interested.  Subscribers typically also generate SUBSCRIBE
>>       requests and send them to notifiers to create subscriptions.
>>
>> Comment:
>> After this information, we can add a normative text which indicates
>> when a SIP entity might get NOTIFY messages without actually
>> sending SUBSCRIBE request. For example, REFER request might end up with
>> implicit subscription. Also mention any other requests which may lead to
>> implicit subscriptions (other than REFER).
>> ------------------------------**-----------------------
>> 2.  Definitions
>>
>> CurrentText:
>>    Subscription:  A subscription is a set of application state
>>       associated with a dialog.  This application state includes a
>>       pointer to the associated dialog, the event package name, and
>>       possibly an identification token.  Event packages will define
>>       additional subscription state information.  By definition,
>>       subscriptions exist in both a subscriber and a notifier.
>>
>> ReplacementText:
>>    Subscription:  A subscription is a set of application **states**
>>       associated with a dialog.  **The** application state includes a
>>       pointer to the associated dialog, the event package name, and
>>       possibly an identification token.  Event packages will define
>>       additional subscription state information.  By definition,
>>       subscriptions exist in both a subscriber and a notifier.
>> ------------------------------**-----------------------
>> 3.1.1.  Subscription Duration
>>
>> CurrentText:
>>
>>    subscription.  In order to keep subscriptions effective beyond the
>>    duration communicated in the "Expires" header field, subscribers need
>>    to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>>    request on the same dialog as defined in [RFC3261].
>>
>> ReplacementText
>> 3.1.1.  Subscription Duration
>>
>>    subscription.  In order to keep subscriptions effective beyond the
>>    duration communicated in the "Expires" header field, subscribers need
>>    to refresh subscriptions **before expiration** on a periodic basis
>> using a new SUBSCRIBE
>>    request on the same dialog as defined in [RFC3261].
>> ------------------------------**-----------------------
>> 3.1.1.  Subscription Duration
>>
>> CurrentText:
>>       In addition to being a request to unsubscribe, a SUBSCRIBE request
>>       with "Expires" of 0 also causes a fetch of state; see
>>       Section 4.4.3.
>>
>> Comment:
>>      The above text has been mentioned to be normative text. But I feel
>> this
>>      shall be written as a regualr text in the draft/rfc. because this
>> is the
>>      expected behavior from the notifier.
>> ------------------------------**-----------------------
>> 3.1.2.  Identification of Subscribed Events and Event Classes
>>
>> CurrentText:
>>    Subscribers MUST include exactly one "Event" header field in
>>    SUBSCRIBE requests, indicating to which event or class of events they
>>    are subscribing.  The "Event" header field will contain a token which
>>    indicates the type of state for which a subscription is being
>>    requested.  This token will be registered with the IANA and will
>>    correspond to an event package which further describes the semantics
>>    of the event or event class.
>>
>>
>> ReplacementText:
>>    Subscribers MUST include exactly one "Event" header field in
>>    SUBSCRIBE requests, indicating to which event or class of events they
>>    are subscribing.  The "Event" header field **MUST** contain a token
>> which
>>    indicates the type of state for which a subscription is being
>>    requested.  This token will be registered with the IANA and will
>>    correspond to an event package which further describes the semantics
>>    of the event or event class.
>> ------------------------------**-----------------------
>> 3.1.  SUBSCRIBE
>>
>> CurrentText:
>>    The SUBSCRIBE method is used to request current state and state
>>    updates from a remote node.  SUBSCRIBE requests are target refresh
>>
>>
>> ReplacementText:
>>    The SUBSCRIBE method is used to request current state and
>> **subsequent** state
>>    updates from a remote node.  SUBSCRIBE requests are target refresh
>>
>>
>> ------------------------------**-----------------------
>> 3.1.1.  Subscription Duration
>>
>> CurrentText:
>>
>>    SUBSCRIBE requests SHOULD contain an "Expires" header field (defined
>>    in [RFC3261]).  This expires value indicates the duration of the
>>    subscription.  In order to keep subscriptions effective beyond the
>>    duration communicated in the "Expires" header field, subscribers need
>>    to refresh subscriptions on a periodic basis using a new SUBSCRIBE
>>    request on the same dialog as defined in [RFC3261].
>>
>> Comment:
>> We can add a normative text saying, the refresh time will be
>> normally 50% to 75% of subscription duration.
>>
>> ------------------------------**-----------------------
>> 3.2.1.
>>
>> CurrentText:
>>    This body will contain either the state of the subscribed resource or
>>    a pointer to such state in the form of a URI (see Section 5.4.13).
>>
>> ReplacementText/Comment:
>>    This body **SHALL** contain either the state of the subscribed
>> resource or
>>    a pointer to such state in the form of a **URL** (see Section 5.4.13).
>> ------------------------------**-----------------------
>> 4.1.2.  Creating and Maintaining Subscriptions
>>
>>
>> CurrentText:
>>
>>    From the subscriber's perspective, a subscription proceeds according
>>    to the following state diagram:
>>
>> Comment:
>> Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
>> received in notify_wait state (because very often this is possible).
>> If no change in state of FSM, then some normative text shall convey the
>> same.
>>         It may be no-operation, but better to mention the same.
>> ------------------------------**-----------------------
>> 4.1.2.1.  Requesting a Subscription
>>
>> CurrentText:
>>    When a subscriber wishes to subscribe to a particular state for a
>>    resource, it forms a SUBSCRIBE request.  If the initial SUBSCRIBE
>>
>> ReplacementText:
>>    When a subscriber wishes to subscribe to a particular state **of** a
>>    resource, it forms a SUBSCRIBE request **according to sec 3.1 and
>> send it across to notifier**.  If the initial SUBSCRIBE
>>
>> ------------------------------**-----------------------
>> 4.1.2.1.  Requesting a Subscription
>>
>> CurrentText:
>>    [RFC3261].  For the sake of clarity: if a SUBSCRIBE request contains
>>    an "Accept" header field, but that field does not indicate a media
>>    type that the notifier is capable of generating in its NOTIFY
>>    requests, then the proper error response is 406 (Not Acceptable).
>>
>> COMMENT:
>> It is better to mention it as a normative text (indented).
>>
>> ------------------------------**-----------------------
>> 4.1.3.  Receiving and Processing State Information
>>
>> CurrentText:
>>
>>    If the "Subscription-State" value is "terminated", the subscriber
>>    MUST consider the subscription terminated.  The "expires" parameter
>>    has no semantics for "terminated" -- notifiers SHOULD NOT include an
>>
>> ReplacementText:
>>
>>    If the "Subscription-State" value is "terminated", the subscriber
>>    MUST consider the subscription terminated.  The "expires" parameter
>>    has no semantics for "terminated" **state**. Notifiers SHOULD NOT
>> include an
>> ------------------------------**-----------------------
>>
>> CurrentText:
>>    deactivated:  The subscription has been terminated, but the
>>       subscriber SHOULD retry immediately with a new subscription.  One
>>
>> ReplacementText:
>>    deactivated:  The subscription has been terminated, but the
>>       subscriber SHOULD retry with a new subscription.  One
>>
>>    Is It mandatory to resubscribe immediately. if not remove the word
>> "immediately".
>>    If it is delayed, what are the consequences ????
>>
>> ------------------------------**-----------------------
>> 4.2.1.  Subscription Establishment and Maintenance
>>
>> CurrentText:
>>       inability to signal the success of a REFER request while signaling
>>       a problem with the subscription; and difficulty performing one
>>       action without the other.  Additionally, the proper exchange of
>>
>> ReplacementText:
>>       inability to signal the success of a REFER request while signaling
>>       a problem with the subscription; and difficulty **in** performing
>> one
>>       action without the other.  Additionally, the proper exchange of
>>
>> ------------------------------**-----------------------
>> 4.2.1.4.  Refreshing of Subscriptions
>>
>> CurrentText:
>>    If no refresh for a notification address is received before its
>>    expiration time, the subscription is removed.  When removing a
>>
>> ReplacementText:
>>    If no refresh for a notification is received before its
>>    expiration time, the subscription is removed.  When removing a
>> Comment:Word "address" needs to be removed.
>> ------------------------------**-----------------------
>> 4.3.  Proxy Behavior
>>
>> CurrentText:
>>    Proxies that did not add a Record-Route header field to the initial
>>    SUBSCRIBE request MUST NOT add a Record-Route header field to any of
>>    the associated NOTIFY requests.
>>
>> ReplacementText:
>>    Proxies that did not add a Record-Route header field to the initial
>>    SUBSCRIBE request MUST NOT add a Record-Route header field to any of
>>    the associated NOTIFY requests **or any SUBSCRIBE refresh requests**.
>> ------------------------------**-----------------------
>> 4.4.1.  Dialog Creation and Termination
>>
>> CurrentText:
>>
>>       A subscriber may send a SUBSCRIBE request with an "Expires" header
>>       field of 0 in order to trigger the sending of such a NOTIFY
>>       request; however, for the purposes of subscription and dialog
>>       lifetime, the subscription is not considered terminated until the
>>       NOTIFY transaction with a "Subscription-State" of "terminated"
>>       completes.
>>
>> Comment:
>> We may have to mention how long the dialog must be kept alive, before
>> being cleaned up. Do we delete the dialog immediately after 200-SUB
>> or wait till NOTIFY with "Subscription-State"="**terminated". If we
>> always
>> wait till Last Notify for dialog termination, then 200-SUB don't have any
>> impact on the dialog termination. Atleast some normative text is of
>> necessity, for better understanding of the topic.
>> ------------------------------**-----------------------
>> 4.4.3.  Polling Resource State
>>
>> CurrentText:
>>
>>    Note that the NOTIFY requests triggered by SUBSCRIBE requests with
>> "Expires" header fields of 0 will contain a "Subscription-State"
>>    value of "terminated", and a "reason" parameter of "timeout".
>>
>> ReplacementText:
>>    Note that the NOTIFY requests triggered by SUBSCRIBE requests with
>> "Expires" header fields of 0 **shall** contain a "Subscription-State"
>>    value of "terminated", and a "reason" parameter of "timeout".
>> ------------------------------**-----------------------
>> 4.5.2.  Sharing Dialogs
>>
>> CurrentText:
>>       for a dialog (e.g., route set handling).  In particular, multiple
>>       subscriptions within a dialog are expire independently, and
>>       require independent subscription refreshes.
>>
>> ReplacementText:
>>       for a dialog (e.g., route set handling).  In particular, multiple
>>       subscriptions within a dialog **operates** independently, and
>>       require independent subscription refreshes.
>> ------------------------------**-----------------------
>> 4.5.2.  Sharing Dialogs
>>
>> CurrentText:
>>    o  Subscribers MAY include an "id" parameter in SUBSCRIBE request
>> "Event" header field to allow differentiation between multiple
>>
>> ReplacementText:
>>    o  Subscribers MAY include an "id" parameter in SUBSCRIBE **request's**
>> "Event" header field to allow differentiation between multiple
>> OR
>>    o  Subscribers MAY include an "id" parameter in **"Event" header field
>>       of SUBSCRIBE request** to allow differentiation between multiple
>> ------------------------------**-----------------------
>> 4.5.2.  Sharing Dialogs
>>
>> CurrentText:
>>    o  When a subscriber refreshes a the subscription timer, the
>>       SUBSCRIBE request MUST contain the same "Event" header field "id"
>>       parameter as was present in the SUBSCRIBE request that created the
>>       subscription.  (Otherwise, the notifier will interpret the
>>
>> ReplacementText:
>>    o  When a subscriber refreshes a the subscription timer, the
>>       SUBSCRIBE request MUST contain the same **"id" parameter in "Event"
>>       header field, which** was present in the SUBSCRIBE request that
>> created the
>>       subscription.  (Otherwise, the notifier will interpret the
>> ------------------------------**-----------------------
>> 4.5.2.  Sharing Dialogs
>>
>> CurrentText:
>>
>>    o  When a subscription is created in the notifier, it stores any
>> "Event" header field "id" parameter as part of the subscription
>>       information (along with the event package name).
>>
>> ReplacementText:
>>    o  When a subscription is created in the notifier, it stores **the**
>>       **"id" parameter in "Event" header field** as part of the
>> subscription
>>       information (along with the event package name).
>> ------------------------------**-----------------------
>> 4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>>
>> CurrentText:
>>
>>    Neither SUBSCRIBE nor NOTIFY requests can be canceled.  If a UAS
>>    receives a CANCEL request that matches a known SUBSCRIBE or NOTIFY
>>    transaction, it MUST respond to the CANCEL request, but otherwise
>>    ignore it.  In particular, the CANCEL request MUST NOT affect
>>    processing of the SUBSCRIBE or NOTIFY request in any way.
>>
>> ReplacementText/Comment:
>> What should be the response code for the CANCEL's reply ???
>> Preferably 200-OK shall be used.
>> ------------------------------**-----------------------
>> 4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>>
>> CurrentText:
>>
>>    UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>>    transactions.
>>
>> ReplacementText:
>>    UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>>    transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY
>>    or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**
>> ------------------------------**-----------------------
>> 5.3.2.  State Deltas
>>
>> CurrentText:
>>
>>    In the case that the state information to be conveyed is large, the
>>    event package may choose to detail a scheme by which NOTIFY requests
>>    contain state deltas instead of complete state.
>>
>>
>> ReplacementText:
>>    In the case that the state information to be conveyed is large, the
>>    event package may choose a scheme by which NOTIFY requests
>>    contain state deltas instead of complete state.
>>
>> comment:deleted words "to detail" in 2nd line.
>>
>> ------------------------------**-----------------------
>> 5.3.2.  State Deltas
>>
>> CurrentText:
>>
>>    If a NOTIFY request arrives that has a version number that is
>>    incremented by more than one, the subscriber knows that a state delta
>>    has been missed; it ignores the NOTIFY request containing the state
>>    delta (except for the version number, which it retains to detect
>>    message loss), and re-sends a SUBSCRIBE request to force a NOTIFY
>>    request containing a complete state snapshot.
>>
>> Comment:
>>    From this text, it infers that if any state delta lost, then subscriber
>>    sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE
>> request inturn lead
>>    to fetching the complete state information.
>>
>>    It may be a good diea to differentiate the re-SUBSCRIBE triggered by
>> different scenarios.
>>    a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
>> duration. In this
>>       case it is *not *necessary to fetch the complete state information
>>
>>    b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec
>> 5.3.2.
>>       VERSION number in received NOTIFY is increased by greater than ONE).
>>       In this case subscriber wants to fetch the complete state
>> information.
>>
>>    If we don't differentiate these two cases, then natural re-SUBSCRIBE
>> request
>>    unnecessarily fetch the complete state information, which may not be
>> necessary / expected.
>>    But uses higher network bandwidth to fetch the complete state
>> information.
>> ------------------------------**-----------------------
>> 5.4.7.  Notifier generation of NOTIFY requests
>>
>> CurrentText:
>>    This section may optionally describe the behavior used to process the
>>    subsequent response.
>>
>> Comment:
>>    In this section 5.4.7, the above statement is not clear, what is
>> really meant to be.
>>
>> ------------------------------**-----------------------
>> 5.4.13.  Use of URIs to Retrieve State
>>
>> CurrentText:
>>
>> ReplacementText:
>>    In this context, I feel Use of "URLs" is more appropriate than "URIs".
>> ------------------------------**-----------------------
>>
>>
>> Thanks,
>> Nataraju A.B.
>> E-ID : nataraju.ab@gmail.com <mailto:nataraju.ab@gmail.com> /
>> nataraju.sip@gmail.com <mailto:nataraju.sip@gmail.com**>
>>
>> +91-98455-95744
>>
>>
>>
>> --
>> Thanks,
>> Nataraju A.B.
>>
>>
>>
>> ______________________________**_________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/**listinfo/sipcore<https://www.ietf.org/mailman/listinfo/sipcore>
>>
>
> ______________________________**_________________
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> sipcore@ietf.org
> https://www.ietf.org/mailman/**listinfo/sipcore<https://www.ietf.org/mailman/listinfo/sipcore>
>



-- 
Thanks,
Nataraju A.B.

--f46d04016a77e086d604be5b2e0b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_extra">Paul,=A0</div><div class=3D"gmail_extra"><br></d=
iv><div class=3D"gmail_extra">I understand, I was late into the party... :(=
</div><div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">The c=
omment is more of an OPTIMIZATION issue. As a minimal change,=A0can=A0we ad=
d a normative text which provide some pointers for implementers to differen=
tiate the re-SUBSCRIBE (to extend the subscription duration) and=A0re-SUBSC=
RIBE (to fetch the complete state, due to lost NOTIFY).=A0</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">This might =
be of great concern for lossy networks (for example, wireless networks, ...=
), where all the subsequent refresh SUBSCRIBE requests would un-necessarily=
 eats up more bandwidth for fetching the complete state which is not necess=
ary. For non-lossy network this is of minor issue...</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Even if we =
decide to handle this optimization issue, we have to analyze thoroughly how=
 to differentiate differentiate these requests, through some new header / h=
eader parameter / any other way. Need more analysis...</div>

<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">You can tak=
e a call, based on the feasibility. Even if dropped it is fine....</div><di=
v class=3D"gmail_extra"><br></div><div class=3D"gmail_extra">Thanks,</div><=
div class=3D"gmail_extra">
Nataraju A B</div>
<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Mon, Apr 23, 2=
012 at 7:32 PM, Paul Kyzivat <span dir=3D"ltr">&lt;<a href=3D"mailto:pkyziv=
at@alum.mit.edu" target=3D"_blank">pkyzivat@alum.mit.edu</a>&gt;</span> wro=
te:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Nataraju,<br>
<br>
Your comments would have been much more helpful if received some months ago=
 when the document went through WGLC. This document is now *very* far down =
the process to publication, having already gone through IESG last call, IAN=
A review, etc. and is now in final IESG Evaluation. Changes at this point a=
re very disruptive.<br>


<br>
If you feel your suggestions are essential then you will have to go to extr=
emes to pull this document back. I don&#39;t see anything below that convin=
ces me to do that. AFAICT the comments below are all minor editorial except=
 for the first, which suggests changing the mechanism.<br>


<br>
 =A0 =A0 =A0 =A0Thanks,<br>
 =A0 =A0 =A0 =A0Paul (as co-chair)<div><div><br>
<br>
On 4/23/12 12:58 AM, Nataraju A.B wrote:<br>
</div></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div><div>
All,<br>
<br>
Some logical comments on the draft-ietf-sipcore-rfc3265bis-<u></u>08.txt.<b=
r>
<br>
------------------------------<u></u>-----------------------<br>
5.3.2. =A0State Deltas<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0If a NOTIFY request arrives that has a version number that is<br>
 =A0 =A0incremented by more than one, the subscriber knows that a state del=
ta<br>
 =A0 =A0has been missed; it ignores the NOTIFY request containing the state=
<br>
 =A0 =A0delta (except for the version number, which it retains to detect<br=
>
 =A0 =A0message loss), and re-sends a SUBSCRIBE request to force a NOTIFY<b=
r>
 =A0 =A0request containing a complete state snapshot.<br>
<br>
Comment:<br>
 =A0 =A0From this text, it infers that if any state delta lost, then subscr=
iber<br>
 =A0 =A0sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE<b=
r>
request inturn lead<br>
 =A0 =A0to fetching the complete state information.<br>
<br>
 =A0 =A0It may be a good diea to differentiate the re-SUBSCRIBE triggered b=
y<br>
different scenarios.<br>
 =A0 =A0a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE<br>
duration. In this<br>
 =A0 =A0 =A0 case it is not necessary to fetch the complete state informati=
on<br>
 =A0 =A0b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec<b=
r>
5.3.2.<br>
 =A0 =A0 =A0 VERSION number in received NOTIFY is increased by greater than=
 ONE).<br>
 =A0 =A0 =A0 In this case subscriber wants to fetch the complete state<br>
information.<br>
<br>
 =A0 =A0If we don&#39;t differentiate these two cases, then natural re-SUBS=
CRIBE<br>
request<br>
 =A0 =A0unnecessarily fetch the complete state information, which may not b=
e<br>
necessary / expected.<br>
 =A0 =A0But uses higher network bandwidth to fetch the complete state<br>
information.<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
4.4.1. =A0Dialog Creation and Termination<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0 =A0 A subscriber may send a SUBSCRIBE request with an &quot;Expire=
s&quot; header<br>
 =A0 =A0 =A0 field of 0 in order to trigger the sending of such a NOTIFY<br=
>
 =A0 =A0 =A0 request; however, for the purposes of subscription and dialog<=
br>
 =A0 =A0 =A0 lifetime, the subscription is not considered terminated until =
the<br>
 =A0 =A0 =A0 NOTIFY transaction with a &quot;Subscription-State&quot; of &q=
uot;terminated&quot;<br>
 =A0 =A0 =A0 completes.<br>
<br>
Comment:<br>
We may have to mention how long the dialog must be kept alive, before<br>
being cleaned up. Should we delete the dialog immediately after 200-SUB<br>
or wait till NOTIFY with &quot;Subscription-State&quot;=3D&quot;<u></u>term=
inated&quot;. If we always<br>
wait till Last Notify for dialog termination, then 200-SUB don&#39;t have a=
ny<br>
impact on the dialog termination. Atleast some normative text is of<br>
necessity, for better understanding of the topic.<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
3.1.1. =A0Subscription Duration<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0SUBSCRIBE requests SHOULD contain an &quot;Expires&quot; header fie=
ld (defined<br>
 =A0 =A0in [RFC3261]). =A0This expires value indicates the duration of the<=
br>
 =A0 =A0subscription. =A0In order to keep subscriptions effective beyond th=
e<br>
 =A0 =A0duration communicated in the &quot;Expires&quot; header field, subs=
cribers need<br>
 =A0 =A0to refresh subscriptions on a periodic basis using a new SUBSCRIBE<=
br>
 =A0 =A0request on the same dialog as defined in [RFC3261].<br>
<br>
Comment:<br>
We can add a normative text saying, the refresh time will be<br>
normally 50% to 75% of subscription duration.<br>
<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
4.1.2.1. =A0Requesting a Subscription<br>
<br>
CurrentText:<br>
 =A0 =A0[RFC3261]. =A0For the sake of clarity: if a SUBSCRIBE request conta=
ins<br>
 =A0 =A0an &quot;Accept&quot; header field, but that field does not indicat=
e a media<br>
 =A0 =A0type that the notifier is capable of generating in its NOTIFY<br>
 =A0 =A0requests, then the proper error response is 406 (Not Acceptable).<b=
r>
<br>
COMMENT:<br>
It is better to mention it as a normative text (indented).<br>
<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
4.1.2. =A0Creating and Maintaining Subscriptions<br>
<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0From the subscriber&#39;s perspective, a subscription proceeds acco=
rding<br>
 =A0 =A0to the following state diagram:<br>
<br>
Comment:<br>
Here it is better to mention, what happens to FSM when 200-SUBSCRIBE<br>
received in notify_wait state (because very often this is possible).<br>
If no change in state of FSM, then some normative text shall convey the<br>
same.<br>
 =A0 =A0 =A0 =A0 It may be no-operation, but better to mention the same.<br=
>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
3.1.1. =A0Subscription Duration<br>
<br>
CurrentText:<br>
 =A0 =A0 =A0 In addition to being a request to unsubscribe, a SUBSCRIBE req=
uest<br>
 =A0 =A0 =A0 with &quot;Expires&quot; of 0 also causes a fetch of state; se=
e<br>
 =A0 =A0 =A0 Section 4.4.3.<br>
<br>
Comment:<br>
 =A0 =A0 =A0The above text has been mentioned to be normative text. But I f=
eel<br>
this<br>
 =A0 =A0 =A0shall be written as a regualr text in the draft/rfc. because th=
is<br>
is the<br>
 =A0 =A0 =A0expected behavior from the notifier.<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
3.2.1.<br>
<br>
CurrentText:<br>
 =A0 =A0This body will contain either the state of the subscribed resource =
or<br>
 =A0 =A0a pointer to such state in the form of a URI (see Section 5.4.13).<=
br>
<br>
ReplacementText/Comment:<br>
 =A0 =A0This body **SHALL** contain either the state of the subscribed<br>
resource or<br>
 =A0 =A0a pointer to such state in the form of a **URL** (see Section 5.4.1=
3).<br>
------------------------------<u></u>-----------------------<br>
------------------------------<u></u>-----------------------<br>
4.2.1. =A0Subscription Establishment and Maintenance<br>
<br>
CurrentText:<br>
 =A0 =A0 =A0 inability to signal the success of a REFER request while signa=
ling<br>
 =A0 =A0 =A0 a problem with the subscription; and difficulty performing one=
<br>
 =A0 =A0 =A0 action without the other. =A0Additionally, the proper exchange=
 of<br>
<br>
ReplacementText:<br>
 =A0 =A0 =A0 inability to signal the success of a REFER request while signa=
ling<br>
 =A0 =A0 =A0 a problem with the subscription; and difficulty **in** perform=
ing one<br>
 =A0 =A0 =A0 action without the other. =A0Additionally, the proper exchange=
 of<br>
<br>
------------------------------<u></u>-----------------------<br>
<br>
Thanks,<br>
Nataraju A B<br>
<br>
---------- Forwarded message ----------<br></div></div><div>
From: *Nataraju A.B* &lt;<a href=3D"mailto:nataraju.sip@gmail.com" target=
=3D"_blank">nataraju.sip@gmail.com</a><br>
&lt;mailto:<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">nata=
raju.sip@gmail.com</a><u></u>&gt;&gt;<br>
Date: Fri, Apr 13, 2012 at 10:53 PM<br>
Subject: draft-ietf-sipcore-rfc3265bis-<u></u>08.txt - Review comments<br>
(Nataraju A B)<br></div><div><div>
To: Adam Roach &lt;<a href=3D"mailto:adam@nostrum.com" target=3D"_blank">ad=
am@nostrum.com</a> &lt;mailto:<a href=3D"mailto:adam@nostrum.com" target=3D=
"_blank">adam@nostrum.com</a>&gt;&gt;,<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a> =
&lt;mailto:<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ie=
tf.org</a>&gt;<br>
<br>
<br>
Adam,<br>
<br>
Please find some editorial comments on the draft<br>
&quot;draft draft-ietf-sipcore-rfc3265bis-<u></u>08.txt&quot;<br>
<br>
For simplicity, I have mentioned both the current text and proposed text.<b=
r>
<br>
The comments / corrective text can be found with prefix and suffix =A0--- *=
*<br>
<br>
------------------------------<u></u>-----------------------<br>
1.1. =A0Overview of Operation<br>
CurrentText:<br>
 =A0 =A0The general concept is that entities in the network can subscribe t=
o<br>
 =A0 =A0resource or call state for various resources or calls in the networ=
k,<br>
 =A0 =A0and those entities (or entities acting on their behalf) can send<br=
>
 =A0 =A0notifications when those states change.<br>
<br>
ReplacementText:<br>
 =A0 =A0The general concept is that entities in the network can subscribe t=
o<br>
 =A0 =A0resource or call state **at / of** various resources or calls in th=
e<br>
network,<br>
 =A0 =A0and those entities (or entities acting on their behalf) **who** can=
 send<br>
 =A0 =A0notifications when those states change.<br>
------------------------------<u></u>-----------------------<br>
<br>
1.1. =A0Overview of Operation<br>
<br>
CurrentText:<br>
 =A0 =A0Subscriptions are expired and must be refreshed by subsequent<br>
 =A0 =A0SUBSCRIBE requests.<br>
<br>
ReplacementText:<br>
 =A0 =A0Subscriptions are expired and must be refreshed by subsequent<br>
 =A0 =A0SUBSCRIBE requests **before expiration**.<br>
------------------------------<u></u>-----------------------<br>
<br>
2. =A0Definitions<br>
<br>
CurrentText:<br>
 =A0 =A0Subscriber: =A0A subscriber is a user agent which receives NOTIFY<b=
r>
 =A0 =A0 =A0 requests from notifiers; these NOTIFY requests contain informa=
tion<br>
 =A0 =A0 =A0 about the state of a resource in which the subscriber is<br>
 =A0 =A0 =A0 interested. =A0Subscribers typically also generate SUBSCRIBE<b=
r>
 =A0 =A0 =A0 requests and send them to notifiers to create subscriptions.<b=
r>
<br>
Comment:<br>
After this information, we can add a normative text which indicates<br>
when a SIP entity might get NOTIFY messages without actually<br>
sending SUBSCRIBE request. For example, REFER request might end up with<br>
implicit subscription. Also mention any other requests which may lead to<br=
>
implicit subscriptions (other than REFER).<br>
------------------------------<u></u>-----------------------<br>
2. =A0Definitions<br>
<br>
CurrentText:<br>
 =A0 =A0Subscription: =A0A subscription is a set of application state<br>
 =A0 =A0 =A0 associated with a dialog. =A0This application state includes a=
<br>
 =A0 =A0 =A0 pointer to the associated dialog, the event package name, and<=
br>
 =A0 =A0 =A0 possibly an identification token. =A0Event packages will defin=
e<br>
 =A0 =A0 =A0 additional subscription state information. =A0By definition,<b=
r>
 =A0 =A0 =A0 subscriptions exist in both a subscriber and a notifier.<br>
<br>
ReplacementText:<br>
 =A0 =A0Subscription: =A0A subscription is a set of application **states**<=
br>
 =A0 =A0 =A0 associated with a dialog. =A0**The** application state include=
s a<br>
 =A0 =A0 =A0 pointer to the associated dialog, the event package name, and<=
br>
 =A0 =A0 =A0 possibly an identification token. =A0Event packages will defin=
e<br>
 =A0 =A0 =A0 additional subscription state information. =A0By definition,<b=
r>
 =A0 =A0 =A0 subscriptions exist in both a subscriber and a notifier.<br>
------------------------------<u></u>-----------------------<br>
3.1.1. =A0Subscription Duration<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0subscription. =A0In order to keep subscriptions effective beyond th=
e<br>
 =A0 =A0duration communicated in the &quot;Expires&quot; header field, subs=
cribers need<br>
 =A0 =A0to refresh subscriptions on a periodic basis using a new SUBSCRIBE<=
br>
 =A0 =A0request on the same dialog as defined in [RFC3261].<br>
<br>
ReplacementText<br>
3.1.1. =A0Subscription Duration<br>
<br>
 =A0 =A0subscription. =A0In order to keep subscriptions effective beyond th=
e<br>
 =A0 =A0duration communicated in the &quot;Expires&quot; header field, subs=
cribers need<br>
 =A0 =A0to refresh subscriptions **before expiration** on a periodic basis<=
br>
using a new SUBSCRIBE<br>
 =A0 =A0request on the same dialog as defined in [RFC3261].<br>
------------------------------<u></u>-----------------------<br>
3.1.1. =A0Subscription Duration<br>
<br>
CurrentText:<br>
 =A0 =A0 =A0 In addition to being a request to unsubscribe, a SUBSCRIBE req=
uest<br>
 =A0 =A0 =A0 with &quot;Expires&quot; of 0 also causes a fetch of state; se=
e<br>
 =A0 =A0 =A0 Section 4.4.3.<br>
<br>
Comment:<br>
 =A0 =A0 =A0The above text has been mentioned to be normative text. But I f=
eel<br>
this<br>
 =A0 =A0 =A0shall be written as a regualr text in the draft/rfc. because th=
is<br>
is the<br>
 =A0 =A0 =A0expected behavior from the notifier.<br>
------------------------------<u></u>-----------------------<br>
3.1.2. =A0Identification of Subscribed Events and Event Classes<br>
<br>
CurrentText:<br>
 =A0 =A0Subscribers MUST include exactly one &quot;Event&quot; header field=
 in<br>
 =A0 =A0SUBSCRIBE requests, indicating to which event or class of events th=
ey<br>
 =A0 =A0are subscribing. =A0The &quot;Event&quot; header field will contain=
 a token which<br>
 =A0 =A0indicates the type of state for which a subscription is being<br>
 =A0 =A0requested. =A0This token will be registered with the IANA and will<=
br>
 =A0 =A0correspond to an event package which further describes the semantic=
s<br>
 =A0 =A0of the event or event class.<br>
<br>
<br>
ReplacementText:<br>
 =A0 =A0Subscribers MUST include exactly one &quot;Event&quot; header field=
 in<br>
 =A0 =A0SUBSCRIBE requests, indicating to which event or class of events th=
ey<br>
 =A0 =A0are subscribing. =A0The &quot;Event&quot; header field **MUST** con=
tain a token<br>
which<br>
 =A0 =A0indicates the type of state for which a subscription is being<br>
 =A0 =A0requested. =A0This token will be registered with the IANA and will<=
br>
 =A0 =A0correspond to an event package which further describes the semantic=
s<br>
 =A0 =A0of the event or event class.<br>
------------------------------<u></u>-----------------------<br>
3.1. =A0SUBSCRIBE<br>
<br>
CurrentText:<br>
 =A0 =A0The SUBSCRIBE method is used to request current state and state<br>
 =A0 =A0updates from a remote node. =A0SUBSCRIBE requests are target refres=
h<br>
<br>
<br>
ReplacementText:<br>
 =A0 =A0The SUBSCRIBE method is used to request current state and<br>
**subsequent** state<br>
 =A0 =A0updates from a remote node. =A0SUBSCRIBE requests are target refres=
h<br>
<br>
<br>
------------------------------<u></u>-----------------------<br>
3.1.1. =A0Subscription Duration<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0SUBSCRIBE requests SHOULD contain an &quot;Expires&quot; header fie=
ld (defined<br>
 =A0 =A0in [RFC3261]). =A0This expires value indicates the duration of the<=
br>
 =A0 =A0subscription. =A0In order to keep subscriptions effective beyond th=
e<br>
 =A0 =A0duration communicated in the &quot;Expires&quot; header field, subs=
cribers need<br>
 =A0 =A0to refresh subscriptions on a periodic basis using a new SUBSCRIBE<=
br>
 =A0 =A0request on the same dialog as defined in [RFC3261].<br>
<br>
Comment:<br>
We can add a normative text saying, the refresh time will be<br>
normally 50% to 75% of subscription duration.<br>
<br>
------------------------------<u></u>-----------------------<br>
3.2.1.<br>
<br>
CurrentText:<br>
 =A0 =A0This body will contain either the state of the subscribed resource =
or<br>
 =A0 =A0a pointer to such state in the form of a URI (see Section 5.4.13).<=
br>
<br>
ReplacementText/Comment:<br>
 =A0 =A0This body **SHALL** contain either the state of the subscribed<br>
resource or<br>
 =A0 =A0a pointer to such state in the form of a **URL** (see Section 5.4.1=
3).<br>
------------------------------<u></u>-----------------------<br>
4.1.2. =A0Creating and Maintaining Subscriptions<br>
<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0From the subscriber&#39;s perspective, a subscription proceeds acco=
rding<br>
 =A0 =A0to the following state diagram:<br>
<br>
Comment:<br>
Here it is better to mention, what happens to FSM when 200-SUBSCRIBE<br>
received in notify_wait state (because very often this is possible).<br>
If no change in state of FSM, then some normative text shall convey the<br>
same.<br>
 =A0 =A0 =A0 =A0 It may be no-operation, but better to mention the same.<br=
>
------------------------------<u></u>-----------------------<br>
4.1.2.1. =A0Requesting a Subscription<br>
<br>
CurrentText:<br>
 =A0 =A0When a subscriber wishes to subscribe to a particular state for a<b=
r>
 =A0 =A0resource, it forms a SUBSCRIBE request. =A0If the initial SUBSCRIBE=
<br>
<br>
ReplacementText:<br>
 =A0 =A0When a subscriber wishes to subscribe to a particular state **of** =
a<br>
 =A0 =A0resource, it forms a SUBSCRIBE request **according to sec 3.1 and<b=
r>
send it across to notifier**. =A0If the initial SUBSCRIBE<br>
<br>
------------------------------<u></u>-----------------------<br>
4.1.2.1. =A0Requesting a Subscription<br>
<br>
CurrentText:<br>
 =A0 =A0[RFC3261]. =A0For the sake of clarity: if a SUBSCRIBE request conta=
ins<br>
 =A0 =A0an &quot;Accept&quot; header field, but that field does not indicat=
e a media<br>
 =A0 =A0type that the notifier is capable of generating in its NOTIFY<br>
 =A0 =A0requests, then the proper error response is 406 (Not Acceptable).<b=
r>
<br>
COMMENT:<br>
It is better to mention it as a normative text (indented).<br>
<br>
------------------------------<u></u>-----------------------<br>
4.1.3. =A0Receiving and Processing State Information<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0If the &quot;Subscription-State&quot; value is &quot;terminated&quo=
t;, the subscriber<br>
 =A0 =A0MUST consider the subscription terminated. =A0The &quot;expires&quo=
t; parameter<br>
 =A0 =A0has no semantics for &quot;terminated&quot; -- notifiers SHOULD NOT=
 include an<br>
<br>
ReplacementText:<br>
<br>
 =A0 =A0If the &quot;Subscription-State&quot; value is &quot;terminated&quo=
t;, the subscriber<br>
 =A0 =A0MUST consider the subscription terminated. =A0The &quot;expires&quo=
t; parameter<br>
 =A0 =A0has no semantics for &quot;terminated&quot; **state**. Notifiers SH=
OULD NOT<br>
include an<br>
------------------------------<u></u>-----------------------<br>
<br>
CurrentText:<br>
 =A0 =A0deactivated: =A0The subscription has been terminated, but the<br>
 =A0 =A0 =A0 subscriber SHOULD retry immediately with a new subscription. =
=A0One<br>
<br>
ReplacementText:<br>
 =A0 =A0deactivated: =A0The subscription has been terminated, but the<br>
 =A0 =A0 =A0 subscriber SHOULD retry with a new subscription. =A0One<br>
<br>
 =A0 =A0Is It mandatory to resubscribe immediately. if not remove the word<=
br>
&quot;immediately&quot;.<br>
 =A0 =A0If it is delayed, what are the consequences ????<br>
<br>
------------------------------<u></u>-----------------------<br>
4.2.1. =A0Subscription Establishment and Maintenance<br>
<br>
CurrentText:<br>
 =A0 =A0 =A0 inability to signal the success of a REFER request while signa=
ling<br>
 =A0 =A0 =A0 a problem with the subscription; and difficulty performing one=
<br>
 =A0 =A0 =A0 action without the other. =A0Additionally, the proper exchange=
 of<br>
<br>
ReplacementText:<br>
 =A0 =A0 =A0 inability to signal the success of a REFER request while signa=
ling<br>
 =A0 =A0 =A0 a problem with the subscription; and difficulty **in** perform=
ing one<br>
 =A0 =A0 =A0 action without the other. =A0Additionally, the proper exchange=
 of<br>
<br>
------------------------------<u></u>-----------------------<br>
4.2.1.4. =A0Refreshing of Subscriptions<br>
<br>
CurrentText:<br>
 =A0 =A0If no refresh for a notification address is received before its<br>
 =A0 =A0expiration time, the subscription is removed. =A0When removing a<br=
>
<br>
ReplacementText:<br>
 =A0 =A0If no refresh for a notification is received before its<br>
 =A0 =A0expiration time, the subscription is removed. =A0When removing a<br=
>
Comment:Word &quot;address&quot; needs to be removed.<br>
------------------------------<u></u>-----------------------<br>
4.3. =A0Proxy Behavior<br>
<br>
CurrentText:<br>
 =A0 =A0Proxies that did not add a Record-Route header field to the initial=
<br>
 =A0 =A0SUBSCRIBE request MUST NOT add a Record-Route header field to any o=
f<br>
 =A0 =A0the associated NOTIFY requests.<br>
<br>
ReplacementText:<br>
 =A0 =A0Proxies that did not add a Record-Route header field to the initial=
<br>
 =A0 =A0SUBSCRIBE request MUST NOT add a Record-Route header field to any o=
f<br>
 =A0 =A0the associated NOTIFY requests **or any SUBSCRIBE refresh requests*=
*.<br>
------------------------------<u></u>-----------------------<br>
4.4.1. =A0Dialog Creation and Termination<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0 =A0 A subscriber may send a SUBSCRIBE request with an &quot;Expire=
s&quot; header<br>
 =A0 =A0 =A0 field of 0 in order to trigger the sending of such a NOTIFY<br=
>
 =A0 =A0 =A0 request; however, for the purposes of subscription and dialog<=
br>
 =A0 =A0 =A0 lifetime, the subscription is not considered terminated until =
the<br>
 =A0 =A0 =A0 NOTIFY transaction with a &quot;Subscription-State&quot; of &q=
uot;terminated&quot;<br>
 =A0 =A0 =A0 completes.<br>
<br>
Comment:<br>
We may have to mention how long the dialog must be kept alive, before<br>
being cleaned up. Do we delete the dialog immediately after 200-SUB<br>
or wait till NOTIFY with &quot;Subscription-State&quot;=3D&quot;<u></u>term=
inated&quot;. If we always<br>
wait till Last Notify for dialog termination, then 200-SUB don&#39;t have a=
ny<br>
impact on the dialog termination. Atleast some normative text is of<br>
necessity, for better understanding of the topic.<br>
------------------------------<u></u>-----------------------<br>
4.4.3. =A0Polling Resource State<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0Note that the NOTIFY requests triggered by SUBSCRIBE requests with<=
br>
&quot;Expires&quot; header fields of 0 will contain a &quot;Subscription-St=
ate&quot;<br>
 =A0 =A0value of &quot;terminated&quot;, and a &quot;reason&quot; parameter=
 of &quot;timeout&quot;.<br>
<br>
ReplacementText:<br>
 =A0 =A0Note that the NOTIFY requests triggered by SUBSCRIBE requests with<=
br>
&quot;Expires&quot; header fields of 0 **shall** contain a &quot;Subscripti=
on-State&quot;<br>
 =A0 =A0value of &quot;terminated&quot;, and a &quot;reason&quot; parameter=
 of &quot;timeout&quot;.<br>
------------------------------<u></u>-----------------------<br>
4.5.2. =A0Sharing Dialogs<br>
<br>
CurrentText:<br>
 =A0 =A0 =A0 for a dialog (e.g., route set handling). =A0In particular, mul=
tiple<br>
 =A0 =A0 =A0 subscriptions within a dialog are expire independently, and<br=
>
 =A0 =A0 =A0 require independent subscription refreshes.<br>
<br>
ReplacementText:<br>
 =A0 =A0 =A0 for a dialog (e.g., route set handling). =A0In particular, mul=
tiple<br>
 =A0 =A0 =A0 subscriptions within a dialog **operates** independently, and<=
br>
 =A0 =A0 =A0 require independent subscription refreshes.<br>
------------------------------<u></u>-----------------------<br>
4.5.2. =A0Sharing Dialogs<br>
<br>
CurrentText:<br>
 =A0 =A0o =A0Subscribers MAY include an &quot;id&quot; parameter in SUBSCRI=
BE request<br>
&quot;Event&quot; header field to allow differentiation between multiple<br=
>
<br>
ReplacementText:<br>
 =A0 =A0o =A0Subscribers MAY include an &quot;id&quot; parameter in SUBSCRI=
BE **request&#39;s**<br>
&quot;Event&quot; header field to allow differentiation between multiple<br=
>
OR<br>
 =A0 =A0o =A0Subscribers MAY include an &quot;id&quot; parameter in **&quot=
;Event&quot; header field<br>
 =A0 =A0 =A0 of SUBSCRIBE request** to allow differentiation between multip=
le<br>
------------------------------<u></u>-----------------------<br>
4.5.2. =A0Sharing Dialogs<br>
<br>
CurrentText:<br>
 =A0 =A0o =A0When a subscriber refreshes a the subscription timer, the<br>
 =A0 =A0 =A0 SUBSCRIBE request MUST contain the same &quot;Event&quot; head=
er field &quot;id&quot;<br>
 =A0 =A0 =A0 parameter as was present in the SUBSCRIBE request that created=
 the<br>
 =A0 =A0 =A0 subscription. =A0(Otherwise, the notifier will interpret the<b=
r>
<br>
ReplacementText:<br>
 =A0 =A0o =A0When a subscriber refreshes a the subscription timer, the<br>
 =A0 =A0 =A0 SUBSCRIBE request MUST contain the same **&quot;id&quot; param=
eter in &quot;Event&quot;<br>
 =A0 =A0 =A0 header field, which** was present in the SUBSCRIBE request tha=
t<br>
created the<br>
 =A0 =A0 =A0 subscription. =A0(Otherwise, the notifier will interpret the<b=
r>
------------------------------<u></u>-----------------------<br>
4.5.2. =A0Sharing Dialogs<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0o =A0When a subscription is created in the notifier, it stores any<=
br>
&quot;Event&quot; header field &quot;id&quot; parameter as part of the subs=
cription<br>
 =A0 =A0 =A0 information (along with the event package name).<br>
<br>
ReplacementText:<br>
 =A0 =A0o =A0When a subscription is created in the notifier, it stores **th=
e**<br>
 =A0 =A0 =A0 **&quot;id&quot; parameter in &quot;Event&quot; header field**=
 as part of the<br>
subscription<br>
 =A0 =A0 =A0 information (along with the event package name).<br>
------------------------------<u></u>-----------------------<br>
4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transactions<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0Neither SUBSCRIBE nor NOTIFY requests can be canceled. =A0If a UAS<=
br>
 =A0 =A0receives a CANCEL request that matches a known SUBSCRIBE or NOTIFY<=
br>
 =A0 =A0transaction, it MUST respond to the CANCEL request, but otherwise<b=
r>
 =A0 =A0ignore it. =A0In particular, the CANCEL request MUST NOT affect<br>
 =A0 =A0processing of the SUBSCRIBE or NOTIFY request in any way.<br>
<br>
ReplacementText/Comment:<br>
What should be the response code for the CANCEL&#39;s reply ???<br>
Preferably 200-OK shall be used.<br>
------------------------------<u></u>-----------------------<br>
4.6. =A0CANCEL Requests for SUBSCRIBE and NOTIFY Transactions<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY<br>
 =A0 =A0transactions.<br>
<br>
ReplacementText:<br>
 =A0 =A0UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY<br>
 =A0 =A0transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY<br>
 =A0 =A0or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**<br>
------------------------------<u></u>-----------------------<br>
5.3.2. =A0State Deltas<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0In the case that the state information to be conveyed is large, the=
<br>
 =A0 =A0event package may choose to detail a scheme by which NOTIFY request=
s<br>
 =A0 =A0contain state deltas instead of complete state.<br>
<br>
<br>
ReplacementText:<br>
 =A0 =A0In the case that the state information to be conveyed is large, the=
<br>
 =A0 =A0event package may choose a scheme by which NOTIFY requests<br>
 =A0 =A0contain state deltas instead of complete state.<br>
<br>
comment:deleted words &quot;to detail&quot; in 2nd line.<br>
<br>
------------------------------<u></u>-----------------------<br>
5.3.2. =A0State Deltas<br>
<br>
CurrentText:<br>
<br>
 =A0 =A0If a NOTIFY request arrives that has a version number that is<br>
 =A0 =A0incremented by more than one, the subscriber knows that a state del=
ta<br>
 =A0 =A0has been missed; it ignores the NOTIFY request containing the state=
<br>
 =A0 =A0delta (except for the version number, which it retains to detect<br=
>
 =A0 =A0message loss), and re-sends a SUBSCRIBE request to force a NOTIFY<b=
r>
 =A0 =A0request containing a complete state snapshot.<br>
<br>
Comment:<br>
 =A0 =A0From this text, it infers that if any state delta lost, then subscr=
iber<br>
 =A0 =A0sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This re-SUBSCRIBE<b=
r>
request inturn lead<br>
 =A0 =A0to fetching the complete state information.<br>
<br>
 =A0 =A0It may be a good diea to differentiate the re-SUBSCRIBE triggered b=
y<br>
different scenarios.<br>
 =A0 =A0a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE<br>
duration. In this<br></div></div>
 =A0 =A0 =A0 case it is *not *necessary to fetch the complete state informa=
tion<div><div><br>
 =A0 =A0b) re-SUBSCRIBE triggered because of lost NOTIFY request (ref sec<b=
r>
5.3.2.<br>
 =A0 =A0 =A0 VERSION number in received NOTIFY is increased by greater than=
 ONE).<br>
 =A0 =A0 =A0 In this case subscriber wants to fetch the complete state<br>
information.<br>
<br>
 =A0 =A0If we don&#39;t differentiate these two cases, then natural re-SUBS=
CRIBE<br>
request<br>
 =A0 =A0unnecessarily fetch the complete state information, which may not b=
e<br>
necessary / expected.<br>
 =A0 =A0But uses higher network bandwidth to fetch the complete state<br>
information.<br>
------------------------------<u></u>-----------------------<br>
5.4.7. =A0Notifier generation of NOTIFY requests<br>
<br>
CurrentText:<br>
 =A0 =A0This section may optionally describe the behavior used to process t=
he<br>
 =A0 =A0subsequent response.<br>
<br>
Comment:<br>
 =A0 =A0In this section 5.4.7, the above statement is not clear, what is<br=
>
really meant to be.<br>
<br>
------------------------------<u></u>-----------------------<br>
5.4.13. =A0Use of URIs to Retrieve State<br>
<br>
CurrentText:<br>
<br>
ReplacementText:<br>
 =A0 =A0In this context, I feel Use of &quot;URLs&quot; is more appropriate=
 than &quot;URIs&quot;.<br>
------------------------------<u></u>-----------------------<br>
<br>
<br>
Thanks,<br>
Nataraju A.B.<br></div></div>
E-ID : <a href=3D"mailto:nataraju.ab@gmail.com" target=3D"_blank">nataraju.=
ab@gmail.com</a> &lt;mailto:<a href=3D"mailto:nataraju.ab@gmail.com" target=
=3D"_blank">nataraju.ab@gmail.com</a>&gt; /<br>
<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gm=
ail.com</a> &lt;mailto:<a href=3D"mailto:nataraju.sip@gmail.com" target=3D"=
_blank">nataraju.sip@gmail.com</a><u></u>&gt;<div><br>
+91-98455-95744<br>
<br>
<br>
<br>
--<br>
Thanks,<br>
Nataraju A.B.<br>
<br>
<br>
<br></div>
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org" target=3D"_blank">sipcore@ietf.org</a><=
br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/<u></u>listinfo/sipcore</a><br>
</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font color=
=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>

<br>
</div>

--f46d04016a77e086d604be5b2e0b--

From adam@nostrum.com  Mon Apr 23 12:23:46 2012
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Subject: Re: [sipcore] comments on draft-ietf-sipcore-rfc3265bis-08.txt
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On 4/23/12 11:32 AM, Nataraju A.B wrote:
>
> The comment is more of an OPTIMIZATION issue. As a minimal 
> change, can we add a normative text which provide some pointers for 
> implementers to differentiate the re-SUBSCRIBE (to extend the 
> subscription duration) and re-SUBSCRIBE (to fetch the complete state, 
> due to lost NOTIFY).

That's what RFC 5839 is for.

/a

From christer.holmberg@ericsson.com  Tue Apr 24 02:45:15 2012
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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 24 Apr 2012 11:45:11 +0200
Thread-Topic: [sipcore] Should we change mandatory-to-implement transports?
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 24 Apr 2012 12:07:05 +0200
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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2012/4/24 Christer Holmberg <christer.holmberg@ericsson.com>:
> A well implemented registrar could use serial forking in this case. But, =
as you say, if the registrations use different route sets, you can't switch=
 from one to another mid-session. In reality, however, the INVITE request/r=
esponse is going to be the largest request within a dialog (unless you e.g.=
 use an Info-Package with a large content).

But take also into account that a re-INVITE could be bigger than an
initial INVITE due to Route headers, and typically the RURI in a
re-INVITE maps to a specific IP:port and transport.




> But, even if a UA registers both using TCP, if there are proxies between =
the UA and the registrar I don't think it can be guaranteed that there will=
 be TCP all the way (similar issue as for transport=3Dtls). Or, is there a =
rule saying that a proxy must not downgrade from TCP to UDP?

Is not. But as said before, I expect that changing to TCP in case of a
big message is easier in proxy-to-proxy communication. The real
problem I see is when a UA uses UDP for common stuf (registration and
so).


BTW I don't think we should care too much about this irresolvable
problem. Instead let's encourage TCP transport in new deployments
regardless UDP "might" be valid.


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From christer.holmberg@ericsson.com  Tue Apr 24 03:14:31 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 24 Apr 2012 12:14:29 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-proxy-feature-01
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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--_000_7F2072F1E0DE894DA4B517B93C6A05852C43EFE714ESESSCMS0356e_
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Hi,

In order to progress the proxy-feature work, I would ask those who requeste=
d a new IANA registry to take a look at the latest version of the draft, wh=
ich creates such registry.

Others are of course also welcome to read/comment the draft :)

Currently there should be no documented open issues.

Thanks!

Regards,

Christer

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Christer Holmberg
Sent: 17. huhtikuuta 2012 10:05
To: sipcore@ietf.org
Subject: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01

Hi,

I've submitted a new version (-01) of the proxy-feature draft.

Based on the discussions in Paris, instead of using feature tags the draft =
now creates a new IANA "feature cap" registry.

Regards,

Christer

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nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'c=
olor:#1F497D'>Hi,<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'>In order to progress the proxy-feature work, I would ask=
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on of the draft, which creates such registry.<o:p></o:p></span></p><p class=
=3DMsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p c=
lass=3DMsoNormal><span style=3D'color:#1F497D'>Others are of course also we=
lcome to read/comment the draft </span><span style=3D'font-family:Wingdings=
;color:#1F497D'>J</span><span style=3D'color:#1F497D'><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'>Currently there sho=
uld be no documented open issues.<o:p></o:p></span></p><p class=3DMsoNormal=
><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNo=
rmal><span style=3D'color:#1F497D'>Thanks!<o:p></o:p></span></p><p class=3D=
MsoNormal><span style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p clas=
s=3DMsoNormal><span style=3D'color:#1F497D'>Regards,<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'>Christer<o:p></o:p></=
span></p><p class=3DMsoNormal><span style=3D'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'f=
ont-size:10.0pt;font-family:"Tahoma","sans-serif"'> sipcore-bounces@ietf.or=
g [mailto:sipcore-bounces@ietf.org] <b>On Behalf Of </b>Christer Holmberg<b=
r><b>Sent:</b> 17. huhtikuuta 2012 10:05<br><b>To:</b> sipcore@ietf.org<br>=
<b>Subject:</b> [sipcore] Draft new version: draft-ietf-sipcore-proxy-featu=
re-01<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p>&nbsp;</o:=
p></p><p class=3DMsoNormal><span lang=3DFI>Hi,<o:p></o:p></span></p><p clas=
s=3DMsoNormal><span lang=3DFI><o:p>&nbsp;</o:p></span></p><p class=3DMsoNor=
mal>I&#8217;ve submitted a new version (-01) of the proxy-feature draft.<o:=
p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>=
Based on the discussions in Paris, instead of using feature tags the draft =
now creates a new IANA &#8220;feature cap&#8221; registry.<o:p></o:p></p><p=
 class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Regards,<o:p><=
/o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Chr=
ister<o:p></o:p></p></div></body></html>=

--_000_7F2072F1E0DE894DA4B517B93C6A05852C43EFE714ESESSCMS0356e_--

From pravindran@sonusnet.com  Tue Apr 24 05:05:01 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
Thread-Index: AQHNIDOCuN/qLU9tjk656lTA6m0G85am5VEAgAD8KaCAAD8sgIABv2aw
Date: Tue, 24 Apr 2012 12:04:44 +0000
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Paul,

Thanks for clarify your view. In case your view is correct, then Usage of S=
IPS Scheme (Sec 9.2) of draft-ibc-sipcore-sip-websocket-02 draft should for=
bid the usage of SIPS URI for websocket transport (WSS, WS) as SIPS URI is =
mistake for TLS, the mistake *MUST NOT* be repeated and the draft text has =
to be updated as

"SIPS URI MUST NOT be used for WS, WSS transport as the end-to-end security=
 is not assured"

Please let me know your opinion on the above recommendation. I added the ex=
act current text of the draft as a note of this mail for your reference.

Thanks
Partha

Note: Current text of draft-ibc-sipcore-sip-websocket-02

9.2.  Usage of SIPS Scheme

   SIPS scheme within a SIP request dictates that the entire request
   path to the target be secured.  If such a path includes a WebSocket
   node it MUST be a secure WebSocket connection.=20

>-----Original Message-----
>From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
>Sent: Monday, April 23, 2012 8:08 PM
>To: Ravindran, Parthasarathi
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 4/23/12 1:32 AM, Ravindran, Parthasarathi wrote:
>> Paul,
>>
>> Could you please explain your view of why SIPS URI is a mistake for
>TLS transport.
>
>My view is that the *intent* of sips was to provide the sort of
>"guarantee" of security that https provides. But sips doesn't provide
>that guarantee, because it isn't e2e and a variety of other things. The
>end result is that with sips you have very little more confidence that
>you are securely talking to who you think than if you use best effort by
>using TLS on your first hop. In essence, sips provides a false sense of
>security to the naive, but little more than that.
>
>	Thanks,
>	Paul
>
>> I'm interested in knowing as apart from RFC 3261, Sec 3.1.4 of RFC
>> 5630 (recent RFC) also recommends the usage of SIPS URI with TCP as a
>> transport for TLS over TCP (http://tools.ietf.org/html/rfc5630#page-5)
>> and there is an explicit statement in the same para
>>
>> "This specification does not make use of the transport=3Dtls parameter."
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>>> Behalf Of Paul Kyzivat
>>> Sent: Monday, April 23, 2012 1:19 AM
>>> To: sipcore@ietf.org
>>> Subject: Re: [sipcore] Websocket is a new transport or new URI?
>>>
>>> Partha,
>>>
>>> IMO the main thing we learned about sips is that it was a bad idea.
>>> I don't support making that mistake again.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>> On 4/21/12 10:55 PM, Ravindran, Parthasarathi wrote:
>>>> Sal&   others,
>>>>
>>>> After reading lot of mails and thinking further, I agree with you
>>>> that
>>> Websocket is an application protocol that is used as an transport for
>>> SIP. TLS is one another application protocol on top of TCP which was
>>> considered as transport (transport=3Dtls) earlier and deprecated as
>>> part of RFC 3261 in Sec 26.2.2. TLS and Websocket has some unique
>>> behavior like hop-by-hop protocol rather than end-to-end.
>>>>
>>>> TLS is adopted within SIP using new URI namely SIPS with
>>> transport=3Dtcp. In the same way, I'm proposing SIPWS which implies
>>> websocket over TCP, SIPWSS which implies websocket over TLS over TCP.
>>> Please let me know your opinion on the same.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>>>>> Behalf Of Salvatore Loreto
>>>>> Sent: Wednesday, April 18, 2012 7:41 PM
>>>>> To: sipcore@ietf.org
>>>>> Subject: Re: [sipcore] Stateless or Transaction stateful proxy
>>>>> possible in Websocket transport? [RE: Call for Adoption:
>>>>> draft-ibc-sipcore-sip- websocket-02]
>>>>>
>>>>> On 4/18/12 4:40 PM, Christer Holmberg wrote:
>>>>>> Hi,
>>>>>>
>>>>>> I may have misunderstood something in the conversation, but my
>>>>> suggestion would be to consider *NOT* mandating SIP WebSocket
>>>>> clients to implement SIP UDP and TCP.
>>>>>>
>>>>>> Especially if I write a JavaScript SIP browser app, I very likely
>>>>> would want to support only WebSocket.
>>>>>>
>>>>>> Also, I don't think we can directly compare with SCTP, since the
>>>>> typical use-case for WebSocket is browser apps (eventhough a native
>>>>> app can of course also use WS).
>>>>>
>>>>> I think that in general we can not compare SCTP to WebSocket at all
>>>>> because the first is a pure Transport Protocol, the latter is an
>>>>> application protocol that is used as transport.
>>>>>
>>>>>   From a pure WebSocket prospective, what you are going to define
>>>>> is SIP as a websocket sub-protocol that is transported on top of.
>>>>>
>>>>> The important thing to standardize from a SIPCore prospective is
>>>>> the interaction/translations that happens between the WebSocket
>>>>> server and the SIP proxy that are two different logical entities.
>>>>>
>>>>> my two cents
>>>>> Sal
>>>>>
>>>>> --
>>>>> Salvatore Loreto, PhD
>>>>> www.sloreto.com
>>>>>
>>>>> _______________________________________________
>>>>> sipcore mailing list
>>>>> sipcore@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>>
>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>


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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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On 04/24/2012 07:04 AM, Ravindran, Parthasarathi wrote:
> Paul,
>
> Thanks for clarify your view. In case your view is correct, then Usage of SIPS Scheme (Sec 9.2) of draft-ibc-sipcore-sip-websocket-02 draft should forbid the usage of SIPS URI for websocket transport (WSS, WS) as SIPS URI is mistake for TLS, the mistake *MUST NOT* be repeated and the draft text has to be updated as
>
> "SIPS URI MUST NOT be used for WS, WSS transport as the end-to-end security is not assured"

How did you reach this conclusion? In this context, 
WebSocket-over-TLS-over-TCP is no different than TLS-over-TCP; usage of 
WebSocket on the first hop does not change the ability of the path to be 
end-to-end secure at all.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Kevin,

Please look into earlier mail of Paul's view on SIPS URI that SIPS URI (TLS=
 over TCP) is not end-to-end (http://www.ietf.org/mail-archive/web/sipcore/=
current/msg04870.html).

I asked the subsequent question to Paul based on his view on SIPS URI. Hope=
 this clarifies how I come to the below conclusion.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Kevin P. Fleming
>Sent: Tuesday, April 24, 2012 5:59 PM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 04/24/2012 07:04 AM, Ravindran, Parthasarathi wrote:
>> Paul,
>>
>> Thanks for clarify your view. In case your view is correct, then Usage
>> of SIPS Scheme (Sec 9.2) of draft-ibc-sipcore-sip-websocket-02 draft
>> should forbid the usage of SIPS URI for websocket transport (WSS, WS)
>> as SIPS URI is mistake for TLS, the mistake *MUST NOT* be repeated and
>> the draft text has to be updated as
>>
>> "SIPS URI MUST NOT be used for WS, WSS transport as the end-to-end
>security is not assured"
>
>How did you reach this conclusion? In this context, WebSocket-over-TLS-
>over-TCP is no different than TLS-over-TCP; usage of WebSocket on the
>first hop does not change the ability of the path to be end-to-end
>secure at all.
>
>--
>Kevin P. Fleming
>Digium, Inc. | Director of Software Technologies
>Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
>kpfleming
>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
>www.digium.com & www.asterisk.org
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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On 04/24/2012 07:34 AM, Ravindran, Parthasarathi wrote:
> Kevin,
>
> Please look into earlier mail of Paul's view on SIPS URI that SIPS URI (TLS over TCP) is not end-to-end (http://www.ietf.org/mail-archive/web/sipcore/current/msg04870.html).
>
> I asked the subsequent question to Paul based on his view on SIPS URI. Hope this clarifies how I come to the below conclusion.

His point was that SIPS itself has design flaws. Usage of WebSocket 
instead of TLS doesn't change that, and there's no reason that the 
WebSocket protocol specification should disallow usage of SIPS URIs over 
WebSocket connections. If an implementor wishes to use SIPS URIs and 
understands the consequences, they should be free to use them.

If you feel that *nobody* should use SIPS URIs under any circumstances 
(whether the transport is TLS, WebSocket, DTLS-SCTP or any other), then 
that's a totally different issue and should be addressed in a separate 
I-D that updates RFC3261.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From pravindran@sonusnet.com  Tue Apr 24 06:18:22 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Kevin P. Fleming" <kpfleming@digium.com>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Kevin,

Could you please let me know whether you agree that SIPS URI usage in Webso=
cket over TLS over TCP (new design) has design flaw as TLS over TCP?  Pleas=
e note that the issue is not to implementer but the (SIPS) information misl=
ead the user.=20

I'm not asking to forbid SIPS for all transport but atleast, SIPS MUST NOT =
be permitted for all the new transport which will be introduced in SIP in c=
ase it accepted as a known design flaw.

OTOH, I proposed to use new SIPWSS URI which shall indicate explicitly that=
 first hop is secured by this mechanism.

Thanks
Partha

>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Kevin P. Fleming
>Sent: Tuesday, April 24, 2012 6:14 PM
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 04/24/2012 07:34 AM, Ravindran, Parthasarathi wrote:
>> Kevin,
>>
>> Please look into earlier mail of Paul's view on SIPS URI that SIPS URI
>(TLS over TCP) is not end-to-end (http://www.ietf.org/mail-
>archive/web/sipcore/current/msg04870.html).
>>
>> I asked the subsequent question to Paul based on his view on SIPS URI.
>Hope this clarifies how I come to the below conclusion.
>
>His point was that SIPS itself has design flaws. Usage of WebSocket
>instead of TLS doesn't change that, and there's no reason that the
>WebSocket protocol specification should disallow usage of SIPS URIs over
>WebSocket connections. If an implementor wishes to use SIPS URIs and
>understands the consequences, they should be free to use them.
>
>If you feel that *nobody* should use SIPS URIs under any circumstances
>(whether the transport is TLS, WebSocket, DTLS-SCTP or any other), then
>that's a totally different issue and should be addressed in a separate
>I-D that updates RFC3261.
>
>--
>Kevin P. Fleming
>Digium, Inc. | Director of Software Technologies
>Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype:
>kpfleming
>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at
>www.digium.com & www.asterisk.org
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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On 04/24/2012 08:18 AM, Ravindran, Parthasarathi wrote:

> Could you please let me know whether you agree that SIPS URI usage in Websocket over TLS over TCP (new design) has design flaw as TLS over TCP?  Please note that the issue is not to implementer but the (SIPS) information mislead the user.
>
> I'm not asking to forbid SIPS for all transport but atleast, SIPS MUST NOT be permitted for all the new transport which will be introduced in SIP in case it accepted as a known design flaw.

The 'design flaws' that have been discussed for SIPS URIs have nothing 
at all to do with the transport mechanism that is chosen. If WebSocket 
as a transport was unable to provide an appropriate level of security to 
meet the SIPS URI requirements/expectations, then I'd agree that the 
WebSocket transport specification should disallow usage of WebSocket for 
SIPS Request URIs. However, that is not the case: WebSocket provides 
nearly identical security properties to those provided by TLS.

Given that, there is no reason to disallow usage of WebSocket for SIPS 
URIs, unless TLS is also going to be disallowed. Just like the previous 
discussion about Path and Outbound, these are implementation choices 
that should be left up to the implementor. Usage of SIPS URIs over 
WebSocket transport is not fundamentally broken, any more than it is 
fundamentally broken on any other secure transport.

If the WG wants to make a statement about usage of SIPS URIs, then it 
should be done directly, for all transports, not just for WebSocket.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

From pkyzivat@alum.mit.edu  Tue Apr 24 09:10:33 2012
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Christer,

[I'm sorry not getting this out sooner. I wrote it several days ago but 
somehow I never sent it.]

Thanks for getting this update in. In general this looks good. I have 
some comments about details:

Much of the syntax (ABNF - section 4.2.2) is a replica of what is in 
RFC3840. IMO this relationship should be acknowledged. My preference 
would be to formally reference the abnf definitions from 3840 and use 
the same names for them.

E.g.

    feature-cap       =  ["+"] fcap-name [EQUAL LDQUOT (fcap-value-list
                         / fcap-string-value ) RDQUOT]
    fcap-name         =  ftag-name
    fcap-value-list   =  tag-value-list
    fcap-string-value =  string-value
    ;; ftag-name, tag-value-list, string-value defined in RFC3840

Also, I am inclined to make the "+" mandatory, even though it really 
serves no purpose. The reason I propose this is to minimize confusion 
with the usage rules in 3840. In 3840 the "+" is mandatory except for 
the "grandfathered" tags. And we have had trouble getting people to 
follow that. If we make it optional here it will further confuse things.

OTOH, the reason for the + is so that feature tags can be distinguished 
from other contact parameters that are not feature tags. That need 
doesn't apply here, which is why the "+" is logically unnecessary. An 
alternative would be to *never* use the "+" here. But I'm inclined to 
require the "+" because there may be some tags defined for use as both 
feature tags and proxy features, and then I'd rather not use the "+" in 
place and not in the other.

I also find it some what confusing that part of the syntax is in this 
section while more of it is in section 5.4. I'd prefer to see these 
consolidated into a single syntax section.

Re Registration Trees:

There is no mention of an IETF tree, analogous to that defined in 2506 
for feature tags. Given the history, and a desire to provide a similar 
structure, I agree it makes sense that we aren't planning on defining 
anything of that sort. But I think it is confusing to anyone who has 
looked at 2506 that there is no mention of names that have only one facet.

So, I think it would be helpful to explicitly state that names 
consisting of only a single facet, as well as names with initial facets 
other than "g" and "sip" are out of scope for this draft. I'm undecided 
if it would be good to define an IETF tree without any content, or if it 
would be better not to define it.

The statements:

    The tree definitions are based on the global tree and sip tree
    defined for media feature tags, in RFC 2506 [RFC2506] and RFC 3841
    [RFC3841].
and
    The SIP feature cap global tree is based on the media feature tag
    global tree defined in RFC 2506 [RFC2506].
and
    The SIP feature cap sip tree is based on the media feature tag sip
    tree defined in RFC 3840 [RFC3840].

seem a little confusing to me, because its not clear what "based on" 
means. I suggest changing this to "similar to".

As far as defining new trees, the current text is:

    Additional feature caps trees can be created by IANA, following the
    same rules and procedures as defined for media feature tags in
    section 3.1.4 of RFC 2506 [RFC2506].

This in effect makes a normative dependency between this document and 
RFC2506. I think it would be better to sever the formal tie. Instead, 
this can be accomplished by adding something similar to section 3.1.4 of 
2506.

Regarding the IANA Registration Template:

    | The reference MUST contain the information listed in
    | section XX of XXXX (IANA: Replace XXXX with assigned
    | RFC number of this specification

I think this should say:

    | The referenced specification MUST contain the information listed
    | in section XX of XXXX (IANA: Replace XXXX with assigned
    | RFC number of this specification

In addition there may be some redundancy between the "SIP feature cap 
specification reference" and the "Related standards or documents". Do 
you think we need both?

Are there other things in there that are just baggage being carried over 
from 2506 that are not useful to us? (E.g. do we need the publication 
delay?)

Section 4.5.2 says:

    The SIP feature cap specification MUST contain an overall description
    of the SIP feature cap: how it is used to indicate support of a
    feature, a description of the feature associated with the SIP feature
    cap, and a description of any additional information (conveyed using
    one or more SIP feature cap values) that can be conveyed together
    with the SIP feature cap.

What this doesn't talk about is how the feature associated with the 
feature cap is exercised/invoked. So I would suggest something like:

    The SIP feature cap specification MUST contain an overall description
    of the SIP feature cap: how it is used to indicate support of a
    feature, a description of the feature associated with the SIP feature
    cap, a description of any additional information (conveyed using
    one or more SIP feature cap values) that can be conveyed together
    with the SIP feature cap, and a description of how the associated
    feature may be exercised/invoked.

Section 4.5.3:

The rules described here sound to me like ones suitable for the name of 
a parameter (e.g. a parameter bound to an event or a mime type, or to a 
sip header.) It doesn't seem very appropriate when discussing the 
representation of a *value* of a parameter/feature. For example, when 
values are enumerations, we might find things like "true" and "false" or 
"yes" and "no". Even if they are "red" / "green" / "blue", the same 
values might well make sense associated with other features. Can you 
please think this through further? It may indeed make sense to say 
*something* here, but I imagine something weaker.

Section 4.5.5 says:

    NOTE: Sometimes a SIP feature cap designer might choose to not reveal
    the implementation details of a SIP feature cap.  However, in order
    to allow multiple implementations to support the SIP feature cap,
    designers are strongly encouraged to provide the implementation
    details.

I recognize that people may not want to publish implementation details. 
But that is different from providing enough specification to all 
independent but interoperable implementations to be developed.

The amount that needs to be disclosed might vary based on which 
registration tree is used. I would expect that the sip tree would 
require sufficient spec to support an independent implementation. If you 
were instituting a URI tree then I can see that it might be considered 
entirely proprietary and hence might not require this. (But you have 
omitted the URI tree, which thus sidesteps this issue.) ISTM that the 
global tree should still require a definition of how to create an 
interoperable implementation. The difference between the global tree and 
the sip tree in this regard might be that in the case of the sip tree 
the referenced specification must be an IETF document, while in the case 
of the global tree it could be some external document.

Section 5.2.1 says:

    When the UA receives the SIP request or the response, the SIP feature
    caps in the topmost Feature-Caps header field will represent the
    supported features "closest" to the UA.

I don't think this is quite what you mean. I think you mean:

    When the UA receives the SIP request or the response, the SIP feature
    caps in the first fc-value will represent the supported features
    "closest" to the UA.

Section 5.2.2 says:

    The procedures in this section applies to UAs that are part of
    B2BUAs, but where the URI in the Contact header field does not
    represent the UA, because the B2BUA is also acting as a proxy and
    inserts its URI e.g. in a Record-Route header field.

IMO this covers some controversial ground, but probably only in ways 
that many will consider pedantic. We are in quite under specified 
territory when we talk about B2BUAs that act as Proxies. Does this mean 
that it was a proxy when the dialog was established, but later the same 
entity redeclares itself as a UA for some subsequent transaction within 
the dialog? Or was it acting as a UA when the dialog was established, 
but still chose to use a Contact URI that doesn't identify itself?

As much as I would like to have an extended discussion on this, doing so 
while working on this draft is probably counterproductive. Perhaps you 
can humor my sensitivities and just change this to:

    The procedures in this section applies to UAs that are part of
    B2BUAs that are referenced in the message by a Record-Route header
    field rather than by the Contact URI.

Section 5.2.2 then says:

    When a UA sends a SIP request, if the UA wants to indicate support of
    features towards its downstream SIP entities, it adds a Feature-Caps
    header field to the request, together with one or more Feature Caps
    associated with the supported features, before it forwards the
    request.

This says "adds" but doesn't say whether to be beginning or the end. 
Also, it isn't strictly necessary to add another header field, it can 
add to an existing header field. So I suggest:

    When a UA sends a SIP request, if the UA wants to indicate support of
    features towards its downstream SIP entities, before it forwards the
    request it adds a fc-value with the supported features to
    the request, after any others that may be present. This may be added
    to the last existing Feature-Caps header field or as a new
    Feature-Caps header field after any others.

Something similar applies a couple more times in this section and in 
sections 5.2.3 and 5.2.4. I realize this is a bit cumbersome - perhaps 
there is a more straightforward way to say it.

Section 5.3.3 says:

    If a SIP feature cap is inserted in a Feature-Caps header field of a
    SIP REGISTER request, or within a response of such request, it
    indicates to the receivers of the request (or response) that the
    feature associated with the SIP feature cap is supported for the
    duration of the registration, and for all SIP transactions associated
    with the registration, until the registration is re-freshed or
    terminated.

There are some cases we need to discuss here. When a UA sends a REGISTER 
request, it could have no contacts, one contact, or several. What do the 
feature caps mean in these cases?

- if there are no contacts in the register, are feature-caps irrelevant?
- if there is more than one contact in the register, do the feature-caps
   apply to all of them?

Then the response should contain all the registered contacts, not just 
those mentioned in the request. So do the feature caps mentioned in the 
reply apply until the last of those contacts expire? Or should the 
requester assume they only apply until *its* contacts expire?

    Unless a SIP feature cap is inserted in a Feature-Caps header field
    or a re-registration, or within a response of such request, it
    indicates to the receivers of the request (or response) that the
    feature is no long supported for the registration.

Suppose a UA registers with a contact and some feature caps. Then later 
it sends a register poll (register with no contacts). Can I assume that 
this will not affect the feature caps associated with the registered 
contact?

Do the enhanced registrar behaviors in outbound require any special 
treatment relative to feature caps?

Section 6:

There are a lot of missing IANA considerations - the establishment of 
the new trees as defined in 4.3 and 4.4.

That is all for now.

	Thanks,
	Paul

On 4/17/12 3:05 AM, Christer Holmberg wrote:
> Hi,
>
> IĀve submitted a new version (-01) of the proxy-feature draft.
>
> Based on the discussions in Paris, instead of using feature tags the
> draft now creates a new IANA Āfeature capĀ registry.
>
> Regards,
>
> Christer
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From dworley@avaya.com  Tue Apr 24 10:57:36 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Shida Schubert <shida@ntt-at.com>
Date: Tue, 24 Apr 2012 13:57:16 -0400
Thread-Topic: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 - sections 10.1 and 11-end
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Subject: Re: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 -	sections 10.1 and 11-end
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> From: Shida Schubert [shida@ntt-at.com]
>=20
> On Apr 20, 2012, at 11:33 AM, Worley, Dale R (Dale) wrote:
>=20
> > Correction of my comments on 10.3:  I proposed removing all of item 6:
> >
> >   6.  Missing entity: If the request clearly has a gap in the hi-entry
> >       (e.g. by evaluating via header to the existing request history to
> >       see if it traversed a domain which doesn't exist in History-Info
> >       header field.), the entity adding an hi-entry MUST add an index a
> >       digit of "0" to the current index prior to adding appropriate
> >       index for the action to be taken.  If the index of the last hi-
> >       entry in the request received was 1.1.2 and there was a missing
> >       hi-entry and the request was being forwarded to the next hop, the
> >       resulting index will be 1.1.2.0.1.
> >
> > But I should have restricted it to removing the "condition", namely
> > "(e.g. by evaluating via header to the existing request history to see
> > if it traversed a domain which doesn't exist in History-Info header
> > field.)"
>=20
>  Okay, that is more clear but I still see no harm in leaving it in 10.3.
>=20
> The text could reside in 9.2 but I personally think it provides more uses
> by having it in 10.3.

Let me reconstitute what I should have said, and then let's see if we
still disagree:

In 9.1, "Receiving a Request" (not 9.2 as I said above), is:

   If the Request-URI of the incoming request does not match the hi-
   targeted-to-uri in the last hi-entry (i.e., the previous SIP entity
   that sent the request did not include a History-Info header field),
   the SIP entity MUST add a hi-entry to end of the cache, on behalf of
   the previous SIP entity, as follows:

The condition in section 10.3 item 6 (as shown above) is:

       If the request clearly has a gap in the hi-entry
       (e.g. by evaluating via header to the existing request history to
       see if it traversed a domain which doesn't exist in History-Info
       header field.),=20

I think this condition should be added to the condition in 9.1 as an
alternative situation in which the procedures of 9.1 should be
performed.  I think we agree that we *intend* the procedures of 9.1 to
be applied in the conditions described in item 6, but I am emphasizing
that 9.1 is the "procedural" text that specifies when in processing
these procedures are done, and 9.1 should list *all* the conditions
that trigger it.

Now that I read 10.3 again, I see that 10.3 is not procedural, but
rather gives in one place all the rules for constructing hi-indexes of
new hi-entries.  And from that point of view, item 6 is proper as
written.  I might suggest that item 6 should refer to 9.1 as 9.1 is
the the implementation of the rules in item 6, but I see that no other
item in 10.3 refers to "procedural" sections; all the references are
in the reverse direction.

So I stand by my recommended change to section 9.1, but 10.3 does not
need to be changed.  Let me try to update 9.1 in a less messy way than
my last attempt:

   The SIP entity MUST detect whether the request clearly has a gap in
   the hi-entries, mandatorily by testing if the Request-URI of the
   incoming request does not match the hi-targeted-to-uri in the last
   hi-entry, and optionally by other tests, e.g. by comparing the Via
   header to the existing request history to see if it traversed a
   domain which doesn't exist in History-Info header field.  If a gap
   is detected, the SIP entity MUST add a hi-entry to end of the
   cache, on behalf of the previous SIP entity, as follows:

The original text referred to by the following remarks is:
|   Gaps are also possible in the case of
|   parallel forking if there is an outstanding request at the time the
|   SIP entity sends a response as described in Section 9.4, in which
|   case the gap will not be visible as the branch responsible for the
|   gap wasn't on the path of the request received.

> > A gap (missing branch) due to an outstanding request can be revealed
> > in both requests and responses.  The final part of the sentence I
> > can't understand.  I wonder if there was a further sentence which was
> > damaged and joined to this previous sentence.
>=20
> We were just trying to describe how there could be an index missing
> not a "0" but a branch can start from 2 instead of 1 due to parallel fork=
ing.

One problem is that "gap" is being used in two senses throughout this
draft.  One sense is "a branch of parallel forking that is not
recorded in this History-Info header, but is revealed by
non-sequential numbering of the children of an entry".  Another sense
is "an ancester of this request did not add an hi-entry".

Another is the phrase "gap will not be visible" -- the *gap* is
visible, I think you mean "the hi-entries whose absence constitute the
gap are not present".

I've tried to revise that sentence but I don't grasp exactly what
situation you are describing.  It seems to be some case of the general
rule that branches of the overall hi-entry tree can be missing at any
point, due to the timings of requests and responses in the propagation
of the request.

Dale

From pkyzivat@alum.mit.edu  Tue Apr 24 12:35:53 2012
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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On 4/24/12 8:28 AM, Kevin P. Fleming wrote:
> On 04/24/2012 07:04 AM, Ravindran, Parthasarathi wrote:
>> Paul,
>>
>> Thanks for clarify your view. In case your view is correct, then Usage
>> of SIPS Scheme (Sec 9.2) of draft-ibc-sipcore-sip-websocket-02 draft
>> should forbid the usage of SIPS URI for websocket transport (WSS, WS)
>> as SIPS URI is mistake for TLS, the mistake *MUST NOT* be repeated and
>> the draft text has to be updated as
>>
>> "SIPS URI MUST NOT be used for WS, WSS transport as the end-to-end
>> security is not assured"
>
> How did you reach this conclusion? In this context,
> WebSocket-over-TLS-over-TCP is no different than TLS-over-TCP; usage of
> WebSocket on the first hop does not change the ability of the path to be
> end-to-end secure at all.

It may be that WebSocket-over-TLS-over-TCP is *conceptually* no 
different than TLS-over-TCP. But its far from clear that this is so with 
the current wording of 3261 and 5630. I expect that a careful review of 
these needs to be done and some normative updates done. I went through 
3261 looking at discussions of sips for things that might be problematic 
or at least need some consideration. Here are some of them:


    Any
    implementation that supports TLS MUST support the SIPS URI scheme.
    The To header field allows for a display name.

    If the request arrived over TLS, and the Request-URI contained a SIPS
    URI, the "secure" flag is set to TRUE.

    If the request was sent over TLS, and the Request-URI contained a
    SIPS URI, the "secure" flag is set to TRUE.

    Furthermore, if the Request-URI contains a SIPS URI, TLS MUST
    be used to communicate with that proxy.

    For a SIPS URI, the transport parameter MUST
    indicate a reliable transport.

    When a request is sent to a SIPS URI, the
    protocol still indicates "SIP", and the transport protocol is TLS.

    When used as the Request-URI of a
    request, the SIPS scheme signifies that each hop over which the
    request is forwarded, until the request reaches the SIP entity
    responsible for the domain portion of the Request-URI, must be
    secured with TLS

    The use of SIPS in particular entails that mutual TLS authentication
    SHOULD be employed, as SHOULD the ciphersuite
    TLS_RSA_WITH_AES_128_CBC_SHA.

    Note that in the SIPS URI scheme, transport is independent of TLS,
    and thus "sips:alice@atlanta.com;transport=tcp" and
    "sips:alice@atlanta.com;transport=sctp" are both valid (although
    note that UDP is not a valid transport for SIPS).  The use of
    "transport=tls" has consequently been deprecated, partly because
    it was specific to a single hop of the request.  This is a change
    since RFC 2543.

    All SIP elements that support TLS MUST also support the SIPS URI
    scheme.

I'm not suggesting exactly *what* needs to be done - just that it needs 
to be figured out.

As a consequence, I would expect that it would be possible to use sips 
with websockets, but no requirement to do so. Whether its helpful, or 
wise, to use sips is a separate question.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Tue Apr 24 12:42:03 2012
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Subject: Re: [sipcore] comments on draft-ietf-sipcore-rfc3265bis-08.txt
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Nataraju,

On 4/23/12 12:32 PM, Nataraju A.B wrote:
> Paul,
>
> I understand, I was late into the party... :(
>
> The comment is more of an OPTIMIZATION issue. As a minimal
> change, can we add a normative text which provide some pointers for
> implementers to differentiate the re-SUBSCRIBE (to extend the
> subscription duration) and re-SUBSCRIBE (to fetch the complete state,
> due to lost NOTIFY).
>
> This might be of great concern for lossy networks (for example, wireless
> networks, ...), where all the subsequent refresh SUBSCRIBE requests
> would un-necessarily eats up more bandwidth for fetching the complete
> state which is not necessary. For non-lossy network this is of minor
> issue...
>
> Even if we decide to handle this optimization issue, we have to analyze
> thoroughly how to differentiate differentiate these requests, through
> some new header / header parameter / any other way. Need more analysis...
>
> You can take a call, based on the feasibility. Even if dropped it is
> fine....
>
> Thanks,
> Nataraju A B

Adam already responded that there exists a mechanism to optimize the 
redundant notifications away. We definitely won't embark on any 
functional changes to this draft at this late stage.

Regarding the editorial comments - its again very late, and I don't find 
any of them compelling.

In the future please try to get your reviews in no later than WGLC.

	Thanks,
	Paul

> On Mon, Apr 23, 2012 at 7:32 PM, Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>> wrote:
>
>     Nataraju,
>
>     Your comments would have been much more helpful if received some
>     months ago when the document went through WGLC. This document is now
>     *very* far down the process to publication, having already gone
>     through IESG last call, IANA review, etc. and is now in final IESG
>     Evaluation. Changes at this point are very disruptive.
>
>     If you feel your suggestions are essential then you will have to go
>     to extremes to pull this document back. I don't see anything below
>     that convinces me to do that. AFAICT the comments below are all
>     minor editorial except for the first, which suggests changing the
>     mechanism.
>
>             Thanks,
>             Paul (as co-chair)
>
>
>     On 4/23/12 12:58 AM, Nataraju A.B wrote:
>
>         All,
>
>         Some logical comments on the draft-ietf-sipcore-rfc3265bis-__08.txt.
>
>         ------------------------------__-----------------------
>         5.3.2.  State Deltas
>
>         CurrentText:
>
>             If a NOTIFY request arrives that has a version number that is
>             incremented by more than one, the subscriber knows that a
>         state delta
>             has been missed; it ignores the NOTIFY request containing
>         the state
>             delta (except for the version number, which it retains to detect
>             message loss), and re-sends a SUBSCRIBE request to force a
>         NOTIFY
>             request containing a complete state snapshot.
>
>         Comment:
>             From this text, it infers that if any state delta lost, then
>         subscriber
>             sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This
>         re-SUBSCRIBE
>         request inturn lead
>             to fetching the complete state information.
>
>             It may be a good diea to differentiate the re-SUBSCRIBE
>         triggered by
>         different scenarios.
>             a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
>         duration. In this
>                case it is not necessary to fetch the complete state
>         information
>             b) re-SUBSCRIBE triggered because of lost NOTIFY request
>         (ref sec
>         5.3.2.
>                VERSION number in received NOTIFY is increased by greater
>         than ONE).
>                In this case subscriber wants to fetch the complete state
>         information.
>
>             If we don't differentiate these two cases, then natural
>         re-SUBSCRIBE
>         request
>             unnecessarily fetch the complete state information, which
>         may not be
>         necessary / expected.
>             But uses higher network bandwidth to fetch the complete state
>         information.
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         4.4.1.  Dialog Creation and Termination
>
>         CurrentText:
>
>                A subscriber may send a SUBSCRIBE request with an
>         "Expires" header
>                field of 0 in order to trigger the sending of such a NOTIFY
>                request; however, for the purposes of subscription and dialog
>                lifetime, the subscription is not considered terminated
>         until the
>                NOTIFY transaction with a "Subscription-State" of
>         "terminated"
>                completes.
>
>         Comment:
>         We may have to mention how long the dialog must be kept alive,
>         before
>         being cleaned up. Should we delete the dialog immediately after
>         200-SUB
>         or wait till NOTIFY with "Subscription-State"="__terminated". If
>         we always
>         wait till Last Notify for dialog termination, then 200-SUB don't
>         have any
>         impact on the dialog termination. Atleast some normative text is of
>         necessity, for better understanding of the topic.
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         3.1.1.  Subscription Duration
>
>         CurrentText:
>
>             SUBSCRIBE requests SHOULD contain an "Expires" header field
>         (defined
>             in [RFC3261]).  This expires value indicates the duration of the
>             subscription.  In order to keep subscriptions effective
>         beyond the
>             duration communicated in the "Expires" header field,
>         subscribers need
>             to refresh subscriptions on a periodic basis using a new
>         SUBSCRIBE
>             request on the same dialog as defined in [RFC3261].
>
>         Comment:
>         We can add a normative text saying, the refresh time will be
>         normally 50% to 75% of subscription duration.
>
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         4.1.2.1.  Requesting a Subscription
>
>         CurrentText:
>             [RFC3261].  For the sake of clarity: if a SUBSCRIBE request
>         contains
>             an "Accept" header field, but that field does not indicate a
>         media
>             type that the notifier is capable of generating in its NOTIFY
>             requests, then the proper error response is 406 (Not
>         Acceptable).
>
>         COMMENT:
>         It is better to mention it as a normative text (indented).
>
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         4.1.2.  Creating and Maintaining Subscriptions
>
>
>         CurrentText:
>
>             From the subscriber's perspective, a subscription proceeds
>         according
>             to the following state diagram:
>
>         Comment:
>         Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
>         received in notify_wait state (because very often this is possible).
>         If no change in state of FSM, then some normative text shall
>         convey the
>         same.
>                  It may be no-operation, but better to mention the same.
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         3.1.1.  Subscription Duration
>
>         CurrentText:
>                In addition to being a request to unsubscribe, a
>         SUBSCRIBE request
>                with "Expires" of 0 also causes a fetch of state; see
>                Section 4.4.3.
>
>         Comment:
>               The above text has been mentioned to be normative text.
>         But I feel
>         this
>               shall be written as a regualr text in the draft/rfc.
>         because this
>         is the
>               expected behavior from the notifier.
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         3.2.1.
>
>         CurrentText:
>             This body will contain either the state of the subscribed
>         resource or
>             a pointer to such state in the form of a URI (see Section
>         5.4.13).
>
>         ReplacementText/Comment:
>             This body **SHALL** contain either the state of the subscribed
>         resource or
>             a pointer to such state in the form of a **URL** (see
>         Section 5.4.13).
>         ------------------------------__-----------------------
>         ------------------------------__-----------------------
>         4.2.1.  Subscription Establishment and Maintenance
>
>         CurrentText:
>                inability to signal the success of a REFER request while
>         signaling
>                a problem with the subscription; and difficulty
>         performing one
>                action without the other.  Additionally, the proper
>         exchange of
>
>         ReplacementText:
>                inability to signal the success of a REFER request while
>         signaling
>                a problem with the subscription; and difficulty **in**
>         performing one
>                action without the other.  Additionally, the proper
>         exchange of
>
>         ------------------------------__-----------------------
>
>         Thanks,
>         Nataraju A B
>
>         ---------- Forwarded message ----------
>         From: *Nataraju A.B* <nataraju.sip@gmail.com
>         <mailto:nataraju.sip@gmail.com>
>         <mailto:nataraju.sip@gmail.com <mailto:nataraju.sip@gmail.com>__>>
>         Date: Fri, Apr 13, 2012 at 10:53 PM
>         Subject: draft-ietf-sipcore-rfc3265bis-__08.txt - Review comments
>         (Nataraju A B)
>         To: Adam Roach <adam@nostrum.com <mailto:adam@nostrum.com>
>         <mailto:adam@nostrum.com <mailto:adam@nostrum.com>>>,
>         sipcore@ietf.org <mailto:sipcore@ietf.org>
>         <mailto:sipcore@ietf.org <mailto:sipcore@ietf.org>>
>
>
>         Adam,
>
>         Please find some editorial comments on the draft
>         "draft draft-ietf-sipcore-rfc3265bis-__08.txt"
>
>         For simplicity, I have mentioned both the current text and
>         proposed text.
>
>         The comments / corrective text can be found with prefix and
>         suffix  --- **
>
>         ------------------------------__-----------------------
>         1.1.  Overview of Operation
>         CurrentText:
>             The general concept is that entities in the network can
>         subscribe to
>             resource or call state for various resources or calls in the
>         network,
>             and those entities (or entities acting on their behalf) can send
>             notifications when those states change.
>
>         ReplacementText:
>             The general concept is that entities in the network can
>         subscribe to
>             resource or call state **at / of** various resources or
>         calls in the
>         network,
>             and those entities (or entities acting on their behalf)
>         **who** can send
>             notifications when those states change.
>         ------------------------------__-----------------------
>
>         1.1.  Overview of Operation
>
>         CurrentText:
>             Subscriptions are expired and must be refreshed by subsequent
>             SUBSCRIBE requests.
>
>         ReplacementText:
>             Subscriptions are expired and must be refreshed by subsequent
>             SUBSCRIBE requests **before expiration**.
>         ------------------------------__-----------------------
>
>         2.  Definitions
>
>         CurrentText:
>             Subscriber:  A subscriber is a user agent which receives NOTIFY
>                requests from notifiers; these NOTIFY requests contain
>         information
>                about the state of a resource in which the subscriber is
>                interested.  Subscribers typically also generate SUBSCRIBE
>                requests and send them to notifiers to create subscriptions.
>
>         Comment:
>         After this information, we can add a normative text which indicates
>         when a SIP entity might get NOTIFY messages without actually
>         sending SUBSCRIBE request. For example, REFER request might end
>         up with
>         implicit subscription. Also mention any other requests which may
>         lead to
>         implicit subscriptions (other than REFER).
>         ------------------------------__-----------------------
>         2.  Definitions
>
>         CurrentText:
>             Subscription:  A subscription is a set of application state
>                associated with a dialog.  This application state includes a
>                pointer to the associated dialog, the event package name, and
>                possibly an identification token.  Event packages will define
>                additional subscription state information.  By definition,
>                subscriptions exist in both a subscriber and a notifier.
>
>         ReplacementText:
>             Subscription:  A subscription is a set of application **states**
>                associated with a dialog.  **The** application state
>         includes a
>                pointer to the associated dialog, the event package name, and
>                possibly an identification token.  Event packages will define
>                additional subscription state information.  By definition,
>                subscriptions exist in both a subscriber and a notifier.
>         ------------------------------__-----------------------
>         3.1.1.  Subscription Duration
>
>         CurrentText:
>
>             subscription.  In order to keep subscriptions effective
>         beyond the
>             duration communicated in the "Expires" header field,
>         subscribers need
>             to refresh subscriptions on a periodic basis using a new
>         SUBSCRIBE
>             request on the same dialog as defined in [RFC3261].
>
>         ReplacementText
>         3.1.1.  Subscription Duration
>
>             subscription.  In order to keep subscriptions effective
>         beyond the
>             duration communicated in the "Expires" header field,
>         subscribers need
>             to refresh subscriptions **before expiration** on a periodic
>         basis
>         using a new SUBSCRIBE
>             request on the same dialog as defined in [RFC3261].
>         ------------------------------__-----------------------
>         3.1.1.  Subscription Duration
>
>         CurrentText:
>                In addition to being a request to unsubscribe, a
>         SUBSCRIBE request
>                with "Expires" of 0 also causes a fetch of state; see
>                Section 4.4.3.
>
>         Comment:
>               The above text has been mentioned to be normative text.
>         But I feel
>         this
>               shall be written as a regualr text in the draft/rfc.
>         because this
>         is the
>               expected behavior from the notifier.
>         ------------------------------__-----------------------
>         3.1.2.  Identification of Subscribed Events and Event Classes
>
>         CurrentText:
>             Subscribers MUST include exactly one "Event" header field in
>             SUBSCRIBE requests, indicating to which event or class of
>         events they
>             are subscribing.  The "Event" header field will contain a
>         token which
>             indicates the type of state for which a subscription is being
>             requested.  This token will be registered with the IANA and will
>             correspond to an event package which further describes the
>         semantics
>             of the event or event class.
>
>
>         ReplacementText:
>             Subscribers MUST include exactly one "Event" header field in
>             SUBSCRIBE requests, indicating to which event or class of
>         events they
>             are subscribing.  The "Event" header field **MUST** contain
>         a token
>         which
>             indicates the type of state for which a subscription is being
>             requested.  This token will be registered with the IANA and will
>             correspond to an event package which further describes the
>         semantics
>             of the event or event class.
>         ------------------------------__-----------------------
>         3.1.  SUBSCRIBE
>
>         CurrentText:
>             The SUBSCRIBE method is used to request current state and state
>             updates from a remote node.  SUBSCRIBE requests are target
>         refresh
>
>
>         ReplacementText:
>             The SUBSCRIBE method is used to request current state and
>         **subsequent** state
>             updates from a remote node.  SUBSCRIBE requests are target
>         refresh
>
>
>         ------------------------------__-----------------------
>         3.1.1.  Subscription Duration
>
>         CurrentText:
>
>             SUBSCRIBE requests SHOULD contain an "Expires" header field
>         (defined
>             in [RFC3261]).  This expires value indicates the duration of the
>             subscription.  In order to keep subscriptions effective
>         beyond the
>             duration communicated in the "Expires" header field,
>         subscribers need
>             to refresh subscriptions on a periodic basis using a new
>         SUBSCRIBE
>             request on the same dialog as defined in [RFC3261].
>
>         Comment:
>         We can add a normative text saying, the refresh time will be
>         normally 50% to 75% of subscription duration.
>
>         ------------------------------__-----------------------
>         3.2.1.
>
>         CurrentText:
>             This body will contain either the state of the subscribed
>         resource or
>             a pointer to such state in the form of a URI (see Section
>         5.4.13).
>
>         ReplacementText/Comment:
>             This body **SHALL** contain either the state of the subscribed
>         resource or
>             a pointer to such state in the form of a **URL** (see
>         Section 5.4.13).
>         ------------------------------__-----------------------
>         4.1.2.  Creating and Maintaining Subscriptions
>
>
>         CurrentText:
>
>             From the subscriber's perspective, a subscription proceeds
>         according
>             to the following state diagram:
>
>         Comment:
>         Here it is better to mention, what happens to FSM when 200-SUBSCRIBE
>         received in notify_wait state (because very often this is possible).
>         If no change in state of FSM, then some normative text shall
>         convey the
>         same.
>                  It may be no-operation, but better to mention the same.
>         ------------------------------__-----------------------
>         4.1.2.1.  Requesting a Subscription
>
>         CurrentText:
>             When a subscriber wishes to subscribe to a particular state
>         for a
>             resource, it forms a SUBSCRIBE request.  If the initial
>         SUBSCRIBE
>
>         ReplacementText:
>             When a subscriber wishes to subscribe to a particular state
>         **of** a
>             resource, it forms a SUBSCRIBE request **according to sec
>         3.1 and
>         send it across to notifier**.  If the initial SUBSCRIBE
>
>         ------------------------------__-----------------------
>         4.1.2.1.  Requesting a Subscription
>
>         CurrentText:
>             [RFC3261].  For the sake of clarity: if a SUBSCRIBE request
>         contains
>             an "Accept" header field, but that field does not indicate a
>         media
>             type that the notifier is capable of generating in its NOTIFY
>             requests, then the proper error response is 406 (Not
>         Acceptable).
>
>         COMMENT:
>         It is better to mention it as a normative text (indented).
>
>         ------------------------------__-----------------------
>         4.1.3.  Receiving and Processing State Information
>
>         CurrentText:
>
>             If the "Subscription-State" value is "terminated", the
>         subscriber
>             MUST consider the subscription terminated.  The "expires"
>         parameter
>             has no semantics for "terminated" -- notifiers SHOULD NOT
>         include an
>
>         ReplacementText:
>
>             If the "Subscription-State" value is "terminated", the
>         subscriber
>             MUST consider the subscription terminated.  The "expires"
>         parameter
>             has no semantics for "terminated" **state**. Notifiers
>         SHOULD NOT
>         include an
>         ------------------------------__-----------------------
>
>         CurrentText:
>             deactivated:  The subscription has been terminated, but the
>                subscriber SHOULD retry immediately with a new
>         subscription.  One
>
>         ReplacementText:
>             deactivated:  The subscription has been terminated, but the
>                subscriber SHOULD retry with a new subscription.  One
>
>             Is It mandatory to resubscribe immediately. if not remove
>         the word
>         "immediately".
>             If it is delayed, what are the consequences ????
>
>         ------------------------------__-----------------------
>         4.2.1.  Subscription Establishment and Maintenance
>
>         CurrentText:
>                inability to signal the success of a REFER request while
>         signaling
>                a problem with the subscription; and difficulty
>         performing one
>                action without the other.  Additionally, the proper
>         exchange of
>
>         ReplacementText:
>                inability to signal the success of a REFER request while
>         signaling
>                a problem with the subscription; and difficulty **in**
>         performing one
>                action without the other.  Additionally, the proper
>         exchange of
>
>         ------------------------------__-----------------------
>         4.2.1.4.  Refreshing of Subscriptions
>
>         CurrentText:
>             If no refresh for a notification address is received before its
>             expiration time, the subscription is removed.  When removing a
>
>         ReplacementText:
>             If no refresh for a notification is received before its
>             expiration time, the subscription is removed.  When removing a
>         Comment:Word "address" needs to be removed.
>         ------------------------------__-----------------------
>         4.3.  Proxy Behavior
>
>         CurrentText:
>             Proxies that did not add a Record-Route header field to the
>         initial
>             SUBSCRIBE request MUST NOT add a Record-Route header field
>         to any of
>             the associated NOTIFY requests.
>
>         ReplacementText:
>             Proxies that did not add a Record-Route header field to the
>         initial
>             SUBSCRIBE request MUST NOT add a Record-Route header field
>         to any of
>             the associated NOTIFY requests **or any SUBSCRIBE refresh
>         requests**.
>         ------------------------------__-----------------------
>         4.4.1.  Dialog Creation and Termination
>
>         CurrentText:
>
>                A subscriber may send a SUBSCRIBE request with an
>         "Expires" header
>                field of 0 in order to trigger the sending of such a NOTIFY
>                request; however, for the purposes of subscription and dialog
>                lifetime, the subscription is not considered terminated
>         until the
>                NOTIFY transaction with a "Subscription-State" of
>         "terminated"
>                completes.
>
>         Comment:
>         We may have to mention how long the dialog must be kept alive,
>         before
>         being cleaned up. Do we delete the dialog immediately after 200-SUB
>         or wait till NOTIFY with "Subscription-State"="__terminated". If
>         we always
>         wait till Last Notify for dialog termination, then 200-SUB don't
>         have any
>         impact on the dialog termination. Atleast some normative text is of
>         necessity, for better understanding of the topic.
>         ------------------------------__-----------------------
>         4.4.3.  Polling Resource State
>
>         CurrentText:
>
>             Note that the NOTIFY requests triggered by SUBSCRIBE
>         requests with
>         "Expires" header fields of 0 will contain a "Subscription-State"
>             value of "terminated", and a "reason" parameter of "timeout".
>
>         ReplacementText:
>             Note that the NOTIFY requests triggered by SUBSCRIBE
>         requests with
>         "Expires" header fields of 0 **shall** contain a
>         "Subscription-State"
>             value of "terminated", and a "reason" parameter of "timeout".
>         ------------------------------__-----------------------
>         4.5.2.  Sharing Dialogs
>
>         CurrentText:
>                for a dialog (e.g., route set handling).  In particular,
>         multiple
>                subscriptions within a dialog are expire independently, and
>                require independent subscription refreshes.
>
>         ReplacementText:
>                for a dialog (e.g., route set handling).  In particular,
>         multiple
>                subscriptions within a dialog **operates** independently, and
>                require independent subscription refreshes.
>         ------------------------------__-----------------------
>         4.5.2.  Sharing Dialogs
>
>         CurrentText:
>             o  Subscribers MAY include an "id" parameter in SUBSCRIBE
>         request
>         "Event" header field to allow differentiation between multiple
>
>         ReplacementText:
>             o  Subscribers MAY include an "id" parameter in SUBSCRIBE
>         **request's**
>         "Event" header field to allow differentiation between multiple
>         OR
>             o  Subscribers MAY include an "id" parameter in **"Event"
>         header field
>                of SUBSCRIBE request** to allow differentiation between
>         multiple
>         ------------------------------__-----------------------
>         4.5.2.  Sharing Dialogs
>
>         CurrentText:
>             o  When a subscriber refreshes a the subscription timer, the
>                SUBSCRIBE request MUST contain the same "Event" header
>         field "id"
>                parameter as was present in the SUBSCRIBE request that
>         created the
>                subscription.  (Otherwise, the notifier will interpret the
>
>         ReplacementText:
>             o  When a subscriber refreshes a the subscription timer, the
>                SUBSCRIBE request MUST contain the same **"id" parameter
>         in "Event"
>                header field, which** was present in the SUBSCRIBE
>         request that
>         created the
>                subscription.  (Otherwise, the notifier will interpret the
>         ------------------------------__-----------------------
>         4.5.2.  Sharing Dialogs
>
>         CurrentText:
>
>             o  When a subscription is created in the notifier, it stores any
>         "Event" header field "id" parameter as part of the subscription
>                information (along with the event package name).
>
>         ReplacementText:
>             o  When a subscription is created in the notifier, it stores
>         **the**
>                **"id" parameter in "Event" header field** as part of the
>         subscription
>                information (along with the event package name).
>         ------------------------------__-----------------------
>         4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>
>         CurrentText:
>
>             Neither SUBSCRIBE nor NOTIFY requests can be canceled.  If a UAS
>             receives a CANCEL request that matches a known SUBSCRIBE or
>         NOTIFY
>             transaction, it MUST respond to the CANCEL request, but
>         otherwise
>             ignore it.  In particular, the CANCEL request MUST NOT affect
>             processing of the SUBSCRIBE or NOTIFY request in any way.
>
>         ReplacementText/Comment:
>         What should be the response code for the CANCEL's reply ???
>         Preferably 200-OK shall be used.
>         ------------------------------__-----------------------
>         4.6.  CANCEL Requests for SUBSCRIBE and NOTIFY Transactions
>
>         CurrentText:
>
>             UACs SHOULD NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>             transactions.
>
>         ReplacementText:
>             UACs **MUST** NOT send CANCEL requests for SUBSCRIBE or NOTIFY
>             transactions. **UASs SHOULD send 200OK for CANCEL within NOTIFY
>             or SUBSCRIBE transaction duration. As per sec 9 of [RFC3261]**
>         ------------------------------__-----------------------
>         5.3.2.  State Deltas
>
>         CurrentText:
>
>             In the case that the state information to be conveyed is
>         large, the
>             event package may choose to detail a scheme by which NOTIFY
>         requests
>             contain state deltas instead of complete state.
>
>
>         ReplacementText:
>             In the case that the state information to be conveyed is
>         large, the
>             event package may choose a scheme by which NOTIFY requests
>             contain state deltas instead of complete state.
>
>         comment:deleted words "to detail" in 2nd line.
>
>         ------------------------------__-----------------------
>         5.3.2.  State Deltas
>
>         CurrentText:
>
>             If a NOTIFY request arrives that has a version number that is
>             incremented by more than one, the subscriber knows that a
>         state delta
>             has been missed; it ignores the NOTIFY request containing
>         the state
>             delta (except for the version number, which it retains to detect
>             message loss), and re-sends a SUBSCRIBE request to force a
>         NOTIFY
>             request containing a complete state snapshot.
>
>         Comment:
>             From this text, it infers that if any state delta lost, then
>         subscriber
>             sends refresh SUBSCRIBE (re-SUBSCRIBE) request. This
>         re-SUBSCRIBE
>         request inturn lead
>             to fetching the complete state information.
>
>             It may be a good diea to differentiate the re-SUBSCRIBE
>         triggered by
>         different scenarios.
>             a) natural re-SUBSCRIBE request, which extends the SUBSCRIBE
>         duration. In this
>                case it is *not *necessary to fetch the complete state
>         information
>
>             b) re-SUBSCRIBE triggered because of lost NOTIFY request
>         (ref sec
>         5.3.2.
>                VERSION number in received NOTIFY is increased by greater
>         than ONE).
>                In this case subscriber wants to fetch the complete state
>         information.
>
>             If we don't differentiate these two cases, then natural
>         re-SUBSCRIBE
>         request
>             unnecessarily fetch the complete state information, which
>         may not be
>         necessary / expected.
>             But uses higher network bandwidth to fetch the complete state
>         information.
>         ------------------------------__-----------------------
>         5.4.7.  Notifier generation of NOTIFY requests
>
>         CurrentText:
>             This section may optionally describe the behavior used to
>         process the
>             subsequent response.
>
>         Comment:
>             In this section 5.4.7, the above statement is not clear, what is
>         really meant to be.
>
>         ------------------------------__-----------------------
>         5.4.13.  Use of URIs to Retrieve State
>
>         CurrentText:
>
>         ReplacementText:
>             In this context, I feel Use of "URLs" is more appropriate
>         than "URIs".
>         ------------------------------__-----------------------
>
>
>         Thanks,
>         Nataraju A.B.
>         E-ID : nataraju.ab@gmail.com <mailto:nataraju.ab@gmail.com>
>         <mailto:nataraju.ab@gmail.com <mailto:nataraju.ab@gmail.com>> /
>         nataraju.sip@gmail.com <mailto:nataraju.sip@gmail.com>
>         <mailto:nataraju.sip@gmail.com <mailto:nataraju.sip@gmail.com>__>
>
>         +91-98455-95744
>
>
>
>         --
>         Thanks,
>         Nataraju A.B.
>
>
>
>         _________________________________________________
>         sipcore mailing list
>         sipcore@ietf.org <mailto:sipcore@ietf.org>
>         https://www.ietf.org/mailman/__listinfo/sipcore
>         <https://www.ietf.org/mailman/listinfo/sipcore>
>
>
>     _________________________________________________
>     sipcore mailing list
>     sipcore@ietf.org <mailto:sipcore@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/sipcore
>     <https://www.ietf.org/mailman/listinfo/sipcore>
>
>
>
>
> --
> Thanks,
> Nataraju A.B.
>


From dworley@avaya.com  Tue Apr 24 12:46:17 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: Shida Schubert <shida@ntt-at.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 24 Apr 2012 15:44:41 -0400
Thread-Topic: New diff for rfc4244bis
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Trying to get all the issues cleaned up...

Comments relative to draft-ietf-sipcore-rfc4244bis-08-04-ss.txt:

Of course, "Comments on draft-ietf-sipcore-rfc4244bis-07 -
substantival questions" remains to be discussed.

In 10.2 is the sentence "That allows Tel-URIs to be handled correctly,
i.e., not receiving a Reason header."  But I wrote that sentence as a
comment on the text I was providing, I don't think it helps the draft
itself!  It probably got copied into the draft by mistake.

> > NOTE:  In my comments on sections 1-5, I was unaware of the (somewhat
> > new) use of "0" to indicate a gap, so I incorrectly referred to
> > "positive integer" instead of "nonnegative integer".
> >
> > NOTE:  Should we add somewhere prominent toward the beginning:
> >
> >   The hi-entries in the History-Info header field are always given in
> >   the preorder of their logical tree structure, which is the
> >   lexicographic ordering of their hi-indexes.
> >
> > This makes it clear that the ordering is always the same and avoids
> > specifying the ordering in multiple places (as if it was somehow a
> > special feature of certain steps of processing).
>=20
>  I thought the description of hi-index was sufficient but if you don't
> think so, we can add this. Where do you recommend we add this?
> In the Introduction??

I might be having some sort of conceptual confusion here.  My
understanding is that the hi-entries are intrinsically organized into
a tree by their hi-indexes, and as a consequence, whenever they are
written as one or more History-Info headers, the hi-entries are in
tree preorder by their hi-indexes.

So the first question is whether everybody agrees with my
understanding.

Within that understanding, there are various places in processing
where any implementation will have to go to the effort of correctly
ordering the hi-entries.  The proper understanding of that effort is
not "sort the hi-entries because that is what you do at this point"
but rather "sort the hi-entries because that is what you do at any
point where you express them as History-Info".  So there should be a
statement of this global principle (contstraint, invariant)
somewhere.  It might be that it should be expressed in section 10.3,
which seems to be the location for all the general or global
principles for hi-entries.

> >   6.  Missing entity: If the request clearly has a gap in the hi-entry
> >       (e.g. by evaluating via header to the existing request history to
> >       see if it traversed a domain which doesn't exist in History-Info
> >       header field.), the entity adding an hi-entry MUST add an index a
> >       digit of "0" to the current index prior to adding appropriate
> >       index for the action to be taken.  If the index of the last hi-
> >       entry in the request received was 1.1.2 and there was a missing
> >       hi-entry and the request was being forwarded to the next hop, the
> >       resulting index will be 1.1.2.0.1.
> >
> > This rule doesn't really belong here.  Properly, it describes an
> > additional condition that triggers the adding of an hi-entry described
> > in section 9.1.
>=20
>  Section 9.1 already describes the condition as to when it
> adds hi-entry.
>=20
>  " If the Request-URI of the incoming request does not match the hi-
>    targeted-to-uri in the last hi-entry (i.e., the previous SIP entity
>    that sent the request did not include a History-Info header field),
>    the SIP entity MUST add a hi-entry to end of the cache, on behalf of
>    the previous SIP entity"
>=20
>  Above is strictly talking about condition when "0" is to be used for
> index.
>=20
>  The fact that last hi-entry and R-URI doesn't match is a rule for
> adding an hi-entry there is no additional rule for adding an
> hi-entry on behalf of the previous hop.

Are you saying that these two processes are not essentially the same?
If so, I have some misunderstanding as to what the difference is.  Can
you clarify what the difference between the two is?

Dale

From shida@ntt-at.com  Tue Apr 24 22:26:56 2012
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Dale;

 My comments inline..

On Apr 25, 2012, at 4:44 AM, Worley, Dale R (Dale) wrote:

> Trying to get all the issues cleaned up...
> 
> Comments relative to draft-ietf-sipcore-rfc4244bis-08-04-ss.txt:
> 
> Of course, "Comments on draft-ietf-sipcore-rfc4244bis-07 -
> substantival questions" remains to be discussed.
> 
> In 10.2 is the sentence "That allows Tel-URIs to be handled correctly,
> i.e., not receiving a Reason header."  But I wrote that sentence as a
> comment on the text I was providing, I don't think it helps the draft
> itself!  It probably got copied into the draft by mistake.

 Will remove it.. Thanks.

> 
>>> NOTE:  In my comments on sections 1-5, I was unaware of the (somewhat
>>> new) use of "0" to indicate a gap, so I incorrectly referred to
>>> "positive integer" instead of "nonnegative integer".
>>> 
>>> NOTE:  Should we add somewhere prominent toward the beginning:
>>> 
>>>  The hi-entries in the History-Info header field are always given in
>>>  the preorder of their logical tree structure, which is the
>>>  lexicographic ordering of their hi-indexes.
>>> 
>>> This makes it clear that the ordering is always the same and avoids
>>> specifying the ordering in multiple places (as if it was somehow a
>>> special feature of certain steps of processing).
>> 
>> I thought the description of hi-index was sufficient but if you don't
>> think so, we can add this. Where do you recommend we add this?
>> In the Introduction??
> 
> I might be having some sort of conceptual confusion here.  My
> understanding is that the hi-entries are intrinsically organized into
> a tree by their hi-indexes, and as a consequence, whenever they are
> written as one or more History-Info headers, the hi-entries are in
> tree preorder by their hi-indexes.

 Correct.

> 
> So the first question is whether everybody agrees with my
> understanding.

 Yes.

> 
> Within that understanding, there are various places in processing
> where any implementation will have to go to the effort of correctly
> ordering the hi-entries.  The proper understanding of that effort is
> not "sort the hi-entries because that is what you do at this point"
> but rather "sort the hi-entries because that is what you do at any
> point where you express them as History-Info".  So there should be a
> statement of this global principle (contstraint, invariant)
> somewhere.  It might be that it should be expressed in section 10.3,
> which seems to be the location for all the general or global
> principles for hi-entries.

 I am still having a difficult time understanding what type of 
description you are desiring in section 10.3. I agree that if we are 
to add something that it will be in section 10.3, but I am still 
unclear what you want to see added.

 Are you referring to mentioning the potential gap(s) (restriction) ?

> 
>>>  6.  Missing entity: If the request clearly has a gap in the hi-entry
>>>      (e.g. by evaluating via header to the existing request history to
>>>      see if it traversed a domain which doesn't exist in History-Info
>>>      header field.), the entity adding an hi-entry MUST add an index a
>>>      digit of "0" to the current index prior to adding appropriate
>>>      index for the action to be taken.  If the index of the last hi-
>>>      entry in the request received was 1.1.2 and there was a missing
>>>      hi-entry and the request was being forwarded to the next hop, the
>>>      resulting index will be 1.1.2.0.1.
>>> 
>>> This rule doesn't really belong here.  Properly, it describes an
>>> additional condition that triggers the adding of an hi-entry described
>>> in section 9.1.
>> 
>> Section 9.1 already describes the condition as to when it
>> adds hi-entry.
>> 
>> " If the Request-URI of the incoming request does not match the hi-
>>   targeted-to-uri in the last hi-entry (i.e., the previous SIP entity
>>   that sent the request did not include a History-Info header field),
>>   the SIP entity MUST add a hi-entry to end of the cache, on behalf of
>>   the previous SIP entity"
>> 
>> Above is strictly talking about condition when "0" is to be used for
>> index.
>> 
>> The fact that last hi-entry and R-URI doesn't match is a rule for
>> adding an hi-entry there is no additional rule for adding an
>> hi-entry on behalf of the previous hop.
> 
> Are you saying that these two processes are not essentially the same?
> If so, I have some misunderstanding as to what the difference is.  Can
> you clarify what the difference between the two is?

 I will comment on this in the other e-mail you sent out. 

 Anyway, I agree that the behavior described in the (e.g. ...) can be 
moved or copied to section 9.1. 

 Regards
  Shida


> 
> Dale


From shida@ntt-at.com  Tue Apr 24 22:48:58 2012
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Subject: Re: [sipcore] Comments on draft-ietf-sipcore-rfc4244bis-07 - sections 10.1 and 11-end
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Hi Dale;

 Many thanks for your feedbacks.

 My comments inline.

On Apr 25, 2012, at 2:57 AM, Worley, Dale R (Dale) wrote:

>> From: Shida Schubert [shida@ntt-at.com]
>>=20
>> On Apr 20, 2012, at 11:33 AM, Worley, Dale R (Dale) wrote:
>>=20
>>> Correction of my comments on 10.3:  I proposed removing all of item =
6:
>>>=20
>>>  6.  Missing entity: If the request clearly has a gap in the =
hi-entry
>>>      (e.g. by evaluating via header to the existing request history =
to
>>>      see if it traversed a domain which doesn't exist in =
History-Info
>>>      header field.), the entity adding an hi-entry MUST add an index =
a
>>>      digit of "0" to the current index prior to adding appropriate
>>>      index for the action to be taken.  If the index of the last hi-
>>>      entry in the request received was 1.1.2 and there was a missing
>>>      hi-entry and the request was being forwarded to the next hop, =
the
>>>      resulting index will be 1.1.2.0.1.
>>>=20
>>> But I should have restricted it to removing the "condition", namely
>>> "(e.g. by evaluating via header to the existing request history to =
see
>>> if it traversed a domain which doesn't exist in History-Info header
>>> field.)"
>>=20
>> Okay, that is more clear but I still see no harm in leaving it in =
10.3.
>>=20
>> The text could reside in 9.2 but I personally think it provides more =
uses
>> by having it in 10.3.
>=20
> Let me reconstitute what I should have said, and then let's see if we
> still disagree:
>=20
> In 9.1, "Receiving a Request" (not 9.2 as I said above), is:
>=20
>   If the Request-URI of the incoming request does not match the hi-
>   targeted-to-uri in the last hi-entry (i.e., the previous SIP entity
>   that sent the request did not include a History-Info header field),
>   the SIP entity MUST add a hi-entry to end of the cache, on behalf of
>   the previous SIP entity, as follows:
>=20
> The condition in section 10.3 item 6 (as shown above) is:
>=20
>       If the request clearly has a gap in the hi-entry
>       (e.g. by evaluating via header to the existing request history =
to
>       see if it traversed a domain which doesn't exist in History-Info
>       header field.),=20
>=20
> I think this condition should be added to the condition in 9.1 as an
> alternative situation in which the procedures of 9.1 should be
> performed.  I think we agree that we *intend* the procedures of 9.1 to
> be applied in the conditions described in item 6, but I am emphasizing
> that 9.1 is the "procedural" text that specifies when in processing
> these procedures are done, and 9.1 should list *all* the conditions
> that trigger it.

 I agree.=20

>=20
> Now that I read 10.3 again, I see that 10.3 is not procedural, but
> rather gives in one place all the rules for constructing hi-indexes of
> new hi-entries.  And from that point of view, item 6 is proper as
> written.  I might suggest that item 6 should refer to 9.1 as 9.1 is
> the the implementation of the rules in item 6, but I see that no other
> item in 10.3 refers to "procedural" sections; all the references are
> in the reverse direction.
>=20
> So I stand by my recommended change to section 9.1, but 10.3 does not
> need to be changed.  Let me try to update 9.1 in a less messy way than
> my last attempt:
>=20
>   The SIP entity MUST detect whether the request clearly has a gap in
>   the hi-entries, mandatorily by testing if the Request-URI of the
>   incoming request does not match the hi-targeted-to-uri in the last
>   hi-entry, and optionally by other tests, e.g. by comparing the Via
>   header to the existing request history to see if it traversed a
>   domain which doesn't exist in History-Info header field.  If a gap
>   is detected, the SIP entity MUST add a hi-entry to end of the
>   cache, on behalf of the previous SIP entity, as follows:

 This looks good.=20

 Although may be simply removing the condition in 10.3=20
(evaluating via etc.) alone may be sufficient.=20

 Comparing via header is probably unnecessary as receiving a=20
request with hi-entry which doesn't match the R-URI is enough=20
to detect a gap. =20

>=20
> The original text referred to by the following remarks is:
> |   Gaps are also possible in the case of
> |   parallel forking if there is an outstanding request at the time =
the
> |   SIP entity sends a response as described in Section 9.4, in which
> |   case the gap will not be visible as the branch responsible for the
> |   gap wasn't on the path of the request received.
>=20
>>> A gap (missing branch) due to an outstanding request can be revealed
>>> in both requests and responses.  The final part of the sentence I
>>> can't understand.  I wonder if there was a further sentence which =
was
>>> damaged and joined to this previous sentence.
>>=20
>> We were just trying to describe how there could be an index missing
>> not a "0" but a branch can start from 2 instead of 1 due to parallel =
forking.
>=20
> One problem is that "gap" is being used in two senses throughout this
> draft.  One sense is "a branch of parallel forking that is not
> recorded in this History-Info header, but is revealed by
> non-sequential numbering of the children of an entry".  Another sense
> is "an ancester of this request did not add an hi-entry".

 The referenced text refers to the gap in the sense that "a branch of=20
parallel forking that is not recorded"=20

>=20
> Another is the phrase "gap will not be visible" -- the *gap* is
> visible, I think you mean "the hi-entries whose absence constitute the
> gap are not present".

 The gap in this sense (parallel forking) may not be visible if you=20
are received a request from the first branch.=20

 Let's say it forked at the third hop and you were the fourth hop the=20
hi-indexes would look like. 1.1.1.1
=20
 Now if you were on the second fork it would look like: 1.1.2.1 without=20=

the 1.1.1 entry, so then you know that forking likely happened and=20
gap is visible.=20

>=20
> I've tried to revise that sentence but I don't grasp exactly what
> situation you are describing.  It seems to be some case of the general
> rule that branches of the overall hi-entry tree can be missing at any
> point, due to the timings of requests and responses in the propagation
> of the request.

 I hope above comments clarify this.=20

 Regards
  Shida

=20

>=20
> Dale


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From pravindran@sonusnet.com  Wed Apr 25 02:18:17 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
Thread-Index: AQHNIDOCuN/qLU9tjk656lTA6m0G85am5VEAgAD8KaCAAD8sgIABv2aw//+u7wCAAHdUgIABOZfA
Date: Wed, 25 Apr 2012 09:18:02 +0000
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Paul,

As you mentioned, it is possible to use SIPS URI even for the new=20
transport like websocket with some normative changes and those=20
changes needs to be figured out. As a result, SIPS URI is not going
to be mistake in the new transport as well.

My intent of this mail is to make consistent mechanism for all SIP=20
application layer transport protocol over TCP. "Websocket=20
over TCP" is same like "TLS over TCP" and has some of similar=20
behavior like

1) Hop by hop behavior.=20
2) Running on top of TCP transport
3) No port management

This mechanism has other added advantages like even if extension=20
of websocket for new transport like SCTP will not impact SIP mechanism.
In case SIPWS URI looks like

    sipws:alice@example.com;transport=3Dtcp

As a theory, if websocket is extended for SCTP transport and then=20
SIPWS URI will looks like

    sipws:alice@example.com;transport=3Dsctp

Please note that TLS over SCTP exists already as RFC3436.=20

As we discussed, there was debate in RFC 3261 whether TLS is a transport=20
or URI in SIP, it was decided as URI mechanism, TLS transport is=20
deprecated and also there is no plan to change this stance as per=20
this mail thread. So IMO, it is more appropriate to define new URI=20
scheme like SIPWS rather than defining websocket as a transport.=20
Please let me know your opinion on the same.

Thanks
Partha


>-----Original Message-----
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>Behalf Of Paul Kyzivat
>Sent: Wednesday, April 25, 2012 1:06 AM
>To: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 4/24/12 8:28 AM, Kevin P. Fleming wrote:
>> On 04/24/2012 07:04 AM, Ravindran, Parthasarathi wrote:
>>> Paul,
>>>
>>> Thanks for clarify your view. In case your view is correct, then
>>> Usage of SIPS Scheme (Sec 9.2) of draft-ibc-sipcore-sip-websocket-02
>>> draft should forbid the usage of SIPS URI for websocket transport
>>> (WSS, WS) as SIPS URI is mistake for TLS, the mistake *MUST NOT* be
>>> repeated and the draft text has to be updated as
>>>
>>> "SIPS URI MUST NOT be used for WS, WSS transport as the end-to-end
>>> security is not assured"
>>
>> How did you reach this conclusion? In this context,
>> WebSocket-over-TLS-over-TCP is no different than TLS-over-TCP; usage
>> of WebSocket on the first hop does not change the ability of the path
>> to be end-to-end secure at all.
>
>It may be that WebSocket-over-TLS-over-TCP is *conceptually* no
>different than TLS-over-TCP. But its far from clear that this is so with
>the current wording of 3261 and 5630. I expect that a careful review of
>these needs to be done and some normative updates done. I went through
>3261 looking at discussions of sips for things that might be problematic
>or at least need some consideration. Here are some of them:
>
>
>    Any
>    implementation that supports TLS MUST support the SIPS URI scheme.
>    The To header field allows for a display name.
>
>    If the request arrived over TLS, and the Request-URI contained a
>SIPS
>    URI, the "secure" flag is set to TRUE.
>
>    If the request was sent over TLS, and the Request-URI contained a
>    SIPS URI, the "secure" flag is set to TRUE.
>
>    Furthermore, if the Request-URI contains a SIPS URI, TLS MUST
>    be used to communicate with that proxy.
>
>    For a SIPS URI, the transport parameter MUST
>    indicate a reliable transport.
>
>    When a request is sent to a SIPS URI, the
>    protocol still indicates "SIP", and the transport protocol is TLS.
>
>    When used as the Request-URI of a
>    request, the SIPS scheme signifies that each hop over which the
>    request is forwarded, until the request reaches the SIP entity
>    responsible for the domain portion of the Request-URI, must be
>    secured with TLS
>
>    The use of SIPS in particular entails that mutual TLS authentication
>    SHOULD be employed, as SHOULD the ciphersuite
>    TLS_RSA_WITH_AES_128_CBC_SHA.
>
>    Note that in the SIPS URI scheme, transport is independent of TLS,
>    and thus "sips:alice@atlanta.com;transport=3Dtcp" and
>    "sips:alice@atlanta.com;transport=3Dsctp" are both valid (although
>    note that UDP is not a valid transport for SIPS).  The use of
>    "transport=3Dtls" has consequently been deprecated, partly because
>    it was specific to a single hop of the request.  This is a change
>    since RFC 2543.
>
>    All SIP elements that support TLS MUST also support the SIPS URI
>    scheme.
>
>I'm not suggesting exactly *what* needs to be done - just that it needs
>to be figured out.
>
>As a consequence, I would expect that it would be possible to use sips
>with websockets, but no requirement to do so. Whether its helpful, or
>wise, to use sips is a separate question.
>
>	Thanks,
>	Paul
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

From saul@ag-projects.com  Wed Apr 25 03:36:37 2012
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <CALiegfkNBi7guNML-oAF5-OUAo2ZXAjXYDo_MShLc4SiOW_wOA@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228035@inba-mail01.sonusnet.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com> <FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228098@inba-mail01.sonusnet.com> <CALiegf=q1HE4bn1dgDz7RqKFe3NCDxr3vOWtR5DTevWAo0rK6A@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C42DF5E46@ESESSCMS0356.eemea.ericsson.se> <4F8ECB82.7070805@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228C1B@inba-mail01.sonusnet.com> <4F9460AC.2080605@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E231B57@inba-mail01.sonusnet.com> <4F956931.7090404@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E2380B2@inba-mail01.sonusnet.com> <4F969C7E.3070107@digium.com> <4F970097.5020002@alum.mit.edu> <387F9047F55E8C4285 0AD6B3A7A03C6C0E239306@inba-mail01.sonusnet.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Partha,

On Apr 25, 2012, at 11:18 AM, Ravindran, Parthasarathi wrote:

> Paul,
>=20
> As you mentioned, it is possible to use SIPS URI even for the new=20
> transport like websocket with some normative changes and those=20
> changes needs to be figured out. As a result, SIPS URI is not going
> to be mistake in the new transport as well.
>=20
> My intent of this mail is to make consistent mechanism for all SIP=20
> application layer transport protocol over TCP. "Websocket=20
> over TCP" is same like "TLS over TCP" and has some of similar=20
> behavior like
>=20

Can you elaborate on what consistent mechanism is needed and why?

> 1) Hop by hop behavior.=20
> 2) Running on top of TCP transport
> 3) No port management
>=20
> This mechanism has other added advantages like even if extension=20
> of websocket for new transport like SCTP will not impact SIP =
mechanism.
> In case SIPWS URI looks like
>=20
>    sipws:alice@example.com;transport=3Dtcp
>=20

So, what does this mean, that we need to reach alice through a WebSocket =
connection? Should alice be able to register using such contact?

> As a theory, if websocket is extended for SCTP transport and then=20
> SIPWS URI will looks like
>=20
>    sipws:alice@example.com;transport=3Dsctp
>=20
> Please note that TLS over SCTP exists already as RFC3436.=20
>=20
> As we discussed, there was debate in RFC 3261 whether TLS is a =
transport=20
> or URI in SIP, it was decided as URI mechanism, TLS transport is=20
> deprecated and also there is no plan to change this stance as per=20
> this mail thread. So IMO, it is more appropriate to define new URI=20
> scheme like SIPWS rather than defining websocket as a transport.=20
> Please let me know your opinion on the same.
>=20

What should we do if tomorrow I send a draft specifying pseudo-TCP =
transport (TCP implemented over UDP) for SIP? Yet another URI scheme? =
Not to mention that all equipment out there would go nuts os most likely =
just ignore URI schemes which are not 'sip' or 'sips'.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From pkyzivat@alum.mit.edu  Wed Apr 25 07:55:05 2012
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References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com> <52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com> <FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228098@inba-mail01.sonusnet.com> <CALiegf=q1HE4bn1dgDz7RqKFe3NCDxr3vOWtR5DTevWAo0rK6A@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C42DF5E46@ESESSCMS0356.eemea.ericsson.se> <4F8ECB82.7070805@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C0E228C1B@inba-mail01.sonusnet.com> <4F9460AC.2080605@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E231B57@inba-mail01.sonusnet.com> <4F956931.7090404@alum.mit.edu> <387F9047F55E8C42850AD6B3A7A03C6C0E2380B2@inba-mail01.sonusnet.com> <4F969C7E.3070107@digium.com> <4F970097.5020002@alum.mit.edu> <387F9047F55E8C4285 0AD6B3A7A03C6C0E239306@inba-mail01.sonusnet.com> <5522E7CB-78FF-4582-8E39-0E8E43301650@ag-projects.com>
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Cc: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Websocket is a new transport or new URI?
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I agree with Saśl here. I don't really understand what it is that Partha 
is proposing.

As I already stated, I think sips turned out to be a mistake - that it 
doesn't provide enough of a guarantee to be able to promise anything to 
an end user. But we investigated this ad nauseum and couldn't come up 
with anything substantially better.

So I'm ok to extend the applicability of sips to other transports that 
can provide at least as much security as TLS over TCP provides, for 
those that find that to be of *some* utility.

	Thanks,
	Paul

On 4/25/12 6:36 AM, Saśl Ibarra Corretgé wrote:
> Partha,
>
> On Apr 25, 2012, at 11:18 AM, Ravindran, Parthasarathi wrote:
>
>> Paul,
>>
>> As you mentioned, it is possible to use SIPS URI even for the new
>> transport like websocket with some normative changes and those
>> changes needs to be figured out. As a result, SIPS URI is not going
>> to be mistake in the new transport as well.
>>
>> My intent of this mail is to make consistent mechanism for all SIP
>> application layer transport protocol over TCP. "Websocket
>> over TCP" is same like "TLS over TCP" and has some of similar
>> behavior like
>>
>
> Can you elaborate on what consistent mechanism is needed and why?
>
>> 1) Hop by hop behavior.
>> 2) Running on top of TCP transport
>> 3) No port management
>>
>> This mechanism has other added advantages like even if extension
>> of websocket for new transport like SCTP will not impact SIP mechanism.
>> In case SIPWS URI looks like
>>
>>     sipws:alice@example.com;transport=tcp
>>
>
> So, what does this mean, that we need to reach alice through a WebSocket connection? Should alice be able to register using such contact?
>
>> As a theory, if websocket is extended for SCTP transport and then
>> SIPWS URI will looks like
>>
>>     sipws:alice@example.com;transport=sctp
>>
>> Please note that TLS over SCTP exists already as RFC3436.
>>
>> As we discussed, there was debate in RFC 3261 whether TLS is a transport
>> or URI in SIP, it was decided as URI mechanism, TLS transport is
>> deprecated and also there is no plan to change this stance as per
>> this mail thread. So IMO, it is more appropriate to define new URI
>> scheme like SIPWS rather than defining websocket as a transport.
>> Please let me know your opinion on the same.
>>
>
> What should we do if tomorrow I send a draft specifying pseudo-TCP transport (TCP implemented over UDP) for SIP? Yet another URI scheme? Not to mention that all equipment out there would go nuts os most likely just ignore URI schemes which are not 'sip' or 'sips'.
>
>
> Regards,
>
> --
> Saśl Ibarra Corretgé
> AG Projects
>
>
>
>


From frankwmiller@frankwmiller.net  Wed Apr 25 08:37:17 2012
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From: "Frank W. Miller" <frankwmiller@frankwmiller.net>
To: <sipcore@ietf.org>
References: <CALiegfmw+ouWqoVbsiMuq2CJ4TxkeREVVbw308FzwKpUUrw7tw@mail.gmail.com><52EA773E-0FA0-4625-8E1F-6D10E543137A@ag-projects.com><387F9047F55E8C42850AD6B3A7A03C6C0E228062@inba-mail01.sonusnet.com><FA41E993-B940-4FE6-9352-E9539E56A971@ag-projects.com><387F9047F55E8C42850AD6B3A7A03C6C0E228098@inba-mail01.sonusnet.com><CALiegf=q1HE4bn1dgDz7RqKFe3NCDxr3vOWtR5DTevWAo0rK6A@mail.gmail.com><7F2072F1E0DE894DA4B517B93C6A05852C42DF5E46@ESESSCMS0356.eemea.ericsson.se><4F8ECB82.7070805@ericsson.com><387F9047F55E8C42850AD6B3A7A03C6C0E228C1B@inba-mail01.sonusnet.com><4F9460AC.2080605@alum.mit.edu><387F9047F55E8C42850AD6B3A7A03C6C0E231B57@inba-mail01.sonusnet.com><4F956931.7090404@alum.mit.edu><387F9047F55E8C42850AD6B3A7A03C6C0E2380B2@inba-mail01.sonusnet.com><4F969C7E.3070107@digium.com> <4F970097.5020002@alum.mit.edu><387F9047F55E8C42850AD6B3A7A03C6C0E239306@inba-mail01.sonusnet.com><5522E7CB-78FF-4582-8E39-0E8E43301650@ag-projects.com> <4F981046.6010508@alum.mit.edu>
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Jeez, with all due respect, I've been following these hundreds of =
messages
that seem to be necessary to add support for Websocket as a transport.

Is this really what the original developers intended when they made the
statement that SIP is transport independent?  Doesn't seem very =
independent
to me.  In my mind independent to me means I switch from SOCK_DGRAM or
SOCK_STREAM to SOCK_WEBSOCKET (or whatever) in my socket() call and I'm
done.

FM

=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On =
Behalf
Of Paul Kyzivat
Sent: Wednesday, April 25, 2012 8:55 AM
To: Sa=FAl Ibarra Corretg=E9
Cc: Ravindran, Parthasarathi; sipcore@ietf.org
Subject: Re: [sipcore] Websocket is a new transport or new URI?

I agree with Sa=FAl here. I don't really understand what it is that =
Partha is
proposing.

As I already stated, I think sips turned out to be a mistake - that it
doesn't provide enough of a guarantee to be able to promise anything to =
an
end user. But we investigated this ad nauseum and couldn't come up with
anything substantially better.

So I'm ok to extend the applicability of sips to other transports that =
can
provide at least as much security as TLS over TCP provides, for those =
that
find that to be of *some* utility.

	Thanks,
	Paul

On 4/25/12 6:36 AM, Sa=FAl Ibarra Corretg=E9 wrote:
> Partha,
>
> On Apr 25, 2012, at 11:18 AM, Ravindran, Parthasarathi wrote:
>
>> Paul,
>>
>> As you mentioned, it is possible to use SIPS URI even for the new=20
>> transport like websocket with some normative changes and those=20
>> changes needs to be figured out. As a result, SIPS URI is not going=20
>> to be mistake in the new transport as well.
>>
>> My intent of this mail is to make consistent mechanism for all SIP=20
>> application layer transport protocol over TCP. "Websocket over TCP"=20
>> is same like "TLS over TCP" and has some of similar behavior like
>>
>
> Can you elaborate on what consistent mechanism is needed and why?
>
>> 1) Hop by hop behavior.
>> 2) Running on top of TCP transport
>> 3) No port management
>>
>> This mechanism has other added advantages like even if extension of=20
>> websocket for new transport like SCTP will not impact SIP mechanism.
>> In case SIPWS URI looks like
>>
>>     sipws:alice@example.com;transport=3Dtcp
>>
>
> So, what does this mean, that we need to reach alice through a =
WebSocket
connection? Should alice be able to register using such contact?
>
>> As a theory, if websocket is extended for SCTP transport and then=20
>> SIPWS URI will looks like
>>
>>     sipws:alice@example.com;transport=3Dsctp
>>
>> Please note that TLS over SCTP exists already as RFC3436.
>>
>> As we discussed, there was debate in RFC 3261 whether TLS is a=20
>> transport or URI in SIP, it was decided as URI mechanism, TLS=20
>> transport is deprecated and also there is no plan to change this=20
>> stance as per this mail thread. So IMO, it is more appropriate to=20
>> define new URI scheme like SIPWS rather than defining websocket as a
transport.
>> Please let me know your opinion on the same.
>>
>
> What should we do if tomorrow I send a draft specifying pseudo-TCP
transport (TCP implemented over UDP) for SIP? Yet another URI scheme? =
Not to
mention that all equipment out there would go nuts os most likely just
ignore URI schemes which are not 'sip' or 'sips'.
>
>
> Regards,
>
> --
> Sa=FAl Ibarra Corretg=E9
> AG Projects
>
>
>
>

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


From vkg@bell-labs.com  Wed Apr 25 09:05:03 2012
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On 04/25/2012 09:55 AM, Paul Kyzivat wrote:
> I agree with Saśl here. I don't really understand what it is that Partha
> is proposing.
>
> As I already stated, I think sips turned out to be a mistake - that it
> doesn't provide enough of a guarantee to be able to promise anything to
> an end user. But we investigated this ad nauseum and couldn't come up
> with anything substantially better.

Time for institutional memory, perhaps?  The long discussions (May -
June 2006) on why sips failed and ideas to come up with new URI schemes
to replace it and attendant problems (upcasting URIs, downcasting URIs
without user consent, hop-by-hop semantics versus end-to-end, etc.) is
summarized in
http://www.ietf.org/mail-archive/web/sip/current/msg24047.html

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
Web:   http://ect.bell-labs.com/who/vkg/

From pkyzivat@alum.mit.edu  Wed Apr 25 09:21:15 2012
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Hi Christer,

We are in agreement on many things. I have inserted further comments 
where I had something more to say.

On 4/25/12 5:06 AM, Christer Holmberg wrote:
> Hi Paul,
>
>> Much of the syntax (ABNF - section 4.2.2) is a replica of what is in RFC3840. IMO this relationship should be acknowledged. My preference would be to formally reference the abnf definitions from 3840 and use the same names for them.
>>
>> E.g.
>>
>>    feature-cap       =  ["+"] fcap-name [EQUAL LDQUOT (fcap-value-list
>>                          / fcap-string-value ) RDQUOT]
>>     fcap-name         =  ftag-name
>>     fcap-value-list   =  tag-value-list
>>     fcap-string-value =  string-value
>>     ;; ftag-name, tag-value-list, string-value defined in RFC3840
>
> I can do that.
>
> I was actually already planning to do that, but I thought that people would not want to reference the RFC3840 ABNF :)

Its a judgment call. In this case I think your intent is to keep the 
syntax the same, and the best way to ensure that is to have the syntax 
defined in only one place.

> ---------------------
>
>
>> Also, I am inclined to make the "+" mandatory, even though it really serves no purpose. The reason I propose this is to minimize confusion with the usage rules in 3840. In 3840 the "+" is mandatory except for the "grandfathered" tags. And we have had
>> trouble getting people to follow that. If we make it optional here it will further confuse things.
>>
>> OTOH, the reason for the + is so that feature tags can be distinguished from other contact parameters that are not feature tags. That need doesn't apply here, which is why the "+" is logically unnecessary. An alternative would be to *never* use the "+"
>> here. But I'm inclined to require the "+" because there may be some tags defined for use as both feature tags and proxy features, and then I'd rather not use the "+" in place and not in the other.
>
> I can make it mandatory for feature-caps, and e.g. add a note which points out the difference between feature-tags.
>
>
> ---------------------
>
>
>> I also find it some what confusing that part of the syntax is in this section while more of it is in section 5.4. I'd prefer to see these consolidated into a single syntax section.
>
> I think the Feature-Caps header field syntax, and the feature-caps syntax, shall be separated, but I can put them in the same section.

I don't see any particular reason to keep them separated,  especially if 
the syntax is simplified by reference to 3840. But I don't have a 
problem with separating it into two subsections, as long as they are 
together in the document.

> ---------------------
>
>
>> Re Registration Trees:
>>
>> There is no mention of an IETF tree, analogous to that defined in 2506 for feature tags. Given the history, and a desire to provide a similar structure, I agree it makes sense that we aren't planning on defining anything of that sort. But I think it is confusing
>> to anyone who has looked at 2506 that there is no mention of names that have only one facet.
>>
>> So, I think it would be helpful to explicitly state that names consisting of only a single facet, as well as names with initial facets other than "g" and "sip" are out of scope for this draft. I'm undecided if it would be good to define an IETF tree without any
>> content, or if it would be better not to define it.
>
> I actually had some text about that, saying that the reason we only define "g" and "sip" is  those are the only ones used for SIP related feature-tags today (and, compared to feature-tags, the feature-caps mechanism is only defined for SIP).
>
> For some reason that text didn't get into the -01 draft, but I will add it to the next version.
>
>
> ---------------------
>
>
>> The statements:
>>
>>     The tree definitions are based on the global tree and sip tree
>>     defined for media feature tags, in RFC 2506 [RFC2506] and RFC 3841
>>     [RFC3841].
>> and
>>     The SIP feature cap global tree is based on the media feature tag
>>     global tree defined in RFC 2506 [RFC2506].
>> and
>>     The SIP feature cap sip tree is based on the media feature tag sip
>>     tree defined in RFC 3840 [RFC3840].
>>
>> seem a little confusing to me, because its not clear what "based on"
>> means. I suggest changing this to "similar to".
>
> Ok.
>
>
> ---------------------
>
>
>> As far as defining new trees, the current text is:
>>
>>     Additional feature caps trees can be created by IANA, following the
>>     same rules and procedures as defined for media feature tags in
>>     section 3.1.4 of RFC 2506 [RFC2506].
>>
>> This in effect makes a normative dependency between this document and RFC2506. I think it would be better to sever the formal tie. Instead, this can be accomplished by adding something similar to section 3.1.4 of 2506.
>
> Ok.
>
>
> ---------------------
>
>
>> Regarding the IANA Registration Template:
>>
>>     | The reference MUST contain the information listed in
>>     | section XX of XXXX (IANA: Replace XXXX with assigned
>>     | RFC number of this specification
>>
>> I think this should say:
>>
>>     | The referenced specification MUST contain the information listed
>>     | in section XX of XXXX (IANA: Replace XXXX with assigned
>>     | RFC number of this specification
>
> Correct.
>
>
> ---------------------
>
>
>> In addition there may be some redundancy between the "SIP feature cap specification reference" and the "Related standards or documents". Do you think we need both?
>
> Probably not, because it is recommended that the SIP feature cap specification contains references to usages and examples.
>
>
> ---------------------
>
>
>> Are there other things in there that are just baggage being carried over from 2506 that are not useful to us? (E.g. do we need the publication
>> delay?)
>
> I don't know. Unless someone wants it, I can remove it.

Please also consider if there are any others that make no sense here and 
can be removed.

> ---------------------
>
>
>> Section 4.5.2 says:
>>
>>     The SIP feature cap specification MUST contain an overall description
>>     of the SIP feature cap: how it is used to indicate support of a
>>     feature, a description of the feature associated with the SIP feature
>>     cap, and a description of any additional information (conveyed using
>>     one or more SIP feature cap values) that can be conveyed together
>>     with the SIP feature cap.
>
> What this doesn't talk about is how the feature associated with the feature cap is exercised/invoked. So I would suggest something like:
>
>>     The SIP feature cap specification MUST contain an overall description
>>     of the SIP feature cap: how it is used to indicate support of a
>>     feature, a description of the feature associated with the SIP feature
>>     cap, a description of any additional information (conveyed using
>>     one or more SIP feature cap values) that can be conveyed together
>>     with the SIP feature cap, and a description of how the associated
>>     feature may be exercised/invoked.
>
> Looks good.
>
>
> ---------------------
>
>
>> Section 4.5.3:
>>
>> The rules described here sound to me like ones suitable for the name of a parameter (e.g. a parameter bound to an event or a mime type, or to a sip header.) It doesn't seem very appropriate when discussing the representation of a *value* of a
>> parameter/feature. For example, when values are enumerations, we might find things like "true" and "false" or "yes" and "no". Even if they are "red" / "green" / "blue", the same values might well make sense associated with other features. Can you
>> please think this through further? It may indeed make sense to say *something* here, but I imagine something weaker.
>
> Maybe we should simply say that the value is only applicable for the feature-cap for which it has been defined. For other feature-caps, the value has to be defined explicitly for those, even if the name and/or semantics would be the same.
>
>
> ---------------------
>
>
>> Section 4.5.5 says:
>>
>>     NOTE: Sometimes a SIP feature cap designer might choose to not reveal
>>     the implementation details of a SIP feature cap.  However, in order
>>     to allow multiple implementations to support the SIP feature cap,
>>     designers are strongly encouraged to provide the implementation
>>     details.
>>
>> I recognize that people may not want to publish implementation details.
>> But that is different from providing enough specification to all independent but interoperable implementations to be developed.
>>
>> The amount that needs to be disclosed might vary based on which registration tree is used. I would expect that the sip tree would require sufficient spec to support an independent implementation. If you were instituting a URI tree then I can see that it>  might be considered entirely proprietary and hence might not require this. (But you have omitted the URI tree, which thus sidesteps this issue.) ISTM that the global tree should still require a definition of how to create an interoperable implementation.>  The difference between the global tree and the sip tree in this regard might be that in the case of the sip tree the referenced specification must be an IETF document, while in the case of the global tree it could be some external document.
>
> I took the text from the Info-Package RFC.
>
> To be honest, however, I have some troubles to understand what is really meant by "implementation details". As described in section 4.5.2, you DO have to describe how the feature-cap is used etc, so I am not sure what extra "implementation details" would bring.

I was about to suggest we consult the authors of the Info-Package RFC, 
but then I realized that you are one of the authors. :-)
So, what did you mean?

In the case of Info packages the point is that you are delivering an 
identified package of information, and so I would think that the 
implementation details would include any sort of "protocol" layered on 
top of that data - e.g. when I send certain info to you I expect that 
you will send back another info with an ack/nack. IOW, whatever is 
needed to successfully use this. (In some cases it may be very simple - 
send anything syntactically valid for the package whenever you want, 
with receiver free to do anything or nothing with it.

> So, maybe we could simply remove section 4.5.5?

Lets not give up just yet, before we understand.
Again, my point is to specify enough so that independent developers can 
create interoperable implementations.

> ---------------------
>
>
>> Section 5.2.1 says:
>>
>>     When the UA receives the SIP request or the response, the SIP feature
>>     caps in the topmost Feature-Caps header field will represent the
>>     supported features "closest" to the UA.
>>
>> I don't think this is quite what you mean. I think you mean:
>>
>>     When the UA receives the SIP request or the response, the SIP feature
>>     caps in the first fc-value will represent the supported features
>>     "closest" to the UA.
>
> Yes.
>
>
> ---------------------
>
>
> Section 5.2.2 says:
>
>>     The procedures in this section applies to UAs that are part of
>>     B2BUAs, but where the URI in the Contact header field does not
>>    represent the UA, because the B2BUA is also acting as a proxy and
>>     inserts its URI e.g. in a Record-Route header field.
>>
>> IMO this covers some controversial ground, but probably only in ways that many will consider pedantic. We are in quite under specified territory when we talk about B2BUAs that act as Proxies. Does this mean that it was a proxy when the dialog was
>> established, but later the same entity redeclares itself as a UA for some subsequent transaction within the dialog? Or was it acting as a UA when the dialog was established, but still chose to use a Contact URI that doesn't identify itself?
>>
>> As much as I would like to have an extended discussion on this, doing so while working on this draft is probably counterproductive. Perhaps you can humor my sensitivities and just change this to:
>>
>>     The procedures in this section applies to UAs that are part of
>>     B2BUAs that are referenced in the message by a Record-Route header
>>     field rather than by the Contact URI.
>
> Looks good.
>
>
> ---------------------
>
>
>> Section 5.2.2 then says:
>>
>>     When a UA sends a SIP request, if the UA wants to indicate support of
>>     features towards its downstream SIP entities, it adds a Feature-Caps
>>     header field to the request, together with one or more Feature Caps
>>     associated with the supported features, before it forwards the
>>    request.
>>
>> This says "adds" but doesn't say whether to be beginning or the end.
>> Also, it isn't strictly necessary to add another header field, it can add to an existing header field. So I suggest:
>>
>>     When a UA sends a SIP request, if the UA wants to indicate support of
>>     features towards its downstream SIP entities, before it forwards the
>>    request it adds a fc-value with the supported features to
>>     the request, after any others that may be present. This may be added
>>     to the last existing Feature-Caps header field or as a new
>>     Feature-Caps header field after any others.
>>
>> Something similar applies a couple more times in this section and in sections 5.2.3 and 5.2.4. I realize this is a bit cumbersome - perhaps there is a more straightforward way to say it.
>
> I will double check, but I THINK we normally only talk about adding a value to an existing header field. The ABNF rules then shows whether multiple instances of the header field is allowed.
>
> We just need to be sure that the text is clear on which features are "closest" to the UA, no matter whether they are indicated in a single header field, or in multiple header fields.

Yes, I think we do usually do that, even though it is imprecise.
Within reason I'm ok with you following precedent of something with 
similar behavior. Record-Route comes to mind.

> ---------------------
>
>
>> Section 5.3.3 says:
>>
>>     If a SIP feature cap is inserted in a Feature-Caps header field of a
>>     SIP REGISTER request, or within a response of such request, it
>>     indicates to the receivers of the request (or response) that the
>>     feature associated with the SIP feature cap is supported for the
>>     duration of the registration, and for all SIP transactions associated
>>     with the registration, until the registration is re-freshed or
>>     terminated.
>>
>> There are some cases we need to discuss here. When a UA sends a REGISTER request, it could have no contacts, one contact, or several. What do the feature caps mean in these cases?
>>
>> - if there are no contacts in the register, are feature-caps irrelevant?
>> - if there is more than one contact in the register, do the feature-caps
>>    apply to all of them?
>>
>> Then the response should contain all the registered contacts, not just those mentioned in the request. So do the feature caps mentioned in the reply apply until the last of those contacts expire? Or should the requester assume they only apply until *its*
>> contacts expire?
>>
>>     Unless a SIP feature cap is inserted in a Feature-Caps header field
>>     or a re-registration, or within a response of such request, it
>>     indicates to the receivers of the request (or response) that the
>>     feature is no long supported for the registration.
>>
>> Suppose a UA registers with a contact and some feature caps. Then later it sends a register poll (register with no contacts). Can I assume that this will not affect the feature caps associated with the registered contact?
>
> It's up to the entities that previously inserted the feature-caps to decide whether they will continue to indicate support when they receive a re-registration request/response - no matter whether it's a poll, addition of contacts etc.
>
> Personally I see no reason why they should *not* insert the same feature-caps in a pure poll request/response, but we should allow for cases where the feature is no longer supported when the poll request/response is received.

I think you missed my point. Its really the same point as my question of 
what it means to put feature caps into a polling register.

Since you want the registrar to consider the received feature caps to be 
stateful for the duration of the register, I want to know what that 
state is attached to. ISTM that it must be attached to particular 
contacts. Then, a polling register would presumably not update the state 
of any registered contact since it doesn't update any of them.

On the flip side of that, if the registrar returns feature caps, what 
does the recipient attach that state to? The only thing it could attach 
it to that has a lifetime is the registration of a particular contact, 
since the registered contacts can expire independently.

> ---------------------
>
>
>> Do the enhanced registrar behaviors in outbound require any special treatment relative to feature caps?
>
> No, I don't think so.
>
> However, I guess we could add a clarification note saying that features are only supported for the specific outbound registration flow where support has been explicitly indicated.

I'm only asking that you look at it. If nothing is needed, that is fine.
The clarification sounds good if it is true. (I don't know if it is true.)

>> Section 6:
>>
>> There are a lot of missing IANA considerations - the establishment of the new trees as defined in 4.3 and 4.4.
>
> I used RFC 2506 as base, where the trees are defined more or less in the same way as in the draft. But, I guess it's better to do it in the same way as in RFC 3840.

Again, we are establishing a new IANA registry here.
I think it unwise to couple it to the definition of a different registry 
in a document we don't control. (Some day 2506 may be updated, and we 
wouldn't want that to affect *this* registry.)

In any case, IANA needs to be given explicit instructions to establish 
the new registry, and couple it to the template.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Wed Apr 25 09:38:39 2012
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Date: Wed, 25 Apr 2012 12:38:36 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, "sipcore@ietf.org" <sipcore@ietf.org>, =?ISO-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Vijay,

Thanks for this reference! Its one thing to have a recollection of a 
discussion long ago, but something else to have the minutes of that 
discussion!

	Thanks,
	Paul

On 4/25/12 12:09 PM, Vijay K. Gurbani wrote:
> On 04/25/2012 09:55 AM, Paul Kyzivat wrote:
>> I agree with Saśl here. I don't really understand what it is that Partha
>> is proposing.
>>
>> As I already stated, I think sips turned out to be a mistake - that it
>> doesn't provide enough of a guarantee to be able to promise anything to
>> an end user. But we investigated this ad nauseum and couldn't come up
>> with anything substantially better.
>
> Time for institutional memory, perhaps? The long discussions (May -
> June 2006) on why sips failed and ideas to come up with new URI schemes
> to replace it and attendant problems (upcasting URIs, downcasting URIs
> without user consent, hop-by-hop semantics versus end-to-end, etc.) is
> summarized in
> http://www.ietf.org/mail-archive/web/sip/current/msg24047.html
>
> - vijay


From ibc@aliax.net  Wed Apr 25 10:25:36 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 25 Apr 2012 19:25:14 +0200
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Hi Partha,


2012/4/25 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> So IMO, it is more appropriate to define new URI
> scheme like SIPWS rather than defining websocket as a transport.


The SIP transport defined by this draft is WebSocket, instead of TCP.
WebSocket (RFC 6455) is defined to work on top of TCP but, even if a
future spec defines WebSocket to run over SCTP, the SIP transport
would still be "WebSocket" (or perhaps "WebSocket-SCTP" but that
something in which I won't waste time now given the fact that RFC 6455
*mandates* TCP layer-3 transport and does not leave the door open for
other layer-3 transports).

WebSocket is, indeed, a layer on top of a network transport (layer 3)
but it's designed to work as a *transport* layer since it defines a
framing mechanism to send application messages and control mesages in
both directions. Applications (i.e. JavaScript scripts, PHP/Java web
apps) use WebSocket as a transport layer.

You can find lot of cases in which a transport runs on top of other
transports. For example, in the recent IETF #83 the RTCWEB WG approved
(or discussed) SCTP-over-DTLS-over-ICE-over-UDP as the transport for
the WebRTC Data-Channel mechanism. A "transport" does not necessarily
mean "layer 3 network transport".



About SIPS, the last S in SIPS stands for "secure" regardless of the
undelying SIP transport used. AFAIK it does not necessarily mean
"TLS". Take for example draft-jennings-sip-dtls-05 [1] which defines
SIP over DTLS over UDP. The draf is expired, but anyhow, it defines a
secure SIP transport which does not use TLS but DTLS (which is similar
but NOT the same). The draft also allows SIPS messages through DTLS
over UDP.

[1] http://tools.ietf.org/html/draft-jennings-sip-dtls-05


When a SIPS message is carried on top of a secure WebSocket connection
(which means SIP over WebSocket over TLS over TCP) the security level
is basically the same as if it was SIP over TLS over TCP. Therefore I
consider that a secure WebSocket connection fits the requirements of
the SIPS mechanism, thus secure WebSocket should be considered as a
secure SIP transport.



About the SIPWS and SIPWSS schemas you propose, don't take me wrong
but I consider that the worst possible idea. First of all because I
don't agree with your rationale (I insist that, from the point of view
of a SIP stack, WebSocket is the ***SIP transport***, instead of TCP).
Secondly, because adding new URI schemas would just break the existing
SIP world (there are still some SIP entities that don't understand
SIPS schema and you are proposing two new ones?).


Thanks a lot for your comments.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From christer.holmberg@ericsson.com  Wed Apr 25 13:17:38 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>, "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
Date: Wed, 25 Apr 2012 22:17:34 +0200
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Hi,

I agree with Inaki.

Whatever "layer" we consider the WebSocket to be, in my opinion the URI sch=
eme is used for addressing (either as an AOR or a contact), and should be i=
ndependent of the transport.

For what would the new scheme be needed  anyway?

Regards,

Christer




________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of I=F1=
aki Baz Castillo [ibc@aliax.net]
Sent: Wednesday, April 25, 2012 8:25 PM
To: Ravindran, Parthasarathi
Cc: sipcore@ietf.org
Subject: Re: [sipcore] Websocket is a new transport or new URI?

Hi Partha,


2012/4/25 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> So IMO, it is more appropriate to define new URI
> scheme like SIPWS rather than defining websocket as a transport.


The SIP transport defined by this draft is WebSocket, instead of TCP.
WebSocket (RFC 6455) is defined to work on top of TCP but, even if a
future spec defines WebSocket to run over SCTP, the SIP transport
would still be "WebSocket" (or perhaps "WebSocket-SCTP" but that
something in which I won't waste time now given the fact that RFC 6455
*mandates* TCP layer-3 transport and does not leave the door open for
other layer-3 transports).

WebSocket is, indeed, a layer on top of a network transport (layer 3)
but it's designed to work as a *transport* layer since it defines a
framing mechanism to send application messages and control mesages in
both directions. Applications (i.e. JavaScript scripts, PHP/Java web
apps) use WebSocket as a transport layer.

You can find lot of cases in which a transport runs on top of other
transports. For example, in the recent IETF #83 the RTCWEB WG approved
(or discussed) SCTP-over-DTLS-over-ICE-over-UDP as the transport for
the WebRTC Data-Channel mechanism. A "transport" does not necessarily
mean "layer 3 network transport".



About SIPS, the last S in SIPS stands for "secure" regardless of the
undelying SIP transport used. AFAIK it does not necessarily mean
"TLS". Take for example draft-jennings-sip-dtls-05 [1] which defines
SIP over DTLS over UDP. The draf is expired, but anyhow, it defines a
secure SIP transport which does not use TLS but DTLS (which is similar
but NOT the same). The draft also allows SIPS messages through DTLS
over UDP.

[1] http://tools.ietf.org/html/draft-jennings-sip-dtls-05


When a SIPS message is carried on top of a secure WebSocket connection
(which means SIP over WebSocket over TLS over TCP) the security level
is basically the same as if it was SIP over TLS over TCP. Therefore I
consider that a secure WebSocket connection fits the requirements of
the SIPS mechanism, thus secure WebSocket should be considered as a
secure SIP transport.



About the SIPWS and SIPWSS schemas you propose, don't take me wrong
but I consider that the worst possible idea. First of all because I
don't agree with your rationale (I insist that, from the point of view
of a SIP stack, WebSocket is the ***SIP transport***, instead of TCP).
Secondly, because adding new URI schemas would just break the existing
SIP world (there are still some SIP entities that don't understand
SIPS schema and you are proposing two new ones?).


Thanks a lot for your comments.


--
I=F1aki Baz Castillo
<ibc@aliax.net>
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore=

From christer.holmberg@ericsson.com  Wed Apr 25 13:57:56 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Wed, 25 Apr 2012 22:57:52 +0200
Thread-Topic: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Hi Paul,

Comments, and a correction to one of my own previous comments, inline.

>>> Much of the syntax (ABNF - section 4.2.2) is a replica of what is in RF=
C3840. IMO this relationship should be acknowledged. My preference would be=
 to formally reference the abnf definitions from 3840 and use the same name=
s for them.
>>>
>>> E.g.
>>>
>>>    feature-cap       =3D  ["+"] fcap-name [EQUAL LDQUOT (fcap-value-lis=
t
>>>                          / fcap-string-value ) RDQUOT]
>>>     fcap-name         =3D  ftag-name
>>>     fcap-value-list   =3D  tag-value-list
>>>     fcap-string-value =3D  string-value
>>>     ;; ftag-name, tag-value-list, string-value defined in RFC3840
>>
>> I can do that.
>>
>> I was actually already planning to do that, but I thought that people wo=
uld not want to reference the RFC3840 ABNF :)
>
> Its a judgment call. In this case I think your intent is to keep the
> syntax the same, and the best way to ensure that is to have the syntax
> defined in only one place.

Ok.


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


>>> I also find it some what confusing that part of the syntax is in this s=
ection while more of it is in section 5.4. I'd prefer to see these consolid=
ated into a single syntax section.
>>
>> I think the Feature-Caps header field syntax, and the feature-caps synta=
x, shall be separated, but I can put them in the same section.
>
> I don't see any particular reason to keep them separated,  especially if
> the syntax is simplified by reference to 3840. But I don't have a
> problem with separating it into two subsections, as long as they are
> together in the document.

I'll keep them separate for now, but move them to the same place.


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


>>> Are there other things in there that are just baggage being carried ove=
r from 2506 that are not useful to us? (E.g. do we need the publication
>>> delay?)
>>
>> I don't know. Unless someone wants it, I can remove it.
>
> Please also consider if there are any others that make no sense here and =
can be removed.

I did already remove some in -01, but I'll check whether there is something=
 else that could also be removed.


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


>>> Section 4.5.5 says:
>>>
>>>     NOTE: Sometimes a SIP feature cap designer might choose to not reve=
al
>>>     the implementation details of a SIP feature cap.  However, in order
>>>     to allow multiple implementations to support the SIP feature cap,
>>>     designers are strongly encouraged to provide the implementation
>>>     details.
>>>
>>> I recognize that people may not want to publish implementation details.
>>> But that is different from providing enough specification to all indepe=
ndent but interoperable implementations to be developed.
>>>
>>> The amount that needs to be disclosed might vary based on which registr=
ation tree is used. I would expect that the sip tree would require sufficie=
nt spec to support an=20
>>> independent implementation. If you were instituting a URI tree then I c=
an see that it might be considered entirely proprietary and hence might=20
>>> not require this. (But you have omitted the URI tree, which thus sidest=
eps this issue.) ISTM that the global tree should still require a definitio=
n of how to create an interoperable implementation.
>>> The difference between the global tree and the sip tree in this regard =
might be that in the case of the sip tree the >>> referenced specification =
must be an IETF=20
>>> document, while in the case of the global tree it could be some externa=
l document.
>>
>> I took the text from the Info-Package RFC.
>>
>> To be honest, however, I have some troubles to understand what is really=
 meant by "implementation details". As described in section 4.5.2, you DO h=
ave to describe how the feature-cap is
>> used etc, so I am not sure what extra "implementation details" would bri=
ng.
>
> I was about to suggest we consult the authors of the Info-Package RFC,
> but then I realized that you are one of the authors. :-)
> So, what did you mean?
>
> In the case of Info packages the point is that you are delivering an
> identified package of information, and so I would think that the
> implementation details would include any sort of "protocol" layered on
> top of that data - e.g. when I send certain info to you I expect that
> you will send back another info with an ack/nack. IOW, whatever is
> needed to successfully use this. (In some cases it may be very simple -
> send anything syntactically valid for the package whenever you want,
> with receiver free to do anything or nothing with it.
>
>> So, maybe we could simply remove section 4.5.5?
>
> Lets not give up just yet, before we understand.
> Again, my point is to specify enough so that independent developers can
> create interoperable implementations.


When looking at RFC 6086, I think the "Implementation Details" part contain=
s stuff that we have put in the "General Description" part. The "General De=
scription" part of=20
RFC 6086 does not talk about providing details on how an Info-Package is us=
ed, only what type of information it carries, and what type of applications=
 are using it.


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


>>> Section 5.2.2 then says:
>>>
>>>     When a UA sends a SIP request, if the UA wants to indicate support =
of
>>>     features towards its downstream SIP entities, it adds a Feature-Cap=
s
>>>     header field to the request, together with one or more Feature Caps
>>>     associated with the supported features, before it forwards the
>>>    request.
>>>
>>> This says "adds" but doesn't say whether to be beginning or the end.
>>> Also, it isn't strictly necessary to add another header field, it can a=
dd to an existing header field. So I suggest:
>>>
>>>     When a UA sends a SIP request, if the UA wants to indicate support =
of
>>>     features towards its downstream SIP entities, before it forwards th=
e
>>>    request it adds a fc-value with the supported features to
>>>     the request, after any others that may be present. This may be adde=
d
>>>     to the last existing Feature-Caps header field or as a new
>>>     Feature-Caps header field after any others.
>>
>>> Something similar applies a couple more times in this section and in se=
ctions 5.2.3 and 5.2.4. I realize this is a bit cumbersome - perhaps there =
is a more straightforward way to say it.
>>
>> I will double check, but I THINK we normally only talk about adding a va=
lue to an existing header field. The ABNF rules then shows whether multiple=
 instances of the header field is allowed.
>>
>> We just need to be sure that the text is clear on which features are "cl=
osest" to the UA, no matter whether they are indicated in a single header f=
ield, or in multiple header fields.
>
>Yes, I think we do usually do that, even though it is imprecise.
>Within reason I'm ok with you following precedent of something with
> similar behavior. Record-Route comes to mind.

Yes, and this is where I need to correct myself.

When a proxy adds Record-Route (or Via), it adds its value *before* any exi=
sting values, and the idea is the same for feature-caps. That is indicated =
in section 5.2.4 (see below), but=20
I guess should be in a general section.

   "When a proxy adds a Feature-Caps header field to a SIP message, it
   MUST place the header field before any existing Feature-Caps header
   fields in the request."

   (s/request/message)


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


>>> Section 5.3.3 says:
>>>
>>>     If a SIP feature cap is inserted in a Feature-Caps header field of =
a
>>>     SIP REGISTER request, or within a response of such request, it
>>>     indicates to the receivers of the request (or response) that the
>>>     feature associated with the SIP feature cap is supported for the
>>>     duration of the registration, and for all SIP transactions associat=
ed
>>>     with the registration, until the registration is re-freshed or
>>>     terminated.
>>>
>>> There are some cases we need to discuss here. When a UA sends a REGISTE=
R request, it could have no contacts, one contact, or several. What do the =
feature caps mean in these cases?
>>>
>>> - if there are no contacts in the register, are feature-caps irrelevant=
?
>>> - if there is more than one contact in the register, do the feature-cap=
s
>>>    apply to all of them?
>>>
>>> Then the response should contain all the registered contacts, not just =
those mentioned in the request. So do the feature caps mentioned in the rep=
ly apply until the last of those=20
>>> contacts expire? Or should the requester assume they only apply until *=
its* contacts expire?
>>>
>>>     Unless a SIP feature cap is inserted in a Feature-Caps header field
>>>     or a re-registration, or within a response of such request, it
>>>     indicates to the receivers of the request (or response) that the
>>>     feature is no long supported for the registration.
>>>
>>> Suppose a UA registers with a contact and some feature caps. Then later=
 it sends a register poll (register with no contacts). Can I assume that th=
is will not affect the feature caps associated with the registered contact?
>>
>> It's up to the entities that previously inserted the feature-caps to dec=
ide whether they will continue to indicate support when they receive a re-r=
egistration request/response - no matter whether it's a poll, addition of c=
ontacts etc.
>>
>> Personally I see no reason why they should *not* insert the same feature=
-caps in a pure poll request/response, but we should allow for cases where =
the feature is no longer supported when the poll request/response is receiv=
ed.
>
> I think you missed my point. Its really the same point as my question of
>what it means to put feature caps into a polling register.
>
> Since you want the registrar to consider the received feature caps to be
> stateful for the duration of the register, I want to know what that
> state is attached to. ISTM that it must be attached to particular
> contacts. Then, a polling register would presumably not update the state
> of any registered contact since it doesn't update any of them.
>
> On the flip side of that, if the registrar returns feature caps, what
> does the recipient attach that state to? The only thing it could attach
> it to that has a lifetime is the registration of a particular contact,
> since the registered contacts can expire independently.

Ok, I think I understand. I need to think a little about it.


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


>>> Do the enhanced registrar behaviors in outbound require any special tre=
atment relative to feature caps?
>>
>> No, I don't think so.
>>
>> However, I guess we could add a clarification note saying that features =
are only supported for the specific outbound registration flow where suppor=
t has been explicitly indicated.
>
> I'm only asking that you look at it. If nothing is needed, that is fine.
> The clarification sounds good if it is true. (I don't know if it is true.=
)

I guess this is related to the issue above.


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


>>> Section 6:
>>>
>>> There are a lot of missing IANA considerations - the establishment of t=
he new trees as defined in 4.3 and 4.4.
>>
>> I used RFC 2506 as base, where the trees are defined more or less in the=
 same way as in the draft. But, I guess it's better to do it in the same wa=
y as in RFC 3840.
>
> Again, we are establishing a new IANA registry here.
> I think it unwise to couple it to the definition of a different registry
> in a document we don't control. (Some day 2506 may be updated, and we
> wouldn't want that to affect *this* registry.)

What I meant was that RFC 2506 does not define its registry in an IANA Cons=
iderations sections.

> In any case, IANA needs to be given explicit instructions to establish th=
e new registry, and couple it to the template.

Yes.

Regards,

Christer=

From pkyzivat@alum.mit.edu  Wed Apr 25 14:20:02 2012
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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This all seems good to me.

	Thanks,
	Paul

On 4/25/12 4:57 PM, Christer Holmberg wrote:
> Hi Paul,
>
> Comments, and a correction to one of my own previous comments, inline.
>
>>>> Much of the syntax (ABNF - section 4.2.2) is a replica of what is in RFC3840. IMO this relationship should be acknowledged. My preference would be to formally reference the abnf definitions from 3840 and use the same names for them.
>>>>
>>>> E.g.
>>>>
>>>>     feature-cap       =  ["+"] fcap-name [EQUAL LDQUOT (fcap-value-list
>>>>                           / fcap-string-value ) RDQUOT]
>>>>      fcap-name         =  ftag-name
>>>>      fcap-value-list   =  tag-value-list
>>>>      fcap-string-value =  string-value
>>>>      ;; ftag-name, tag-value-list, string-value defined in RFC3840
>>>
>>> I can do that.
>>>
>>> I was actually already planning to do that, but I thought that people would not want to reference the RFC3840 ABNF :)
>>
>> Its a judgment call. In this case I think your intent is to keep the
>> syntax the same, and the best way to ensure that is to have the syntax
>> defined in only one place.
>
> Ok.
>
>
> ---------------------
>
>
>>>> I also find it some what confusing that part of the syntax is in this section while more of it is in section 5.4. I'd prefer to see these consolidated into a single syntax section.
>>>
>>> I think the Feature-Caps header field syntax, and the feature-caps syntax, shall be separated, but I can put them in the same section.
>>
>> I don't see any particular reason to keep them separated,  especially if
>> the syntax is simplified by reference to 3840. But I don't have a
>> problem with separating it into two subsections, as long as they are
>> together in the document.
>
> I'll keep them separate for now, but move them to the same place.
>
>
> ---------------------
>
>
>>>> Are there other things in there that are just baggage being carried over from 2506 that are not useful to us? (E.g. do we need the publication
>>>> delay?)
>>>
>>> I don't know. Unless someone wants it, I can remove it.
>>
>> Please also consider if there are any others that make no sense here and can be removed.
>
> I did already remove some in -01, but I'll check whether there is something else that could also be removed.
>
>
> ---------------------
>
>
>>>> Section 4.5.5 says:
>>>>
>>>>      NOTE: Sometimes a SIP feature cap designer might choose to not reveal
>>>>      the implementation details of a SIP feature cap.  However, in order
>>>>      to allow multiple implementations to support the SIP feature cap,
>>>>      designers are strongly encouraged to provide the implementation
>>>>      details.
>>>>
>>>> I recognize that people may not want to publish implementation details.
>>>> But that is different from providing enough specification to all independent but interoperable implementations to be developed.
>>>>
>>>> The amount that needs to be disclosed might vary based on which registration tree is used. I would expect that the sip tree would require sufficient spec to support an
>>>> independent implementation. If you were instituting a URI tree then I can see that it might be considered entirely proprietary and hence might
>>>> not require this. (But you have omitted the URI tree, which thus sidesteps this issue.) ISTM that the global tree should still require a definition of how to create an interoperable implementation.
>>>> The difference between the global tree and the sip tree in this regard might be that in the case of the sip tree the>>>  referenced specification must be an IETF
>>>> document, while in the case of the global tree it could be some external document.
>>>
>>> I took the text from the Info-Package RFC.
>>>
>>> To be honest, however, I have some troubles to understand what is really meant by "implementation details". As described in section 4.5.2, you DO have to describe how the feature-cap is
>>> used etc, so I am not sure what extra "implementation details" would bring.
>>
>> I was about to suggest we consult the authors of the Info-Package RFC,
>> but then I realized that you are one of the authors. :-)
>> So, what did you mean?
>>
>> In the case of Info packages the point is that you are delivering an
>> identified package of information, and so I would think that the
>> implementation details would include any sort of "protocol" layered on
>> top of that data - e.g. when I send certain info to you I expect that
>> you will send back another info with an ack/nack. IOW, whatever is
>> needed to successfully use this. (In some cases it may be very simple -
>> send anything syntactically valid for the package whenever you want,
>> with receiver free to do anything or nothing with it.
>>
>>> So, maybe we could simply remove section 4.5.5?
>>
>> Lets not give up just yet, before we understand.
>> Again, my point is to specify enough so that independent developers can
>> create interoperable implementations.
>
>
> When looking at RFC 6086, I think the "Implementation Details" part contains stuff that we have put in the "General Description" part. The "General Description" part of
> RFC 6086 does not talk about providing details on how an Info-Package is used, only what type of information it carries, and what type of applications are using it.
>
>
> ---------------------
>
>
>>>> Section 5.2.2 then says:
>>>>
>>>>      When a UA sends a SIP request, if the UA wants to indicate support of
>>>>      features towards its downstream SIP entities, it adds a Feature-Caps
>>>>      header field to the request, together with one or more Feature Caps
>>>>      associated with the supported features, before it forwards the
>>>>     request.
>>>>
>>>> This says "adds" but doesn't say whether to be beginning or the end.
>>>> Also, it isn't strictly necessary to add another header field, it can add to an existing header field. So I suggest:
>>>>
>>>>      When a UA sends a SIP request, if the UA wants to indicate support of
>>>>      features towards its downstream SIP entities, before it forwards the
>>>>     request it adds a fc-value with the supported features to
>>>>      the request, after any others that may be present. This may be added
>>>>      to the last existing Feature-Caps header field or as a new
>>>>      Feature-Caps header field after any others.
>>>
>>>> Something similar applies a couple more times in this section and in sections 5.2.3 and 5.2.4. I realize this is a bit cumbersome - perhaps there is a more straightforward way to say it.
>>>
>>> I will double check, but I THINK we normally only talk about adding a value to an existing header field. The ABNF rules then shows whether multiple instances of the header field is allowed.
>>>
>>> We just need to be sure that the text is clear on which features are "closest" to the UA, no matter whether they are indicated in a single header field, or in multiple header fields.
>>
>> Yes, I think we do usually do that, even though it is imprecise.
>> Within reason I'm ok with you following precedent of something with
>> similar behavior. Record-Route comes to mind.
>
> Yes, and this is where I need to correct myself.
>
> When a proxy adds Record-Route (or Via), it adds its value *before* any existing values, and the idea is the same for feature-caps. That is indicated in section 5.2.4 (see below), but
> I guess should be in a general section.
>
>     "When a proxy adds a Feature-Caps header field to a SIP message, it
>     MUST place the header field before any existing Feature-Caps header
>     fields in the request."
>
>     (s/request/message)
>
>
> ---------------------
>
>
>>>> Section 5.3.3 says:
>>>>
>>>>      If a SIP feature cap is inserted in a Feature-Caps header field of a
>>>>      SIP REGISTER request, or within a response of such request, it
>>>>      indicates to the receivers of the request (or response) that the
>>>>      feature associated with the SIP feature cap is supported for the
>>>>      duration of the registration, and for all SIP transactions associated
>>>>      with the registration, until the registration is re-freshed or
>>>>      terminated.
>>>>
>>>> There are some cases we need to discuss here. When a UA sends a REGISTER request, it could have no contacts, one contact, or several. What do the feature caps mean in these cases?
>>>>
>>>> - if there are no contacts in the register, are feature-caps irrelevant?
>>>> - if there is more than one contact in the register, do the feature-caps
>>>>     apply to all of them?
>>>>
>>>> Then the response should contain all the registered contacts, not just those mentioned in the request. So do the feature caps mentioned in the reply apply until the last of those
>>>> contacts expire? Or should the requester assume they only apply until *its* contacts expire?
>>>>
>>>>      Unless a SIP feature cap is inserted in a Feature-Caps header field
>>>>      or a re-registration, or within a response of such request, it
>>>>      indicates to the receivers of the request (or response) that the
>>>>      feature is no long supported for the registration.
>>>>
>>>> Suppose a UA registers with a contact and some feature caps. Then later it sends a register poll (register with no contacts). Can I assume that this will not affect the feature caps associated with the registered contact?
>>>
>>> It's up to the entities that previously inserted the feature-caps to decide whether they will continue to indicate support when they receive a re-registration request/response - no matter whether it's a poll, addition of contacts etc.
>>>
>>> Personally I see no reason why they should *not* insert the same feature-caps in a pure poll request/response, but we should allow for cases where the feature is no longer supported when the poll request/response is received.
>>
>> I think you missed my point. Its really the same point as my question of
>> what it means to put feature caps into a polling register.
>>
>> Since you want the registrar to consider the received feature caps to be
>> stateful for the duration of the register, I want to know what that
>> state is attached to. ISTM that it must be attached to particular
>> contacts. Then, a polling register would presumably not update the state
>> of any registered contact since it doesn't update any of them.
>>
>> On the flip side of that, if the registrar returns feature caps, what
>> does the recipient attach that state to? The only thing it could attach
>> it to that has a lifetime is the registration of a particular contact,
>> since the registered contacts can expire independently.
>
> Ok, I think I understand. I need to think a little about it.
>
>
> ---------------------
>
>
>>>> Do the enhanced registrar behaviors in outbound require any special treatment relative to feature caps?
>>>
>>> No, I don't think so.
>>>
>>> However, I guess we could add a clarification note saying that features are only supported for the specific outbound registration flow where support has been explicitly indicated.
>>
>> I'm only asking that you look at it. If nothing is needed, that is fine.
>> The clarification sounds good if it is true. (I don't know if it is true.)
>
> I guess this is related to the issue above.
>
>
> ---------------------
>
>
>>>> Section 6:
>>>>
>>>> There are a lot of missing IANA considerations - the establishment of the new trees as defined in 4.3 and 4.4.
>>>
>>> I used RFC 2506 as base, where the trees are defined more or less in the same way as in the draft. But, I guess it's better to do it in the same way as in RFC 3840.
>>
>> Again, we are establishing a new IANA registry here.
>> I think it unwise to couple it to the definition of a different registry
>> in a document we don't control. (Some day 2506 may be updated, and we
>> wouldn't want that to affect *this* registry.)
>
> What I meant was that RFC 2506 does not define its registry in an IANA Considerations sections.
>
>> In any case, IANA needs to be given explicit instructions to establish the new registry, and couple it to the template.
>
> Yes.
>
> Regards,
>
> Christer


From pravindran@sonusnet.com  Wed Apr 25 18:46:03 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Vijay K. Gurbani" <vkg@bell-labs.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
Thread-Topic: [sipcore] Websocket is a new transport or new URI?
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Vijay,

Thanks for forwarding the fruitful information. Now I understand why Paul
is saying that SIPS URI is a mistake in RFC 3261. The point to be noted is=
=20
that there is no IETF document to clarify this.

Hi all,

I agree that it is better to proceed websokcet as a transport rather than=20
new URI to avoid earlier mistake now.

SIPS is not deprecated for TLS over TCP as it is widely deployed.
Having said that, I'm not fully convinced with SIPS usage for websocket
as IMO, it is extending the mistake to the new transport as well. Knowing
that it is a mistake in the base RFC 3261, it is not the right thing to be=
=20
allowed for new transport introduced. Let us discuss this open item in=20
this thread or other thread.

>From the TCP/IP protocol stack perspective, TLS as URI and WebSocket=20
as transport is not the right thing as it is confusing as follows.

     sips:alice@example.com;transport=3Dws

The better approach will be=20

    sip:alice@example.com;transport=3Dwss

Thanks
Partha

>-----Original Message-----
>From: Vijay K. Gurbani [mailto:vkg@bell-labs.com]
>Sent: Wednesday, April 25, 2012 9:40 PM
>To: Paul Kyzivat
>Cc: Sa=FAl Ibarra Corretg=E9; Ravindran, Parthasarathi; sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 04/25/2012 09:55 AM, Paul Kyzivat wrote:
>> I agree with Sa=FAl here. I don't really understand what it is that
>> Partha is proposing.
>>
>> As I already stated, I think sips turned out to be a mistake - that it
>> doesn't provide enough of a guarantee to be able to promise anything
>> to an end user. But we investigated this ad nauseum and couldn't come
>> up with anything substantially better.
>
>Time for institutional memory, perhaps?  The long discussions (May -
>June 2006) on why sips failed and ideas to come up with new URI schemes
>to replace it and attendant problems (upcasting URIs, downcasting URIs
>without user consent, hop-by-hop semantics versus end-to-end, etc.) is
>summarized in http://www.ietf.org/mail-
>archive/web/sip/current/msg24047.html
>
>- vijay
>--
>Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
>Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
>Web:   http://ect.bell-labs.com/who/vkg/

From christer.holmberg@ericsson.com  Thu Apr 26 00:57:54 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Thu, 26 Apr 2012 09:57:46 +0200
Thread-Topic: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Hi Paul,

Some thinking on the contact issue.

>>>> Section 5.3.3 says:
>>>>
>>>>      If a SIP feature cap is inserted in a Feature-Caps header field o=
f a
>>>>      SIP REGISTER request, or within a response of such request, it
>>>>      indicates to the receivers of the request (or response) that the
>>>>      feature associated with the SIP feature cap is supported for the
>>>>      duration of the registration, and for all SIP transactions associ=
ated
>>>>      with the registration, until the registration is re-freshed or
>>>>      terminated.
>>>>
>>>> There are some cases we need to discuss here. When a UA sends a REGIST=
ER request, it could have no contacts, one contact, or several. What do the=
 feature caps mean in these cases?
>>>>
>>>> - if there are no contacts in the register, are feature-caps irrelevan=
t?
>>>> - if there is more than one contact in the register, do the feature-ca=
ps
>>>>     apply to all of them?
>>>>
>>>> Then the response should contain all the registered contacts, not=20
>>>> just those mentioned in the request. So do the feature caps mentioned =
in the reply apply until the last of those contacts expire? Or should the r=
equester assume they only apply until *its* contacts expire?
>>>>
>>>>      Unless a SIP feature cap is inserted in a Feature-Caps header fie=
ld
>>>>      or a re-registration, or within a response of such request, it
>>>>      indicates to the receivers of the request (or response) that the
>>>>      feature is no long supported for the registration.
>>>>
>>>> Suppose a UA registers with a contact and some feature caps. Then late=
r it sends a register poll (register with no contacts). Can I assume that t=
his will not affect the feature caps associated with the registered contact=
?
>>>
>>> It's up to the entities that previously inserted the feature-caps to de=
cide whether they will continue to indicate support when they receive a re-=
registration request/response - no matter whether it's a poll, addition of =
contacts etc.
>>>
>>> Personally I see no reason why they should *not* insert the same featur=
e-caps in a pure poll request/response, but we should allow for cases where=
 the feature is no longer supported when the poll request/response is recei=
ved.
>>
>> I think you missed my point. Its really the same point as my question=20
>> of what it means to put feature caps into a polling register.
>>
>> Since you want the registrar to consider the received feature caps to=20
>> be stateful for the duration of the register, I want to know what=20
>> that state is attached to. ISTM that it must be attached to=20
>> particular contacts. Then, a polling register would presumably not=20
>> update the state of any registered contact since it doesn't update any o=
f them.
>>
>> On the flip side of that, if the registrar returns feature caps, what=20
>> does the recipient attach that state to? The only thing it could=20
>> attach it to that has a lifetime is the registration of a particular=20
>> contact, since the registered contacts can expire independently.
>
> Ok, I think I understand. I need to think a little about it.

My suggestion would be:

- When outbound is *not* used, if support of feature F9 is indicated when c=
ontact C1 is registered, the feature is supported until C1 expires, is expl=
icitly removed, or explicitly refreshed.

- When outbound *is* used, the same as above applies, but only for the spec=
ific C1 registration flow for which support of F9 has been indicated.

	Example: If I register C1, using two outbound flows (O1 and O2), and suppo=
rt of F9 is indicated for O1, when O1 is expired/removed/refreshed F9 is no=
 longer supported.

- Whenever a contact is explicitly refreshed, entities MUST insert Feature-=
Caps if they still want to indicate support of the feature.

- For pure poll messages, where the REGISTER request does *not* contain a c=
ontact, my suggestion is that we say that the document does not define any =
semantics for usage of Feature-Caps in such requests/responses. If no seman=
tics is defined, poll messages will not have any impact on previously indic=
ated support of features, and entities MUST NOT insert a Feature-Caps heade=
r field in such requests, or their associated responses.


How does that sound? :)


Regards,

Christer







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To: Christer Holmberg <christer.holmberg@ericsson.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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Hi Christer,

When I argued for new URI, the contact shall use new URI with=20
the existing TCP transport like

   sipws:alice@example.com;transport=3Dtcp

"sipws" URI shall be compare with "ws" URI in Websocket protocol=20
(Sec 3 of RFC 6455).=20

Thanks
Partha

>-----Original Message-----
>From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>Sent: Thursday, April 26, 2012 1:48 AM
>To: I=F1aki Baz Castillo; Ravindran, Parthasarathi
>Cc: sipcore@ietf.org
>Subject: RE: [sipcore] Websocket is a new transport or new URI?
>
>Hi,
>
>I agree with Inaki.
>
>Whatever "layer" we consider the WebSocket to be, in my opinion the URI
>scheme is used for addressing (either as an AOR or a contact), and
>should be independent of the transport.
>
>For what would the new scheme be needed  anyway?
>
>Regards,
>
>Christer
>
>
>
>
>________________________________________
>From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of
>I=F1aki Baz Castillo [ibc@aliax.net]
>Sent: Wednesday, April 25, 2012 8:25 PM
>To: Ravindran, Parthasarathi
>Cc: sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>Hi Partha,
>
>
>2012/4/25 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
>> So IMO, it is more appropriate to define new URI scheme like SIPWS
>> rather than defining websocket as a transport.
>
>
>The SIP transport defined by this draft is WebSocket, instead of TCP.
>WebSocket (RFC 6455) is defined to work on top of TCP but, even if a
>future spec defines WebSocket to run over SCTP, the SIP transport would
>still be "WebSocket" (or perhaps "WebSocket-SCTP" but that something in
>which I won't waste time now given the fact that RFC 6455
>*mandates* TCP layer-3 transport and does not leave the door open for
>other layer-3 transports).
>
>WebSocket is, indeed, a layer on top of a network transport (layer 3)
>but it's designed to work as a *transport* layer since it defines a
>framing mechanism to send application messages and control mesages in
>both directions. Applications (i.e. JavaScript scripts, PHP/Java web
>apps) use WebSocket as a transport layer.
>
>You can find lot of cases in which a transport runs on top of other
>transports. For example, in the recent IETF #83 the RTCWEB WG approved
>(or discussed) SCTP-over-DTLS-over-ICE-over-UDP as the transport for the
>WebRTC Data-Channel mechanism. A "transport" does not necessarily mean
>"layer 3 network transport".
>
>
>
>About SIPS, the last S in SIPS stands for "secure" regardless of the
>undelying SIP transport used. AFAIK it does not necessarily mean "TLS".
>Take for example draft-jennings-sip-dtls-05 [1] which defines SIP over
>DTLS over UDP. The draf is expired, but anyhow, it defines a secure SIP
>transport which does not use TLS but DTLS (which is similar but NOT the
>same). The draft also allows SIPS messages through DTLS over UDP.
>
>[1] http://tools.ietf.org/html/draft-jennings-sip-dtls-05
>
>
>When a SIPS message is carried on top of a secure WebSocket connection
>(which means SIP over WebSocket over TLS over TCP) the security level is
>basically the same as if it was SIP over TLS over TCP. Therefore I
>consider that a secure WebSocket connection fits the requirements of the
>SIPS mechanism, thus secure WebSocket should be considered as a secure
>SIP transport.
>
>
>
>About the SIPWS and SIPWSS schemas you propose, don't take me wrong but
>I consider that the worst possible idea. First of all because I don't
>agree with your rationale (I insist that, from the point of view of a
>SIP stack, WebSocket is the ***SIP transport***, instead of TCP).
>Secondly, because adding new URI schemas would just break the existing
>SIP world (there are still some SIP entities that don't understand SIPS
>schema and you are proposing two new ones?).
>
>
>Thanks a lot for your comments.
>
>
>--
>I=F1aki Baz Castillo
><ibc@aliax.net>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

From rick@openfortress.nl  Thu Apr 26 02:19:06 2012
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From: Rick van Rein <rick@openfortress.nl>
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Subject: [sipcore] Proposal: sip6 URI scheme
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--Q68bSM7Ycu6FN28Q
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Hello,

I am developing to get better support for SIP over IPv6.  One of the
problems that I have encountered is the interoperability between
endpoints that run only IPv4 or IPv6.  To solve this situations prior
to call setup, I am proposing a sip6 URI scheme.  Using this, end users
and their tools should have an easier time deciding whether or not to
use the URI.

A similar thing applies to enforced ZRTP --being sure before calling
that the call will be private-- and I combined that with the URI scheme
proposal, so as not to register more URI schemes than required.

Your feedback is kindly welcomed.

Best wishes,

Rick van Rein
OpenFortress

--Q68bSM7Ycu6FN28Q
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   URI scheme name.

      sip6
      
   Status.

      permanent

   URI scheme syntax.

      sip6:<user>[:<password>]@<host>[:<port>][;<uri-parameters>][?<headers>]
      
      The scheme is the same as for sip: with the exception of the URI scheme.

   URI scheme semantics.

      SIP entities [RFC3261] that process the sip6 scheme URI ensure that
      the following additional constraints for the sip6 URI scheme are not
      broken, by implementing them inasfar as it concerns their processing:

       * All SIP, media descriptive and media traffic that follows from a
	 sip6 URI scheme MUST be communicated over IPv6, and describe
	 IPv6-capable resources.

       * All RTP traffic that follows from a a sip6 URI MUST support
         ZRTP [RFC6189] and actively attempt to establish ZRTP.
         Descriptions of such RTP communications SHOULD offer ZRTP.

      These additional requirements describe a method of deploying SIP
      that brings it nearer to its ideal configuration; namely, without
      any need for an intermediate party in the media path. There are
      advantages to deploying configurations that only work within these
      constraints, but with only a sip URI scheme such configurations
      would not be discovered until a call setup was being attempted,
      leading to potential end-user experiences of broken calls.  The
      sip6 URI scheme is intended to avoid that problematic behavior.

   Applications/protocols that use this URI scheme name.

      The Session Initiation Protocol can be used with the sip6 URI
      scheme.  With the exception that the aforementioned additional
      requirements apply, the processing is not in any way different.
      Specifically, the protocol continues to be named SIP in transport
      descriptions, and _sip._udp continues to be used a DNS name for
      SRV records.

      The syntactic declaration of support of this scheme can help to
      make an early selection of the proper tools, either by an
      end-user who can be properly informed, or by generic browsing tools
      that start different handling agents for the sip and sip6 URI
      schemes.

      A SIP phone that dials a sip6 URI can be certain that IPv6-only
      communication suffices; the phone can refuse attempts to dial
      such a number if IPv6 is not available, and prefer a sip URI
      alternative if it is available.  On the other hand, current SIP
      phones that only support IPv4 will not recognise the sip6: URI
      scheme, and as a result skip attempts to contact them.

      There is no prohibition on translating proxies that map the sip6
      URI sphere to the sip URI sphere, provided that the sip6 side
      of such proxies conceals the parts of the sip side that are not
      compliant to the additional requirements on the sip6 side.

   Interoperability considerations.

      The sip URI suggests a capability to communicate over IPv4 and
      IPv6, but in practice these protocols are incompatible.  The
      result of having both address schemes under the same URI scheme
      can lead to interoperability problems.  The introduction of a
      separate URI scheme for SIP over IPv6 improves this situation,
      by reflecting the incompability of the IPv4 and IPv6 spaces in
      an easily processed form.

      The required support for ZRTP implies that any implementations that
      use RTP but do not implement ZRTP facilities are inoperable with
      the sip6 URI format.  This too is a deliberate choice that leads
      to more certainty to be derived from the connections being setup.

   Security considerations.

      The usual security considerations for SIP apply, with the
      exception that the requirement of ZRTP support reduces the
      risks of phone tapping and man-in-the-middle attacks.

   Contact.

      Rick van Rein <rick@openfortress.nl>

   Author/Change controller.

      Rick van Rein <rick@openfortress.nl>

   References.
   
   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC6189]  Zimmermann, P., Zfone Project, Johnston, E., Ayaya and
              Calls, J., "ZRTP: Media Path Key Agreement for Unicast
              Secure RTP", April 2011.


--Q68bSM7Ycu6FN28Q--

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26 apr 2012 kl. 11:27 skrev Rick van Rein:

> Hello,
>=20
> I am developing to get better support for SIP over IPv6.  One of the
> problems that I have encountered is the interoperability between
> endpoints that run only IPv4 or IPv6.  To solve this situations prior
> to call setup, I am proposing a sip6 URI scheme.  Using this, end =
users
> and their tools should have an easier time deciding whether or not to
> use the URI.
>=20

A new URI scheme doesn't make any sense, since a SIP uri can be resolved =
to many different hosts using NAPTR/SRV records.

A phone that is dual stack can register with two contacts, one for each =
address family. ICE will take care of media handling.


/O=

From rick@openfortress.nl  Thu Apr 26 02:29:03 2012
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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Hello Olle,

> A new URI scheme doesn't make any sense, since a SIP uri can be resolved to many different hosts using NAPTR/SRV records.

This only helps with the SIP communication -- but it gives no certainty
about RTP.  Interoperability between IPv4-only and IPv6-only relating to
media can only be found when trying to setup a call, right?

> A phone that is dual stack can register with two contacts, one for each address family. ICE will take care of media handling.

There is no formal relation between the IP version used for SIP and used
for RTP.  This is what I am proposing to solve with sip6:


Thanks,
 -Rick

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2012/4/26 Rick van Rein <rick@openfortress.nl>:
> I am developing to get better support for SIP over IPv6. =C2=A0One of the
> problems that I have encountered is the interoperability between
> endpoints that run only IPv4 or IPv6. =C2=A0To solve this situations prio=
r
> to call setup, I am proposing a sip6 URI scheme.

Hi Rick, I'm pretty sure that's not the way to go. As Olle pointed out
a SIP URI with domain could point, after RFC 3263 resolution, to many
transport:address:port, some of them could be IPv4 and others IPv6.

Your proposal would indeed break existing implementations.

If for example alice uses just IPv4 and bob just IPv6, and both use
and register in a proxy/registrar with IP dual stack, alice should be
able to call bob's AoR (sip:bob@example.net) without knowing that bob
is using IPv6 or not. This is already possible with current specs but
it would be not possible with your proposal.

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From oej@edvina.net  Thu Apr 26 02:32:55 2012
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To: Rick van Rein <rick@openfortress.nl>
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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26 apr 2012 kl. 11:37 skrev Rick van Rein:

> Hello Olle,
>=20
>> A new URI scheme doesn't make any sense, since a SIP uri can be =
resolved to many different hosts using NAPTR/SRV records.
>=20
> This only helps with the SIP communication -- but it gives no =
certainty
> about RTP.  Interoperability between IPv4-only and IPv6-only relating =
to
> media can only be found when trying to setup a call, right?
But a URI scheme doesn't help here.
My softphone keeps moving between networks, some dual stack and some =
not.

>=20
>> A phone that is dual stack can register with two contacts, one for =
each address family. ICE will take care of media handling.
>=20
> There is no formal relation between the IP version used for SIP and =
used
> for RTP.  This is what I am proposing to solve with sip6:

The URI scheme is still WRONG. YOu might want to add a UA capability to =
the Contact registred.
That's something we should look into. Especially if you have a situation =
where a server needs to decide whether to force an RTP proxy or not.
According to the RFC the client should organize this itself with a TURN =
server, but in reality we might need to know on the server.


There's a separate SIPv6 mailing list operated by the SIP forum and a =
SIPv6 forum on Facebook if you want to join the discussion there too :-)

/O=

From ibc@aliax.net  Thu Apr 26 02:33:05 2012
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Cc: uri-review@ietf.org, sipcore@ietf.org, "Olle E. Johansson" <oej@edvina.net>
Subject: Re: [sipcore] Proposal: sip6 URI scheme
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2012/4/26 Rick van Rein <rick@openfortress.nl>:
> There is no formal relation between the IP version used for SIP and used
> for RTP. =C2=A0This is what I am proposing to solve with sip6:

A SIP UA could just use IPv4 for signaling and IPv4 and IPv6 for
media. SIP gateways usually use different address for signaling and
media.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From saul@ag-projects.com  Thu Apr 26 02:36:32 2012
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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Hi Rick,

[snip]
>=20
> There is no formal relation between the IP version used for SIP and =
used
> for RTP.  This is what I am proposing to solve with sip6:
>=20

Why would you need that? If running in a dual stacked environment maybe =
some firewall policy won't allow your RTP packets to go out through IPv6 =
but they could go out through IPv4, ICE can pick the best path for media =
regardless of signaling.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From christer.holmberg@ericsson.com  Thu Apr 26 03:17:08 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Rick van Rein <rick@openfortress.nl>, "uri-review@ietf.org" <uri-review@ietf.org>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 26 Apr 2012 12:17:03 +0200
Thread-Topic: [sipcore] Proposal: sip6 URI scheme
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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Hi,

Maybe you could describe the use-case you have in mind, and what you want t=
o achieve, and then we can look at possible solutions.

Regards,

Christer


-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Rick van Rein
Sent: 26. huhtikuuta 2012 12:27
To: uri-review@ietf.org; sipcore@ietf.org
Subject: [sipcore] Proposal: sip6 URI scheme

Hello,

I am developing to get better support for SIP over IPv6.  One of the proble=
ms that I have encountered is the interoperability between endpoints that r=
un only IPv4 or IPv6.  To solve this situations prior to call setup, I am p=
roposing a sip6 URI scheme.  Using this, end users and their tools should h=
ave an easier time deciding whether or not to use the URI.

A similar thing applies to enforced ZRTP --being sure before calling that t=
he call will be private-- and I combined that with the URI scheme proposal,=
 so as not to register more URI schemes than required.

Your feedback is kindly welcomed.

Best wishes,

Rick van Rein
OpenFortress

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To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>, "Gurbani, Vijay K (Vijay)" <vijay.gurbani@alcatel-lucent.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [sipcore] Websocket is a new transport or new URI?
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I thought RFC 5630 went a long way to explaining exactly what you got when =
you specified SIPS.

It is up to the reader to work out how useful it is as a result - not exact=
ly a great leap.

Keith

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ravindran, Parthasarathi
Sent: 26 April 2012 02:46
To: Gurbani, Vijay K (Vijay); Paul Kyzivat; I=F1aki Baz Castillo
Cc: sipcore@ietf.org; Sa=FAl Ibarra Corretg=E9
Subject: Re: [sipcore] Websocket is a new transport or new URI?

Vijay,

Thanks for forwarding the fruitful information. Now I understand why Paul
is saying that SIPS URI is a mistake in RFC 3261. The point to be noted is=
=20
that there is no IETF document to clarify this.

Hi all,

I agree that it is better to proceed websokcet as a transport rather than=20
new URI to avoid earlier mistake now.

SIPS is not deprecated for TLS over TCP as it is widely deployed.
Having said that, I'm not fully convinced with SIPS usage for websocket
as IMO, it is extending the mistake to the new transport as well. Knowing
that it is a mistake in the base RFC 3261, it is not the right thing to be=
=20
allowed for new transport introduced. Let us discuss this open item in=20
this thread or other thread.

>From the TCP/IP protocol stack perspective, TLS as URI and WebSocket=20
as transport is not the right thing as it is confusing as follows.

     sips:alice@example.com;transport=3Dws

The better approach will be=20

    sip:alice@example.com;transport=3Dwss

Thanks
Partha

>-----Original Message-----
>From: Vijay K. Gurbani [mailto:vkg@bell-labs.com]
>Sent: Wednesday, April 25, 2012 9:40 PM
>To: Paul Kyzivat
>Cc: Sa=FAl Ibarra Corretg=E9; Ravindran, Parthasarathi; sipcore@ietf.org
>Subject: Re: [sipcore] Websocket is a new transport or new URI?
>
>On 04/25/2012 09:55 AM, Paul Kyzivat wrote:
>> I agree with Sa=FAl here. I don't really understand what it is that
>> Partha is proposing.
>>
>> As I already stated, I think sips turned out to be a mistake - that it
>> doesn't provide enough of a guarantee to be able to promise anything
>> to an end user. But we investigated this ad nauseum and couldn't come
>> up with anything substantially better.
>
>Time for institutional memory, perhaps?  The long discussions (May -
>June 2006) on why sips failed and ideas to come up with new URI schemes
>to replace it and attendant problems (upcasting URIs, downcasting URIs
>without user consent, hop-by-hop semantics versus end-to-end, etc.) is
>summarized in http://www.ietf.org/mail-
>archive/web/sip/current/msg24047.html
>
>- vijay
>--
>Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
>1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60563 (USA)
>Email: vkg@{bell-labs.com,acm.org} / vijay.gurbani@alcatel-lucent.com
>Web:   http://ect.bell-labs.com/who/vkg/
_______________________________________________
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: Rick van Rein <rick@openfortress.nl>
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Cc: uri-review@ietf.org, sipcore@ietf.org, "Olle E. Johansson" <oej@edvina.net>
Subject: Re: [sipcore] Proposal: sip6 URI scheme
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On Thu, Apr 26, 2012 at 3:07 PM, Rick van Rein <rick@openfortress.nl> wrote:

> Hello Olle,
>
> > A new URI scheme doesn't make any sense, since a SIP uri can be resolved
> to many different hosts using NAPTR/SRV records.
>
> This only helps with the SIP communication -- but it gives no certainty
> about RTP.  Interoperability between IPv4-only and IPv6-only relating to
> media can only be found when trying to setup a call, right?
>
> > A phone that is dual stack can register with two contacts, one for each
> address family. ICE will take care of media handling.
>
> There is no formal relation between the IP version used for SIP and used
> for RTP.  This is what I am proposing to solve with sip6:
>

[ABN] There is no need for any formal relation between signalling and
media. if you can share your use case you can get the useful replies.

Theoretically, it is acceptable to use different addressing schemes for
signalling and media. AFAIK, There is no constraint to use same IP version
for both media and signalling.

It is very well possible to publish the IPV6 media address (in SDP) whereas
IPv4 is being used for SIP signalling purposes.


>
>
> Thanks,
>  -Rick
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>



-- 
Thanks,
Nataraju A.B.

--e89a8f23509748907904be92bd9f
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu, Apr 2=
6, 2012 at 3:07 PM, Rick van Rein <span dir=3D"ltr">&lt;<a href=3D"mailto:r=
ick@openfortress.nl" target=3D"_blank">rick@openfortress.nl</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">Hello Olle,<br>
<div class=3D"im"><br>
&gt; A new URI scheme doesn&#39;t make any sense, since a SIP uri can be re=
solved to many different hosts using NAPTR/SRV records.<br>
<br>
</div>This only helps with the SIP communication -- but it gives no certain=
ty<br>
about RTP. =A0Interoperability between IPv4-only and IPv6-only relating to<=
br>
media can only be found when trying to setup a call, right?<br>
<div class=3D"im"><br>
&gt; A phone that is dual stack can register with two contacts, one for eac=
h address family. ICE will take care of media handling.<br>
<br>
</div>There is no formal relation between the IP version used for SIP and u=
sed<br>
for RTP. =A0This is what I am proposing to solve with sip6:<br></blockquote=
><div>=A0</div><div>[ABN] There is no need for any formal relation between =
signalling and media. if you can share your use case you can get the useful=
 replies.</div>
<div><br></div><div>Theoretically, it is acceptable to use different addres=
sing schemes for signalling and media. AFAIK, There is no constraint to use=
 same IP version for both media and signalling.</div><div><br></div><div>
It is very well possible to publish the IPV6 media address (in SDP) whereas=
 IPv4 is being used for SIP signalling purposes.</div><div>=A0</div><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

<br>
<br>
Thanks,<br>
=A0-Rick<br>
<div class=3D"HOEnZb"><div class=3D"h5">___________________________________=
____________<br>
sipcore mailing list<br>
<a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/sipcore" target=3D"_blank"=
>https://www.ietf.org/mailman/listinfo/sipcore</a><br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>
</div>

--e89a8f23509748907904be92bd9f--

From rick@openfortress.nl  Thu Apr 26 04:08:22 2012
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From: Rick van Rein <rick@openfortress.nl>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "uri-review@ietf.org" <uri-review@ietf.org>, Rick van Rein <rick@openfortress.nl>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Proposal: sip6 URI scheme
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Hello all,

Thanks for your feedback.

> Maybe you could describe the use-case you have in mind, and what you want
> to achieve, and then we can look at possible solutions.

I am thinking of direct media connections between SIP callers, after setup
of a call from ENUM entries, or perhaps freenum.org, or to one's hosted SIP
proxy under a domain name, e.g. sip:rick@openfortress.nl -- these schemes
are part of the promise that SIP holds for telephony over the Internet.

Direct calls are ideally served with IPv6.  But as long as there is no way
to infer if a service is IPv6-only, there will be a practical need to support
IPv4 as well, so as to avoid failed call setups due to inavailability of a
caller-assumed transport.  With IPv4 as an always-present fallback, there
is no real reason to implement IPv6 as well.  Specifically, for IPv6 as
the connecting protocol between the caller's side and the callee's side.
Everyone is waiting for everyone else.

IPv6 however, is very useful for direct calls over the Internet, as it
has no problems with NAT (only with firewalls, which is easily solved)
and so it never requires an RTP proxy.  RTP proxies are potentially
problematic for a number of reasons:
 - efficiency: RTP traffic may pass through more hops than needed
 - quality: Call quality may suffer from extra media indirections
 - complexity: setting up a private RTP proxy is not straightforward
 - freedom: most end-users will be dependent on an external party's RTP proxy
 - facilities: external RTP proxies may not support all media wishes
   (think of high-bandwidth video or the deaf's realtime-text protocol)
 - privacy: external RTP proxies are often regulated into tapping phone calls

Having a separate indication of SIP over IPv6 clearly states the opposite
situation; there is no need to attempt IPv4 on setups, and so it makes
no sense to use such URIs on any implementation but an IPv6-capable one.
This may include tools that setup IPv6 over a tunnel to create the call.

Please note that nothing prohibits proxies to handle both sip6: and sip:
URIs, and perhaps translate them to each other's format.  The only thing
that would be required, is that such a proxy would use IPv6 only on the
sip6: side, so it may end up being an RTP proxy as well -- but only for
the local end, where a peer has not implemented SIP over IPv6 yet.  The
lookup of a sip6: party is an IPv6-only process, which means that the
default has shifted from SIP over IPv4 to SIP over IPv6, whenever possible.

A new URI scheme is not the only way to accomplish this; in fact, anything
in the syntax of a sip: URI would do it.  The reason why I proposed a URI
scheme is that it works well in browsers etc. that generally choose a
handler application based on the URI scheme; it would not start a sip:
handler if it failed to recognise a ?transport=udp-ipv6 annotation, and it
could usually be configured to start a different handler for sip6: without
knowing anything about the SIP protocol (a protocol handler could register
for handling the sip6: URI and another could register for the sip: URI,
in case of a mobile platform).  These are pragmatic considerations only,
but they made me wish for a URI scheme.

FWIW, I've engineered beta-level solutions that make it possible in
all common usecases to use SIP over IPv6, and IPv6 alone.  Information on
that can be found on http://devel.0cpm.org/ -- mostly work in progress.


Thanks to all for your feedback,


Best wishes,

Rick van Rein
OpenFortress

From pkyzivat@alum.mit.edu  Thu Apr 26 07:37:57 2012
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Christer,

I *think* this sounds right.

But I'm not certain in the case of outbound. It seems like it might 
depend on the particulars of the feature. I say this because features 
are still so fuzzy. In some cases it might be that the behavior of the 
feature is coupled to the flow. In others it might not. Perhaps if I 
understood proxy-features better I wouldn't have that doubt.

	Thanks,
	Paul

On 4/26/12 3:57 AM, Christer Holmberg wrote:
> Hi Paul,
>
> Some thinking on the contact issue.
>
>>>>> Section 5.3.3 says:
>>>>>
>>>>>       If a SIP feature cap is inserted in a Feature-Caps header field of a
>>>>>       SIP REGISTER request, or within a response of such request, it
>>>>>       indicates to the receivers of the request (or response) that the
>>>>>       feature associated with the SIP feature cap is supported for the
>>>>>       duration of the registration, and for all SIP transactions associated
>>>>>       with the registration, until the registration is re-freshed or
>>>>>       terminated.
>>>>>
>>>>> There are some cases we need to discuss here. When a UA sends a REGISTER request, it could have no contacts, one contact, or several. What do the feature caps mean in these cases?
>>>>>
>>>>> - if there are no contacts in the register, are feature-caps irrelevant?
>>>>> - if there is more than one contact in the register, do the feature-caps
>>>>>      apply to all of them?
>>>>>
>>>>> Then the response should contain all the registered contacts, not
>>>>> just those mentioned in the request. So do the feature caps mentioned in the reply apply until the last of those contacts expire? Or should the requester assume they only apply until *its* contacts expire?
>>>>>
>>>>>       Unless a SIP feature cap is inserted in a Feature-Caps header field
>>>>>       or a re-registration, or within a response of such request, it
>>>>>       indicates to the receivers of the request (or response) that the
>>>>>       feature is no long supported for the registration.
>>>>>
>>>>> Suppose a UA registers with a contact and some feature caps. Then later it sends a register poll (register with no contacts). Can I assume that this will not affect the feature caps associated with the registered contact?
>>>>
>>>> It's up to the entities that previously inserted the feature-caps to decide whether they will continue to indicate support when they receive a re-registration request/response - no matter whether it's a poll, addition of contacts etc.
>>>>
>>>> Personally I see no reason why they should *not* insert the same feature-caps in a pure poll request/response, but we should allow for cases where the feature is no longer supported when the poll request/response is received.
>>>
>>> I think you missed my point. Its really the same point as my question
>>> of what it means to put feature caps into a polling register.
>>>
>>> Since you want the registrar to consider the received feature caps to
>>> be stateful for the duration of the register, I want to know what
>>> that state is attached to. ISTM that it must be attached to
>>> particular contacts. Then, a polling register would presumably not
>>> update the state of any registered contact since it doesn't update any of them.
>>>
>>> On the flip side of that, if the registrar returns feature caps, what
>>> does the recipient attach that state to? The only thing it could
>>> attach it to that has a lifetime is the registration of a particular
>>> contact, since the registered contacts can expire independently.
>>
>> Ok, I think I understand. I need to think a little about it.
>
> My suggestion would be:
>
> - When outbound is *not* used, if support of feature F9 is indicated when contact C1 is registered, the feature is supported until C1 expires, is explicitly removed, or explicitly refreshed.
>
> - When outbound *is* used, the same as above applies, but only for the specific C1 registration flow for which support of F9 has been indicated.
>
> 	Example: If I register C1, using two outbound flows (O1 and O2), and support of F9 is indicated for O1, when O1 is expired/removed/refreshed F9 is no longer supported.
>
> - Whenever a contact is explicitly refreshed, entities MUST insert Feature-Caps if they still want to indicate support of the feature.
>
> - For pure poll messages, where the REGISTER request does *not* contain a contact, my suggestion is that we say that the document does not define any semantics for usage of Feature-Caps in such requests/responses. If no semantics is defined, poll messages will not have any impact on previously indicated support of features, and entities MUST NOT insert a Feature-Caps header field in such requests, or their associated responses.
>
>
> How does that sound? :)
>
>
> Regards,
>
> Christer
>
>
>
>
>
>
>


From christer.holmberg@ericsson.com  Thu Apr 26 09:31:31 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Thu, 26 Apr 2012 18:31:26 +0200
Thread-Topic: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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Hi,

> I *think* this sounds right.
>
> But I'm not certain in the case of outbound. It seems like it might
> depend on the particulars of the feature. I say this because features
> are still so fuzzy. In some cases it might be that the behavior of the
> feature is coupled to the flow. In others it might not. Perhaps if I
> understood proxy-features better I wouldn't have that doubt.

I don't think that a feature, explicitly indicated for flow O1, can be impl=
icitly applied also for O2. The support has to be explicitly indicated on e=
ach flow for which it applies.

Because, O2 may not even traverse the entity which indicates the support of=
 the feature for O1, which means that when O2 is refreshed there will be no=
 support indication - which would mean that the feature is no longer suppor=
ted (at least not until O1 is refreshed, and the indication might be added =
again...).

(Probably that could be fixed, e.g. by saying that only the flow where supp=
ort has been indicated can later refresh/remove that support. But, I think =
that would be too messy, e.g. if that flow expires etc.)

So, my suggestion would be that the support is indicated per flow. That's h=
ow it works in the 3GPP, where support is sometimes indicated based on the =
type of access network associated with a flow.

Regards,

Christer






On 4/26/12 3:57 AM, Christer Holmberg wrote:
> Hi Paul,
>
> Some thinking on the contact issue.
>
>>>>> Section 5.3.3 says:
>>>>>
>>>>>       If a SIP feature cap is inserted in a Feature-Caps header field=
 of a
>>>>>       SIP REGISTER request, or within a response of such request, it
>>>>>       indicates to the receivers of the request (or response) that th=
e
>>>>>       feature associated with the SIP feature cap is supported for th=
e
>>>>>       duration of the registration, and for all SIP transactions asso=
ciated
>>>>>       with the registration, until the registration is re-freshed or
>>>>>       terminated.
>>>>>
>>>>> There are some cases we need to discuss here. When a UA sends a REGIS=
TER request, it could have no contacts, one contact, or several. What do th=
e feature caps mean in these cases?
>>>>>
>>>>> - if there are no contacts in the register, are feature-caps irreleva=
nt?
>>>>> - if there is more than one contact in the register, do the feature-c=
aps
>>>>>      apply to all of them?
>>>>>
>>>>> Then the response should contain all the registered contacts, not
>>>>> just those mentioned in the request. So do the feature caps mentioned=
 in the reply apply until the last of those contacts expire? Or should the =
requester assume they only apply until *its* contacts expire?
>>>>>
>>>>>       Unless a SIP feature cap is inserted in a Feature-Caps header f=
ield
>>>>>       or a re-registration, or within a response of such request, it
>>>>>       indicates to the receivers of the request (or response) that th=
e
>>>>>       feature is no long supported for the registration.
>>>>>
>>>>> Suppose a UA registers with a contact and some feature caps. Then lat=
er it sends a register poll (register with no contacts). Can I assume that =
this will not affect the feature caps associated with the registered contac=
t?
>>>>
>>>> It's up to the entities that previously inserted the feature-caps to d=
ecide whether they will continue to indicate support when they receive a re=
-registration request/response - no matter whether it's a poll, addition of=
 contacts etc.
>>>>
>>>> Personally I see no reason why they should *not* insert the same featu=
re-caps in a pure poll request/response, but we should allow for cases wher=
e the feature is no longer supported when the poll request/response is rece=
ived.
>>>
>>> I think you missed my point. Its really the same point as my question
>>> of what it means to put feature caps into a polling register.
>>>
>>> Since you want the registrar to consider the received feature caps to
>>> be stateful for the duration of the register, I want to know what
>>> that state is attached to. ISTM that it must be attached to
>>> particular contacts. Then, a polling register would presumably not
>>> update the state of any registered contact since it doesn't update any =
of them.
>>>
>>> On the flip side of that, if the registrar returns feature caps, what
>>> does the recipient attach that state to? The only thing it could
>>> attach it to that has a lifetime is the registration of a particular
>>> contact, since the registered contacts can expire independently.
>>
>> Ok, I think I understand. I need to think a little about it.
>
> My suggestion would be:
>
> - When outbound is *not* used, if support of feature F9 is indicated when=
 contact C1 is registered, the feature is supported until C1 expires, is ex=
plicitly removed, or explicitly refreshed.
>
> - When outbound *is* used, the same as above applies, but only for the sp=
ecific C1 registration flow for which support of F9 has been indicated.
>
>       Example: If I register C1, using two outbound flows (O1 and O2), an=
d support of F9 is indicated for O1, when O1 is expired/removed/refreshed F=
9 is no longer supported.
>
> - Whenever a contact is explicitly refreshed, entities MUST insert Featur=
e-Caps if they still want to indicate support of the feature.
>
> - For pure poll messages, where the REGISTER request does *not* contain a=
 contact, my suggestion is that we say that the document does not define an=
y semantics for usage of Feature-Caps in such requests/responses. If no sem=
antics is defined, poll messages will not have any impact on previously ind=
icated support of features, and entities MUST NOT insert a Feature-Caps hea=
der field in such requests, or their associated responses.
>
>
> How does that sound? :)
>
>
> Regards,
>
> Christer
>
>
>
>
>
>
>=

From pkyzivat@alum.mit.edu  Thu Apr 26 09:46:44 2012
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-proxy-feature-01
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OK.

	Thanks,
	Paul

On 4/26/12 12:31 PM, Christer Holmberg wrote:
> Hi,
>
>> I *think* this sounds right.
>>
>> But I'm not certain in the case of outbound. It seems like it might
>> depend on the particulars of the feature. I say this because features
>> are still so fuzzy. In some cases it might be that the behavior of the
>> feature is coupled to the flow. In others it might not. Perhaps if I
>> understood proxy-features better I wouldn't have that doubt.
>
> I don't think that a feature, explicitly indicated for flow O1, can be implicitly applied also for O2. The support has to be explicitly indicated on each flow for which it applies.
>
> Because, O2 may not even traverse the entity which indicates the support of the feature for O1, which means that when O2 is refreshed there will be no support indication - which would mean that the feature is no longer supported (at least not until O1 is refreshed, and the indication might be added again...).
>
> (Probably that could be fixed, e.g. by saying that only the flow where support has been indicated can later refresh/remove that support. But, I think that would be too messy, e.g. if that flow expires etc.)
>
> So, my suggestion would be that the support is indicated per flow. That's how it works in the 3GPP, where support is sometimes indicated based on the type of access network associated with a flow.
>
> Regards,
>
> Christer
>
>
>
>
>
>
> On 4/26/12 3:57 AM, Christer Holmberg wrote:
>> Hi Paul,
>>
>> Some thinking on the contact issue.
>>
>>>>>> Section 5.3.3 says:
>>>>>>
>>>>>>        If a SIP feature cap is inserted in a Feature-Caps header field of a
>>>>>>        SIP REGISTER request, or within a response of such request, it
>>>>>>        indicates to the receivers of the request (or response) that the
>>>>>>        feature associated with the SIP feature cap is supported for the
>>>>>>        duration of the registration, and for all SIP transactions associated
>>>>>>        with the registration, until the registration is re-freshed or
>>>>>>        terminated.
>>>>>>
>>>>>> There are some cases we need to discuss here. When a UA sends a REGISTER request, it could have no contacts, one contact, or several. What do the feature caps mean in these cases?
>>>>>>
>>>>>> - if there are no contacts in the register, are feature-caps irrelevant?
>>>>>> - if there is more than one contact in the register, do the feature-caps
>>>>>>       apply to all of them?
>>>>>>
>>>>>> Then the response should contain all the registered contacts, not
>>>>>> just those mentioned in the request. So do the feature caps mentioned in the reply apply until the last of those contacts expire? Or should the requester assume they only apply until *its* contacts expire?
>>>>>>
>>>>>>        Unless a SIP feature cap is inserted in a Feature-Caps header field
>>>>>>        or a re-registration, or within a response of such request, it
>>>>>>        indicates to the receivers of the request (or response) that the
>>>>>>        feature is no long supported for the registration.
>>>>>>
>>>>>> Suppose a UA registers with a contact and some feature caps. Then later it sends a register poll (register with no contacts). Can I assume that this will not affect the feature caps associated with the registered contact?
>>>>>
>>>>> It's up to the entities that previously inserted the feature-caps to decide whether they will continue to indicate support when they receive a re-registration request/response - no matter whether it's a poll, addition of contacts etc.
>>>>>
>>>>> Personally I see no reason why they should *not* insert the same feature-caps in a pure poll request/response, but we should allow for cases where the feature is no longer supported when the poll request/response is received.
>>>>
>>>> I think you missed my point. Its really the same point as my question
>>>> of what it means to put feature caps into a polling register.
>>>>
>>>> Since you want the registrar to consider the received feature caps to
>>>> be stateful for the duration of the register, I want to know what
>>>> that state is attached to. ISTM that it must be attached to
>>>> particular contacts. Then, a polling register would presumably not
>>>> update the state of any registered contact since it doesn't update any of them.
>>>>
>>>> On the flip side of that, if the registrar returns feature caps, what
>>>> does the recipient attach that state to? The only thing it could
>>>> attach it to that has a lifetime is the registration of a particular
>>>> contact, since the registered contacts can expire independently.
>>>
>>> Ok, I think I understand. I need to think a little about it.
>>
>> My suggestion would be:
>>
>> - When outbound is *not* used, if support of feature F9 is indicated when contact C1 is registered, the feature is supported until C1 expires, is explicitly removed, or explicitly refreshed.
>>
>> - When outbound *is* used, the same as above applies, but only for the specific C1 registration flow for which support of F9 has been indicated.
>>
>>        Example: If I register C1, using two outbound flows (O1 and O2), and support of F9 is indicated for O1, when O1 is expired/removed/refreshed F9 is no longer supported.
>>
>> - Whenever a contact is explicitly refreshed, entities MUST insert Feature-Caps if they still want to indicate support of the feature.
>>
>> - For pure poll messages, where the REGISTER request does *not* contain a contact, my suggestion is that we say that the document does not define any semantics for usage of Feature-Caps in such requests/responses. If no semantics is defined, poll messages will not have any impact on previously indicated support of features, and entities MUST NOT insert a Feature-Caps header field in such requests, or their associated responses.
>>
>>
>> How does that sound? :)
>>
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>>


From dwing@cisco.com  Thu Apr 26 17:47:26 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Rick van Rein'" <rick@openfortress.nl>, <uri-review@ietf.org>, <sipcore@ietf.org>
References: <20120426092725.GC27002@newphantom.local>
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Rick van Rein
> Sent: Thursday, April 26, 2012 2:27 AM
> To: uri-review@ietf.org; sipcore@ietf.org
> Subject: [sipcore] Proposal: sip6 URI scheme
> 
> Hello,
> 
> I am developing to get better support for SIP over IPv6.  One of the
> problems that I have encountered is the interoperability between
> endpoints that run only IPv4 or IPv6.  To solve this situations prior
> to call setup, I am proposing a sip6 URI scheme.  Using this, end users
> and their tools should have an easier time deciding whether or not to
> use the URI.
> 
> A similar thing applies to enforced ZRTP --being sure before calling
> that the call will be private-- and I combined that with the URI scheme
> proposal, so as not to register more URI schemes than required.
> 
> Your feedback is kindly welcomed.

It's a layering violation, and -- worse -- will cause far more harm
than it could possibly help.  Please don't do this.

http://tools.ietf.org/html/rfc6157 describes how SIP is expected to
support IPv6, and transition from IPv4-only to dual-stack to IPv6-only
with the least amount of harm.  It works over NAT64 and NAT46, too
(SIP6, if deployed, can't), and NAT64 will be necessary for IPv6-only
endpoints to access IPv4-only endpoints, such as being deployed by
T-Mobile USA's IPv6-only service, Swisscom's IPv6-only trial, and
probably other mobile operators and, some day, eventually wireline
operators.

If RFC6157 fails, which is why you are proposing SIP6, you need
to back up and explain to everybody the failure of RFC6157.

-d



From rick@openfortress.nl  Thu Apr 26 22:21:13 2012
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Cc: uri-review@ietf.org, 'Rick van Rein' <rick@openfortress.nl>, sipcore@ietf.org
Subject: Re: [sipcore] Proposal: sip6 URI scheme
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Hello,

> > Your feedback is kindly welcomed.
> 
> It's a layering violation, and -- worse -- will cause far more harm
> than it could possibly help.  Please don't do this.

Thanks all for looking into this proposal.  It is clear that it will not
gain wide acceptance, and should not be taken any further.

I'm withdrawing the proposal.

Best wishes,
 -Rick

From nataraju.sip@gmail.com  Fri Apr 27 01:39:23 2012
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Subject: [sipcore] RFC 6455 - conflicting statements
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--90e6ba30952a72651f04bea50a10
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Hi All,

Following is the snippet from the RFC6455.

<RFC6455>

2.  Conformance Requirements

   All diagrams, examples, and notes in this specification are non-
   normative, as are all sections explicitly marked non-normative.
   *Everything *else in this specification is normative.


1.9.  Subprotocols Using the WebSocket Protocol

   _This section is non-normative._

   The client can request that the server use a specific subprotocol by
   including the |Sec-WebSocket-Protocol| field in its handshake.  If it
   is specified, the server needs to include the same field and one of
   the selected subprotocol values in its response for the connection to
   be established.


3.  WebSocket URIs

   This specification defines two URI schemes, using the ABNF syntax
   defined in RFC 5234 [RFC5234], and terminology and ABNF productions
   defined by the URI specification RFC 3986 [RFC3986].

          ws-URI = "ws:" "//" host [ ":" port ] path [ "?" query ]
          wss-URI = "wss:" "//" host [ ":" port ] path [ "?" query ]

          host = <host, defined in [RFC3986], Section 3.2.2>
          port = <port, defined in [RFC3986], Section 3.2.3>
          path = <path-abempty, defined in [RFC3986], Section 3.3>
          query = <query, defined in [RFC3986], Section 3.4>

</RFC6455>

*Comment*: According to statement in sec 2, section 3 - WebSocket URIs (for
example) is normative. But I don't think that it correct. Section 3,
similarly sec 4, 5, 6, 7, 8 must be changed as non-normative text.

Am I missing something here in basic understanding ???

Thanks,
Nataraju A.B.

--90e6ba30952a72651f04bea50a10
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">Hi All,</=
font><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"=
><br></font></div><div><font face=3D"&#39;courier new&#39;, monospace" colo=
r=3D"#000099">Following is the snippet from the RFC6455.</font></div>


<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"><br>=
</font></div><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"=
#000099">&lt;</font><span style=3D"color:rgb(0,0,153);font-family:&#39;cour=
ier new&#39;,monospace">RFC6455&gt;</span></div>
<div><pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D=
"&#39;courier new&#39;, monospace" color=3D"#000099">2.  Conformance Requir=
ements

   All diagrams, examples, and notes in this specification are non-
   normative, as are all sections explicitly marked non-normative.
   <b><u>Everything </u></b>else in this specification is normative.</font>=
</pre><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099=
"><br></font></div><div><pre style=3D"word-wrap:break-word;white-space:pre-=
wrap">
<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">1.9.  Sub=
protocols Using the WebSocket Protocol

   _This section is non-normative._

   The client can request that the server use a specific subprotocol by
   including the |Sec-WebSocket-Protocol| field in its handshake.  If it
   is specified, the server needs to include the same field and one of
   the selected subprotocol values in its response for the connection to
   be established.</font></pre><pre style=3D"word-wrap:break-word;white-spa=
ce:pre-wrap"><font face=3D"&#39;courier new&#39;, monospace" color=3D"#0000=
99"><br></font></pre></div><div><pre style=3D"word-wrap:break-word;white-sp=
ace:pre-wrap">
<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">3.  WebSo=
cket URIs

   This specification defines two URI schemes, using the ABNF syntax
   defined in RFC 5234 [RFC5234], and terminology and ABNF productions
   defined by the URI specification RFC 3986 [RFC3986].

          ws-URI =3D &quot;ws:&quot; &quot;//&quot; host [ &quot;:&quot; po=
rt ] path [ &quot;?&quot; query ]
          wss-URI =3D &quot;wss:&quot; &quot;//&quot; host [ &quot;:&quot; =
port ] path [ &quot;?&quot; query ]

          host =3D &lt;host, defined in [RFC3986], Section 3.2.2&gt;
          port =3D &lt;port, defined in [RFC3986], Section 3.2.3&gt;
          path =3D &lt;path-abempty, defined in [RFC3986], Section 3.3&gt;
          query =3D &lt;query, defined in [RFC3986], Section 3.4&gt;
</font></pre><font face=3D"&#39;courier new&#39;, monospace" color=3D"#0000=
99">&lt;/</font><span style=3D"color:rgb(0,0,153);font-family:&#39;courier =
new&#39;,monospace">RFC6455&gt;</span>
</div><div><span style=3D"color:rgb(0,0,153);font-family:&#39;courier new&#=
39;,monospace"><br></span></div><div><font face=3D"&#39;courier new&#39;, m=
onospace" color=3D"#000099"><b>Comment</b>: According to statement in sec 2=
, section 3 -=A0</font><span style=3D"color:rgb(0,0,153);font-family:&#39;c=
ourier new&#39;,monospace;white-space:pre-wrap">WebSocket URIs (for example=
)</span><span style=3D"color:rgb(0,0,153);font-family:&#39;courier new&#39;=
,monospace">=A0is normative. But I don&#39;t think that it correct.=A0</spa=
n><span style=3D"color:rgb(0,0,153);font-family:&#39;courier new&#39;,monos=
pace">Section 3, similarly sec 4, 5, 6, 7, 8 must be changed as non-normati=
ve text.</span></div>
<div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br>
</font></div><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, mo=
nospace">Am I missing something here in basic understanding ???</font></div=
><div><font color=3D"#000099" face=3D"&#39;courier new&#39;, monospace"><br=
></font></div>
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div>
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Nataraju A.B.</font></font></div>
</div>

--90e6ba30952a72651f04bea50a10--

From ibc@aliax.net  Fri Apr 27 01:50:22 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 27 Apr 2012 10:49:59 +0200
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Subject: Re: [sipcore] RFC 6455 - conflicting statements
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2012/4/27 Nataraju A.B <nataraju.sip@gmail.com>:
> Comment: According to statement in sec 2, section 3 -=C2=A0WebSocket URIs=
 (for
> example)=C2=A0is normative. But I don't think that it correct.=C2=A0Secti=
on 3,
> similarly sec 4, 5, 6, 7, 8 must be changed as non-normative text.
>
> Am I missing something here in basic understanding ???

Nataraju, RFC 6455 belongs to HYBI Working Group, and not to SIPCORE WG.

Anyhow, it's too late to fix/improve an already published RFC, so I
suggest you to report an issue in the IETF tracker for RFC 6455.

Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From nataraju.sip@gmail.com  Fri Apr 27 01:55:40 2012
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Subject: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - editorial comments
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--f46d040715299ffa1304bea544ca
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Hi All,

Following is the snippet of text from rfc6455 and
"draft-ibc-sipcore-sip-websocket-02.txt".

<draft-ibc-sipcore-sip-websocket-02.txt>

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

<draft-ibc-sipcore-sip-websocket-02.txt>

<rfc6455>

2.  Conformance Requirements

   All diagrams, examples, and notes in this specification are non-
   normative, as are all sections explicitly marked non-normative.
   Everything else in this specification is normative.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

</rfc6455>

Usage of text "_This section is non-normative._" will not convey same
meaning in both the documents.

Please add the first paragraph of sec.2,rfc6455 to sec.2 of this draft.
Otherwise the draft would be confusing about usage of the text "_This
section is non-normative._"

I know this is minor comment, but required to make the draft clearer.

Thanks,
*Nataraju A.B.*

--f46d040715299ffa1304bea544ca
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">Hi A=
ll,</font></div><div><font face=3D"&#39;courier new&#39;, monospace" color=
=3D"#000099"><br></font></div><div><font face=3D"&#39;courier new&#39;, mon=
ospace" color=3D"#000099">Following is the snippet of text from rfc6455 and=
 &quot;draft-ibc-sipcore-sip-websocket-02.txt&quot;.</font></div>

<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"><br>=
</font></div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#0000=
99">&lt;draft-ibc-sipcore-sip-websocket-02.txt&gt;=A0<br clear=3D"all"></fo=
nt><div>

<pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D"&#39=
;courier new&#39;, monospace" color=3D"#000099">2.  Terminology

   The key words &quot;MUST&quot;, &quot;MUST NOT&quot;, &quot;REQUIRED&quo=
t;, &quot;SHALL&quot;, &quot;SHALL NOT&quot;,
   &quot;SHOULD&quot;, &quot;SHOULD NOT&quot;, &quot;RECOMMENDED&quot;, &qu=
ot;MAY&quot;, and &quot;OPTIONAL&quot; in this
   document are to be interpreted as described in [RFC2119].
</font></pre><font face=3D"&#39;courier new&#39;, monospace" color=3D"#0000=
99">&lt;draft-ibc-sipcore-sip-websocket-02.txt&gt;
</font></div><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"=
#000099"><br></font></div><div><font face=3D"&#39;courier new&#39;, monospa=
ce" color=3D"#000099">&lt;rfc6455&gt;</font></div><div><pre style=3D"word-w=
rap:break-word;white-space:pre-wrap">
<font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099">2.  Confo=
rmance Requirements

   All diagrams, examples, and notes in this specification are non-
   normative, as are all sections explicitly marked non-normative.
   Everything else in this specification is normative.

   The key words &quot;MUST&quot;, &quot;MUST NOT&quot;, &quot;REQUIRED&quo=
t;, &quot;SHALL&quot;, &quot;SHALL NOT&quot;,
   &quot;SHOULD&quot;, &quot;SHOULD NOT&quot;, &quot;RECOMMENDED&quot;, &qu=
ot;MAY&quot;, and &quot;OPTIONAL&quot; in this
   document are to be interpreted as described in [RFC2119].
</font></pre><font face=3D"&#39;courier new&#39;, monospace" color=3D"#0000=
99">&lt;/rfc6455&gt;
</font></div><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"=
#000099"><br></font></div><div><font face=3D"&#39;courier new&#39;, monospa=
ce" color=3D"#000099">Usage of text &quot;<span style=3D"white-space:pre-wr=
ap">_This section is non-normative._</span>&quot; will not convey same mean=
ing in both the documents.</font></div>

<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"><br>=
</font></div><div><font face=3D"&#39;courier new&#39;, monospace" color=3D"=
#000099">Please add the first paragraph of sec.2,rfc6455 to sec.2 of this d=
raft. Otherwise the draft would be confusing about usage of the text &quot;=
<span style=3D"white-space:pre-wrap">_This section is non-normative._&quot;=
</span></font></div>

<div><font face=3D"&#39;courier new&#39;, monospace" color=3D"#000099"><spa=
n style=3D"white-space:pre-wrap"><br></span></font></div><div><font face=3D=
"&#39;courier new&#39;, monospace" color=3D"#000099"><span style=3D"white-s=
pace:pre-wrap">I know this is minor comment, but required to make the draft=
 clearer.</span></font></div>

<div><span style=3D"white-space:pre-wrap"><font face=3D"&#39;courier new&#3=
9;, monospace" color=3D"#000099"><br></font></span></div><font face=3D"&#39=
;courier new&#39;, monospace"><font color=3D"#000099">Thanks,</font></font>=
<div>
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace"><b>=
Nataraju A.B.</b></font></font></div>
<br>

--f46d040715299ffa1304bea544ca--

From ibc@aliax.net  Fri Apr 27 02:21:29 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 27 Apr 2012 11:21:08 +0200
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - editorial comments
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2012/4/27 Nataraju A.B <nataraju.sip@gmail.com>:
> Usage of text "_This section is non-normative._" will not convey same
> meaning in both the documents.
>
> Please add the first paragraph of sec.2,rfc6455 to sec.2 of this draft.
> Otherwise the draft would be confusing about usage of the text "_This
> section is non-normative._"
>
> I know this is minor comment, but required to make the draft clearer.

Hi Nataraju, we will add it in next revision of the draft.

Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From dwing@cisco.com  Fri Apr 27 07:50:40 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Rick van Rein'" <rick@openfortress.nl>
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Subject: Re: [sipcore] Proposal: sip6 URI scheme
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> -----Original Message-----
> From: Rick van Rein [mailto:rick@openfortress.nl]
> Sent: Thursday, April 26, 2012 10:30 PM
> To: Dan Wing
> Cc: 'Rick van Rein'; uri-review@ietf.org; sipcore@ietf.org
> Subject: Re: [sipcore] Proposal: sip6 URI scheme
> 
> Hello,
> 
> > > Your feedback is kindly welcomed.
> >
> > It's a layering violation, and -- worse -- will cause far more harm
> > than it could possibly help.  Please don't do this.
> 
> Thanks all for looking into this proposal.  It is clear that it will
> not
> gain wide acceptance, and should not be taken any further.
> 
> I'm withdrawing the proposal.

I do like the attention to IPv6, though -- don't get me wrong!  I
expect ICE is sufficient to find a working IPv6 or working IPv4
media path, and choosing between them.  If it is deficient, please
point it out, as ICE is being used by WEBRTC on the Internet (where
the IPv4/IPv6 address selection problem will be more severe than
within controlled networks).

-d



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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "sipcore@ietf.org" <sipcore@ietf.org>
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From nataraju.sip@gmail.com  Fri Apr 27 21:26:07 2012
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Date: Sat, 28 Apr 2012 09:56:02 +0530
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From: "Nataraju A.B" <nataraju.sip@gmail.com>
To: =?ISO-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: "SIPCORE \(Session Initiation Protocol Core\) WG" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - editorial comments
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--f46d04016a775f94df04beb59e67
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A better option is to use the default notion used in most of the rfc's.
That is, by default consider complete text/information in draft/rfc is
non-normative (mandatory to follow, but with different strength based on
SHOULD, MUST, etc....) and normative (informative) text is specified by an
indented TAB (this is optional to satisfy)

<websocket-02.txt>

5.3.  Locating a SIP Server

   RFC 3263 [RFC3263] specifies the procedures which should be followed
   by SIP entities for locating SIP servers.  This specification defines
   the NAPTR service value "SIP+D2W" for SIP WebSocket Servers that
   support plain WebSocket transport and "SIPS+D2W" for SIP WebSocket
   Servers that support secure WebSocket transport.

      Unfortunately neither JavaScript stacks nor WebSocket stacks
      running in current web browsers are capable of performing DNS
      NAPTR/SRV queries.

</websocket-02.txt>

Here the 2nd paragraph is normative text, hence it is only information
to implementer. But would not lead to any inter-operability issue, if
not followed religiously.


On Fri, Apr 27, 2012 at 2:51 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> 2012/4/27 Nataraju A.B <nataraju.sip@gmail.com>:
> > Usage of text "_This section is non-normative._" will not convey same
> > meaning in both the documents.
> >
> > Please add the first paragraph of sec.2,rfc6455 to sec.2 of this draft.
> > Otherwise the draft would be confusing about usage of the text "_This
> > section is non-normative._"
> >
> > I know this is minor comment, but required to make the draft clearer.
>
> Hi Nataraju, we will add it in next revision of the draft.
>
> Thanks a lot.
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>



--=20
Thanks,
Nataraju A.B.

--f46d04016a775f94df04beb59e67
Content-Type: text/html; charset=ISO-8859-1
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<div class=3D"gmail_extra">A better option is to use the default notion use=
d in most of the rfc&#39;s. That is, by default consider complete text/info=
rmation in draft/rfc is non-normative (mandatory to follow, but with differ=
ent strength based on SHOULD, MUST, etc....) and normative (informative) te=
xt is specified by an indented TAB (this is optional to satisfy)</div>
<div class=3D"gmail_extra"><br></div><div class=3D"gmail_extra"><div class=
=3D"gmail_extra">&lt;websocket-02.txt&gt;</div><pre style=3D"word-wrap:brea=
k-word;white-space:pre-wrap">5.3.  Locating a SIP Server

   RFC 3263 [RFC3263] specifies the procedures which should be followed
   by SIP entities for locating SIP servers.  This specification defines
   the NAPTR service value &quot;SIP+D2W&quot; for SIP WebSocket Servers th=
at
   support plain WebSocket transport and &quot;SIPS+D2W&quot; for SIP WebSo=
cket
   Servers that support secure WebSocket transport.

      Unfortunately neither JavaScript stacks nor WebSocket stacks
      running in current web browsers are capable of performing DNS
      NAPTR/SRV queries.
</pre><div class=3D"gmail_extra">&lt;/websocket-02.txt&gt;</div><div class=
=3D"gmail_extra"><br></div>Here the 2nd paragraph is normative text, hence =
it is only information to=A0implementer. But would not lead to any inter-op=
erability issue, if not=A0followed=A0religiously.</div>
<div class=3D"gmail_extra"><br class=3D"Apple-interchange-newline"><br><div=
 class=3D"gmail_quote">On Fri, Apr 27, 2012 at 2:51 PM, I=F1aki Baz Castill=
o <span dir=3D"ltr">&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">=
ibc@aliax.net</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">2012/4/27 Nataraju A.B &lt;<a href=3D"mailto=
:nataraju.sip@gmail.com">nataraju.sip@gmail.com</a>&gt;:<br>
<div class=3D"im">&gt; Usage of text &quot;_This section is non-normative._=
&quot; will not convey same<br>
&gt; meaning in both the documents.<br>
&gt;<br>
&gt; Please add the first paragraph of sec.2,rfc6455 to sec.2 of this draft=
.<br>
&gt; Otherwise the draft would be confusing about usage of the text &quot;_=
This<br>
&gt; section is non-normative._&quot;<br>
&gt;<br>
&gt; I know this is minor comment, but required to make the draft clearer.<=
br>
<br>
</div>Hi Nataraju, we will add it in next revision of the draft.<br>
<br>
Thanks a lot.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net">ibc@aliax.net</a>&gt;<br>
</font></span></blockquote></div><br><br clear=3D"all"><div><br></div>-- <b=
r><font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" s=
ize=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&=
#39;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></di=
v>
<br>
</div>

--f46d04016a775f94df04beb59e67--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sat, 28 Apr 2012 16:56:05 +0200
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Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - editorial comments
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Hi Nataraju,

After reading various mails from you about this topic I think that you
are confusing the meaning of "normative" and "informative" words (hope
I'm not wrong).

- "normative" text is a text that creates rules.
- "informative" text does not create rules but just helps or adds some
useful information.


But by reading your text:


2012/4/28 Nataraju A.B <nataraju.sip@gmail.com>:
> consider complete text/information in draft/rfc is
> non-normative (mandatory to follow, but with different strength based on
> SHOULD, MUST, etc....)

That is "normative" instead of "non-normative".



> and normative (informative) text is specified by an
> indented TAB (this is optional to satisfy)

That is "informative" and therefore also "non-normative".



Regards.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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On 4/18/12 8:51 PM, IĆ±aki Baz Castillo wrote:
>> >  if we want to use SIP URI with any other secure communication mechanism like
>> >  WSS, might require
>> >  update to RFC5630. like
>> >  <rfc5630>  sec 3.1.3.
>> >
>> >      If one wants to use "best-effort TLS/WSS" for SIP, one just needs to use
>> >      a SIP URI, and send the request over TLS ** or any other secure transport
>> >  mechanism like WSS**
>> >
>> >  </rfc5630>
> TLS is a extra-layer on top of TCP and also on top of WebSocket.
TLS is not on top of WebSocket, but it is the other way around.
Once again:
WebSocket is not a transport protocol, but a protocol that is used to 
transport content as HTTP is.
WebSocket can run on top of TCP (WS) or on top of TLC (WSS).

So once again please do not say WebSocket is a transport protocol 
because it is not!

/Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com


>   When
> applied on top of TCP transport it is commonly named "TLS", but it
> should be better to call it "TLS-TCP" (in the same way that TLS over
> SCTP is named "TLS-SCTP" for Via transport value in RFC 4168.
>
>
>
>
>



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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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On 4/18/12 5:24 PM, IĆ±aki Baz Castillo wrote:
> 2012/4/18 Salvatore Loreto<salvatore.loreto@ericsson.com>:
>> The important thing to standardize from a SIPCore prospective is the
>> interaction/translations that happens between the WebSocket server and the
>> SIP proxy that are two different logical entities.
> Hi Salvatore, I don't consider them two different logical entities.
> The draft 02 states the following in "Definitions" section (which was
> added after your review):
>
>     SIP WebSocket Server:  A SIP entity capable of listening for inbound
>           connections from WebSocket clients and speaking the WebSocket
>           SIP Sub-Protocol as defined by this document.
>
>
> So when the WebSocket message/arrives to the WebSocket listening
> server, the application on top of it (a SIP proxy/UAS) should extract
> the SIP message from the WS message, parse it and process it in the
> same way that if the SIP message arrived via another SIP transport.
I understand clearly your point, but perhaps I wasn't able to express so 
good mine.
Once more: I could use your argument with almost all the protocols and say
for example XMPP (just s/XMPP/SIP in your above text):

    when a XMPP message/arrives to the WebSocket listening server, the
    application
    on top of it (a SIP proxy/UAS) should extract the SIP message from
    the XMPP message,
    parse it and process it in the same way that is the SIP message
    message arrived via another
    SIP transport.

BTW: "SIP transport" is also an interesting definition, indeed WebSocket 
is not a transport protocol
but can be used to transport SIP message as well as other stuff.

I would hear the opinion of the Chairs about this,
if they are OK to define (and talk in this draft of) WebSocket I will 
shut up!
Chairs?

> Just it. And for sending responses the SIP proxy/UAS should reuse the
> existing WS connection (which implies creating a WS message and
> writting the SIP response inside). Just it.
>
>
> The same is true for a SIP WebSocket Client:
>
>     SIP WebSocket Client:  A SIP entity capable of opening outbound
>           connections with WebSocket servers and speaking the WebSocket
>           SIP Sub-Protocol as defined by this document.
>
>
> For example, if the SIP application is a JavaScript code running on a
> web browser, the SIP stack (JS code) sends and receives SIP messages
> to/from its transport layer (as in any other SIP transport), but in
> this case the transport layer is a WebSocket client interface which
> generates a WS message (using the JS WebSocket API) and sends it. Same
> occurs when receiving an incoming SIP request/response which is
> extracted by the JS app (the SIP stack) from the received WS message.
>
>
> So honestly I see no "interaction/translations that happens between
> the WebSocket server and the SIP proxy".
So honestly I don't buy your explanation.

Salvatore

-- 
Salvatore Loreto, PhD
www.sloreto.com



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    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    On 4/18/12 5:24 PM, IĆ±aki Baz Castillo wrote:
    <blockquote
cite="mid:CALiegf=V35tRhaKvk0AWJkkVVPOPBX+fX0qdHcpABFJnr70Udw@mail.gmail.com"
      type="cite">
      <pre wrap="">2012/4/18 Salvatore Loreto <a class="moz-txt-link-rfc2396E" href="mailto:salvatore.loreto@ericsson.com">&lt;salvatore.loreto@ericsson.com&gt;</a>:
</pre>
      <blockquote type="cite">
        <pre wrap="">The important thing to standardize from a SIPCore prospective is the
interaction/translations that happens between the WebSocket server and the
SIP proxy that are two different logical entities.
</pre>
      </blockquote>
      <pre wrap="">
Hi Salvatore, I don't consider them two different logical entities.
The draft 02 states the following in "Definitions" section (which was
added after your review):

   SIP WebSocket Server:  A SIP entity capable of listening for inbound
         connections from WebSocket clients and speaking the WebSocket
         SIP Sub-Protocol as defined by this document.


So when the WebSocket message/arrives to the WebSocket listening
server, the application on top of it (a SIP proxy/UAS) should extract
the SIP message from the WS message, parse it and process it in the
same way that if the SIP message arrived via another SIP transport.</pre>
    </blockquote>
    I understand clearly your point, but perhaps I wasn't able to
    express so good mine.<br>
    Once more: I could use your argument with almost all the protocols
    and say<br>
    for example XMPP (just s/XMPP/SIP in your above text): <br>
    <blockquote>when a XMPP message/arrives to the WebSocket listening
      server, the application<br>
      on top of it (a SIP proxy/UAS) should extract the SIP message from
      the XMPP message,<br>
      parse it and process it in the same way that is the SIP message
      message arrived via another<br>
      SIP transport.<br>
    </blockquote>
    BTW: "SIP transport" is also an interesting definition, indeed
    WebSocket is not a transport protocol<br>
    but can be used to transport SIP message as well as other stuff.<br>
    <br>
    I would hear the opinion of the Chairs about this,<br>
    if they are OK to define (and talk in this draft of) WebSocket I
    will shut up!<br>
    Chairs?<br>
    Ā <br>
    <blockquote
cite="mid:CALiegf=V35tRhaKvk0AWJkkVVPOPBX+fX0qdHcpABFJnr70Udw@mail.gmail.com"
      type="cite">
      <pre wrap="">
Just it. And for sending responses the SIP proxy/UAS should reuse the
existing WS connection (which implies creating a WS message and
writting the SIP response inside). Just it.


The same is true for a SIP WebSocket Client:

   SIP WebSocket Client:  A SIP entity capable of opening outbound
         connections with WebSocket servers and speaking the WebSocket
         SIP Sub-Protocol as defined by this document.


For example, if the SIP application is a JavaScript code running on a
web browser, the SIP stack (JS code) sends and receives SIP messages
to/from its transport layer (as in any other SIP transport), but in
this case the transport layer is a WebSocket client interface which
generates a WS message (using the JS WebSocket API) and sends it. Same
occurs when receiving an incoming SIP request/response which is
extracted by the JS app (the SIP stack) from the received WS message.


So honestly I see no "interaction/translations that happens between
the WebSocket server and the SIP proxy".</pre>
    </blockquote>
    So honestly I don't buy your explanation.<br>
    <br>
    Salvatore<br>
    <pre class="moz-signature" cols="72">-- 
Salvatore Loreto, PhD
<a class="moz-txt-link-abbreviated" href="http://www.sloreto.com">www.sloreto.com</a></pre>
    <br>
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Subject: Re: [sipcore] Should we change mandatory-to-implement transports?
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On 4/20/12 6:03 PM, Paul Kyzivat wrote:
> I've been looking for a suitable place to jump back into this thread.
> Maybe this is as good a place as any. I've changed the subject to
> reflect the current focus of the discussion.
>
> We have two ways forward on this - the narrow way, and the broad way.
>
> The narrow way it to craft a specific exemption for implementers of the
> websocket transport - that they aren't required to implement UDP and
> TCP. Such an exemption could be included in the websocket transport draft.

I would go for the narrow, at least from a WebSocket prospective.

 From a WebSocket point of view,
what we are defining in SIPCore is ***SIP as a WebSocket subprotocol***
indeed we are going also to register into IANA a value for it.

If we take this approach, everything will become simpler.

my 2 cents

/Sal

>
> The broad way is to do a standalone update to 3261 that removes/relaxes
> the requirement to implement UDP and TCP transports.
>
> The narrow way will be easier to achieve. Its impact is narrower and
> easier to understand. So there is likely to be much less objection to
> it. And of course it doesn't require a separate document.
>
> The broad way will cover many more cases all in one swipe of the pen.
> But that means that it will require much more scrutiny to avoid
> unintended consequences. As a result it will be harder to craft and
> harder to pass. My main concern is that it has to potential to erect new
> barriers to interoperability. The point of having some must-implement
> transports is to ensure some common basis for interop. This has the
> potential to lose that. I am myself uncertain if I would support such a
> change. This would depend on some very careful analysis.
>
> I can't promise that the chairs or the AD will even approve of the broad
> approach.
>
> (I guess there is a third way - to do*nothing*  about the
> mandatory-to-implement transport requirement. Leave
> javascript-browser-based sip implementations that can't do UDP or TCP
> non-compliant. IMO that is an irresponsible direction that I don't want
> to consider.)
>
> Given all of that, I would like to have some discussion of which way the
> group would like to go. Please do more that vote - actually make the
> case for why you want to go the way you do.
>
> 	Thanks,
> 	Paul [as co-chair]



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    On 4/20/12 6:03 PM, Paul Kyzivat wrote:
    <blockquote cite="mid:4F917AD4.2040500@alum.mit.edu" type="cite">
      <pre wrap="">I've been looking for a suitable place to jump back into this thread.
Maybe this is as good a place as any. I've changed the subject to 
reflect the current focus of the discussion.

We have two ways forward on this - the narrow way, and the broad way.

The narrow way it to craft a specific exemption for implementers of the 
websocket transport - that they aren't required to implement UDP and 
TCP. Such an exemption could be included in the websocket transport draft.</pre>
    </blockquote>
    <br>
    I would go for the narrow, at least from a WebSocket prospective.<br>
    <br>
    From a WebSocket point of view,<br>
    what we are defining in SIPCore is ***SIP as a WebSocket
    subprotocol***<br>
    indeed we are going also to register into IANA a value for it.<br>
    <br>
    If we take this approach, everything will become simpler.<br>
    <br>
    my 2 cents<br>
    <br>
    /Sal<br>
    <br>
    <blockquote cite="mid:4F917AD4.2040500@alum.mit.edu" type="cite">
      <pre wrap="">

The broad way is to do a standalone update to 3261 that removes/relaxes 
the requirement to implement UDP and TCP transports.

The narrow way will be easier to achieve. Its impact is narrower and 
easier to understand. So there is likely to be much less objection to 
it. And of course it doesn't require a separate document.

The broad way will cover many more cases all in one swipe of the pen. 
But that means that it will require much more scrutiny to avoid 
unintended consequences. As a result it will be harder to craft and 
harder to pass. My main concern is that it has to potential to erect new 
barriers to interoperability. The point of having some must-implement 
transports is to ensure some common basis for interop. This has the 
potential to lose that. I am myself uncertain if I would support such a 
change. This would depend on some very careful analysis.

I can't promise that the chairs or the AD will even approve of the broad 
approach.

(I guess there is a third way - to do <b class="moz-txt-star"><span class="moz-txt-tag">*</span>nothing<span class="moz-txt-tag">*</span></b> about the 
mandatory-to-implement transport requirement. Leave 
javascript-browser-based sip implementations that can't do UDP or TCP 
non-compliant. IMO that is an irresponsible direction that I don't want 
to consider.)

Given all of that, I would like to have some discussion of which way the 
group would like to go. Please do more that vote - actually make the 
case for why you want to go the way you do.

	Thanks,
	Paul [as co-chair]</pre>
    </blockquote>
    <br>
    <br>
  </body>
</html>

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From nataraju.sip@gmail.com  Sun Apr 29 02:12:22 2012
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Inaki,

You are right, I was wrong while using the words normative and informative.

On Sat, Apr 28, 2012 at 8:26 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote=
:

> Hi Nataraju,
>
> After reading various mails from you about this topic I think that you
> are confusing the meaning of "normative" and "informative" words (hope
> I'm not wrong).
>
> - "normative" text is a text that creates rules.
> - "informative" text does not create rules but just helps or adds some
> useful information.
>
>
> But by reading your text:
>
>
> 2012/4/28 Nataraju A.B <nataraju.sip@gmail.com>:
> > consider complete text/information in draft/rfc is
> > non-normative (mandatory to follow, but with different strength based o=
n
> > SHOULD, MUST, etc....)
>
> That is "normative" instead of "non-normative".
>
>
>
> > and normative (informative) text is specified by an
> > indented TAB (this is optional to satisfy)
>
> That is "informative" and therefore also "non-normative".
>
>
>
> Regards.
>
>
> --
> I=F1aki Baz Castillo
> <ibc@aliax.net>
>



--=20
Thanks,
Nataraju A.B.

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Inaki,<div><br></div><div>You are right, I was wrong while using the words =
normative and informative.=A0</div><div><br></div><div><div class=3D"gmail_=
quote">

On Sat, Apr 28, 2012 at 8:26 PM, I=F1aki Baz Castillo <span dir=3D"ltr">&lt=
;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt;</=
span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">



Hi Nataraju,<br>
<br>
After reading various mails from you about this topic I think that you<br>
are confusing the meaning of &quot;normative&quot; and &quot;informative&qu=
ot; words (hope<br>
I&#39;m not wrong).<br>
<br>
- &quot;normative&quot; text is a text that creates rules.<br>
- &quot;informative&quot; text does not create rules but just helps or adds=
 some<br>
useful information.<br>
<br>
<br>
But by reading your text:<br>
<br>
<br>
2012/4/28 Nataraju A.B &lt;<a href=3D"mailto:nataraju.sip@gmail.com" target=
=3D"_blank">nataraju.sip@gmail.com</a>&gt;:<br>
<div>&gt; consider complete text/information in draft/rfc is<br>
&gt; non-normative (mandatory to follow, but with different strength based =
on<br>
&gt; SHOULD, MUST, etc....)<br>
<br>
</div>That is &quot;normative&quot; instead of &quot;non-normative&quot;.<b=
r>
<div><br>
<br>
<br>
&gt; and normative (informative) text is specified by an<br>
&gt; indented TAB (this is optional to satisfy)<br>
<br>
</div>That is &quot;informative&quot; and therefore also &quot;non-normativ=
e&quot;.<br>
<br>
<br>
<br>
Regards.<br>
<div><div><br>
<br>
--<br>
I=F1aki Baz Castillo<br>
&lt;<a href=3D"mailto:ibc@aliax.net" target=3D"_blank">ibc@aliax.net</a>&gt=
;<br>
</div></div></blockquote></div><br><br clear=3D"all"><div><br></div>-- <br>=
<font color=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" siz=
e=3D"1">Thanks,</font></font><div><font color=3D"#000099"><font face=3D"&#3=
9;courier new&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>



<br>
</div>

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Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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2012/4/28 Salvatore Loreto <salvatore.loreto@ericsson.com>:
> WebSocket is not a transport protocol, but a protocol that is used to
> transport content as HTTP is.
> WebSocket can run on top of TCP (WS) or on top of TLC (WSS).
>
> So once again please do not say WebSocket is a transport protocol because=
 it
> is not!

Hi Salvatore,let me clarify this subject. The draft defines two things:

1) A WebSocket Sub-Protocol called "sip".

2) A new SIP transport called "WebSocket".


The SIP transport defined by this draft is WebSocket. WebSocket is,
indeed, a layer on top of a network transport (layer 3) but it's
designed to work as a *transport* layer since it defines a framing
mechanism to send application messages and control mesages in both
directions. Applications (i.e. JavaScript scripts, PHP/Java web apps)
use WebSocket as a transport layer.

There are other cases in which a transport runs on top of other
transports. For example, in the recent IETF #83 the RTCWEB WG approved
(or discussed) SCTP-over-DTLS-over-ICE-over-UDP as the transport for
the WebRTC Data-Channel mechanism. In some cases a "transport" does
not necessarily mean "layer 3 network transport".

So, regardless WebSocket is not a layer 3 transport, it's used as a
transport by this specification and therefore it's considered a "SIP
transport".


Regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] Stateless or Transaction stateful proxy possible in Websocket transport? [RE: Call for Adoption: draft-ibc-sipcore-sip-websocket-02]
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2012/4/28 Salvatore Loreto <salvatore.loreto@ericsson.com>:
> BTW: "SIP transport" is also an interesting definition, indeed WebSocket =
is
> not a transport protocol
> but can be used to transport SIP message as well as other stuff.
>
> I would hear the opinion of the Chairs about this,
> if they are OK to define (and talk in this draft of) WebSocket I will shu=
t
> up!
> Chairs?


Hi Salvatore, I expect that this subject has been already taked into
account by chairs since they declared (after IETF 83) a new SIPCORE WG
milestone:

  Apr 2013  WebSockets transport definition for SIP to the IESG
(proposed standard)

:)




> > So honestly I see no "interaction/translations that happens between
> > the WebSocket server and the SIP proxy".

> So honestly I don't buy your explanation.

Without more arguments I cannot give a better explanation. However I
would like to clarify that this is not about having a Apache web
server with "mod_websocket" enabled, and passing SIP messages received
via the WebSocket connection to a "real" SIP proxy using some
proprietary/custom communication. This is not what this specification
attempts to achieve.

The draft states (in "Definitions" section):

   SIP WebSocket Server:  A SIP entity capable of listening for inbound
         connections from WebSocket clients and speaking the WebSocket
         SIP Sub-Protocol as defined by this document.

So, this SIP entity includes a WebSocket server for SIP-over-WebSocket
in the same way it includes a TCP server for SIP-over-TCP or a UDP
socket for SIP-over-UDP. And therefore the SIP entity and the
WebSocket server are not two logical independent entities (not when we
are speaking about a "SIP WebSocket Client" or a "SIP WebSocket
Server"). This is what the draft talks about, and not about
communication between WebSocket servers and SIP entities.



Best regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] draft-ibc-sipcore-sip-websocket-02.txt - Review comments (Nataraju A B)
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On 4/29/12 7:07 PM, IĆ±aki Baz Castillo wrote:
> 2012/4/28 Salvatore Loreto<salvatore.loreto@ericsson.com>:
>> WebSocket is not a transport protocol, but a protocol that is used to
>> transport content as HTTP is.
>> WebSocket can run on top of TCP (WS) or on top of TLC (WSS).
>>
>> So once again please do not say WebSocket is a transport protocol because it
>> is not!
> Hi Salvatore,let me clarify this subject. The draft defines two things:
>
> 1) A WebSocket Sub-Protocol called "sip".
that is OK and I am not arguing against point 1.
>
> 2) A new SIP transport called "WebSocket".
I am not comfortable with this point, but maybe it is just me be a purist

>
>
> The SIP transport defined by this draft is WebSocket. WebSocket is,
> indeed, a layer on top of a network transport (layer 3) but it's
> designed to work as a *transport* layer since it defines a framing
> mechanism to send application messages and control mesages in both
> directions. Applications (i.e. JavaScript scripts, PHP/Java web apps)
> use WebSocket as a transport layer.
my point is that there is a difference between an application transport 
layer
and a transport protocol!

>
> There are other cases in which a transport runs on top of other
> transports. For example, in the recent IETF #83 the RTCWEB WG approved
> (or discussed) SCTP-over-DTLS-over-ICE-over-UDP as the transport for
> the WebRTC Data-Channel mechanism. In some cases a "transport" does
> not necessarily mean "layer 3 network transport".

this is slightly different
in RTCWeb the Data-Channel all the stack if formed by different 
transport protocols
however the fact the part of them run in the user-land space and then 
encapsulated in UDP
introduce some limitation to the SCTP futures etc.
but the UDP is still used in the full way on each side

there is a huge list of examples in literature of transport protocols 
designed and run on top
of UDP due to the flexibility of this transport protocol (e.g. pseud TCP 
protocols etc.)

>
> So, regardless WebSocket is not a layer 3 transport, it's used as a
> transport by this specification and therefore it's considered a "SIP
> transport".
>
> Regards.
>
>
>


From salvatore.loreto@ericsson.com  Sun Apr 29 11:02:35 2012
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Subject: [sipcore] WebSocket in a Browser and outside it (was Re: draft-ibc-sipcore-sip-websocket-02.txt )
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--------------000604080400040109030903
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5.2.3.  Sending Responses

    This specification updates the section 18.2.2 "Sending Responses" in
    [RFC3261] by adding the following:

    o  If the Via "sent-protocol" is "WS" or "WSS" the response MUST be
       sent using the existing WebSocket connection to the source of the
       original request that created the transaction, if that connection
       is still open.  If that connection is no longer open, the server
       SHOULD NOT attempt to open a WebSocket connection for sending the
       response.


       This is due to the nature of the WebSocket protocol in which just
       the WebSocket client can establish a connection with the WebSocket
       server.  Typically a WebSocket client does not listen for inbound
       connections and WebSocket servers do not open outbound
       connections.




The last paragraph of this section confuses the limitation of the 
WebSocketProtocol
implemented in a Browser as a general limitation.
It is tautological saying that a WebSocket Client can act only as a client,
however in theory a UA (outside a Browser sandbox) could act as both a 
WebSocket client and WebSocket server.

As the authors have several time claimed that they want define the usage 
of SIP on top of WebSocket for the most
general scenario and not only for the Browser/JavaScript, I would invite 
to be more clear

/Salvatore


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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <pre>5.2.3.  Sending Responses

   This specification updates the section 18.2.2 "Sending Responses" in
   [RFC3261] by adding the following:

   o  If the Via "sent-protocol" is "WS" or "WSS" the response MUST be
      sent using the existing WebSocket connection to the source of the
      original request that created the transaction, if that connection
      is still open.  If that connection is no longer open, the server
      SHOULD NOT attempt to open a WebSocket connection for sending the
      response.


      This is due to the nature of the WebSocket protocol in which just
      the WebSocket client can establish a connection with the WebSocket
      server.  Typically a WebSocket client does not listen for inbound
      connections and WebSocket servers do not open outbound
      connections.



</pre>
    The last paragraph of this section confuses the limitation of the
    WebSocketProtocol<br>
    implemented in a Browser as a general limitation.<br>
    It is
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <span id="result_box" class="short_text" lang="en"><span class="hps">tautological
        saying that a WebSocket Client can act only as a client,</span></span><br>
    however in theory a UA (outside a Browser sandbox) could act as both
    a WebSocket client and WebSocket server.<br>
    <br>
    As the authors have several time claimed that they want define the
    usage of SIP on top of WebSocket for the most<br>
    general scenario and not only for the Browser/JavaScript, I would
    invite to be more clear<br>
    <br>
    /Salvatore<br>
    <br>
  </body>
</html>

--------------000604080400040109030903--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
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2012/4/29 Salvatore Loreto <salvatore.loreto@ericsson.com>:
>> 2) A new SIP transport called "WebSocket".
>
> I am not comfortable with this point, but maybe it is just me be a purist

A "SIP transport" involves VIA transport values, SIP URI's transport
parameter value, DNS resolution (locating servers, NAPTR/SRV records,
default ports), secure or not secure transport, reliable transport or
not (so different timers in RFC 3261 are taken depending on that), and
so on.

That is what I mean with "SIP transport", and not just the layer
through which messages are transmitted/carried/transported.

Is it ok with this definition? :)

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Subject: Re: [sipcore] WebSocket in a Browser and outside it (was Re: draft-ibc-sipcore-sip-websocket-02.txt )
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2012/4/29 Salvatore Loreto <salvatore.loreto@ericsson.com>:
> 5.2.3.  Sending Responses
>
>    This specification updates the section 18.2.2 "Sending Responses" in
>    [RFC3261] by adding the following:
>
>    o  If the Via "sent-protocol" is "WS" or "WSS" the response MUST be
>       sent using the existing WebSocket connection to the source of the
>       original request that created the transaction, if that connection
>       is still open.  If that connection is no longer open, the server
>       SHOULD NOT attempt to open a WebSocket connection for sending the
>       response.
>
>
>       This is due to the nature of the WebSocket protocol in which just
>       the WebSocket client can establish a connection with the WebSocket
>       server.  Typically a WebSocket client does not listen for inbound
>       connections and WebSocket servers do not open outbound
>       connections.
>
>
>
> The last paragraph of this section confuses the limitation of the
> WebSocketProtocol
> implemented in a Browser as a general limitation.
> It is tautological saying that a WebSocket Client can act only as a clien=
t,
> however in theory a UA (outside a Browser sandbox) could act as both a
> WebSocket client and WebSocket server.
>
> As the authors have several time claimed that they want define the usage =
of
> SIP on top of WebSocket for the most
> general scenario and not only for the Browser/JavaScript, I would invite =
to
> be more clear


Very good point, the authors already discussed about that exact point
and decided to keep it for now until further discussion takes place.

IMHO you are right and the best I can suggest is just removing section
5.2.3. As you say, if a SIP UA implements a WebSocket client and
server, it has no problems in receiving responses in a new inbound WS
connection.

Thanks a lot.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From internet-drafts@ietf.org  Mon Apr 30 12:28:37 2012
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc3265bis-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : SIP-Specific Event Notification
	Author(s)       : Adam Roach
	Filename        : draft-ietf-sipcore-rfc3265bis-09.txt
	Pages           : 53
	Date            : 2012-04-30

   This document describes an extension to the Session Initiation
   Protocol (SIP) defined by RFC 3261.  The purpose of this extension is
   to provide an extensible framework by which SIP nodes can request
   notification from remote nodes indicating that certain events have
   occurred.

   Note that the event notification mechanisms defined herein are NOT
   intended to be a general-purpose infrastructure for all classes of
   event subscription and notification.

   This document represents a backwards-compatible improvement on the
   original mechanism described by RFC 3265, taking into account several
   years of implementation experience.  Accordingly, this document
   obsoletes RFC 3265.  This document also updates RFC 4660 slightly to
   accommodate some small changes to the mechanism that were discussed
   in that document.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-09.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-09.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc3265bis/


From adam@nostrum.com  Mon Apr 30 12:28:46 2012
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Subject: Re: [sipcore] RFC3265bis: reason code registration policy
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[as chair]

Lacking any feedback on this topic, the document has been revised 
according to IESG feedback. The new text appears in section 7.2 of 
draft-ietf-sipcore-rfc3265bis-09:

    Registrations with the IANA include the reason code being registered
    and a reference to a published document which describes the event
    package.  Insertion of such values takes place as part of the RFC
    publication process or as the result of inter-SDO liaison activity,
    the result of which will be publication of an associated RFC.  New
    reason codes must conform to the syntax of the ABNF "token" element
    defined in [RFC3261].

At this point, any objections to the text should be brought to the 
attention of the SIPCORE chairs and our RAI AD (presently Robert Sparks).

Thank you.

/a

On 4/12/12 12:19 PM, Adam Roach wrote:
> [as document author]
>
> As you probably know, RFC3265bis has been handed off to the IESG for 
> pre-publication evaluation.
>
> Section 7.2 of the RFC3265bis document discusses the new IANA registry 
> for the "reason" code. The gen-art review of this document turned up 
> the following discrepancy:
>
>  * 'Following the policies outlined in "Guidelines for Writing an IANA
>    Considerations Section in RFCs" [RFC5226], new reason codes require
>    a Standards Action.'
>
>  * ' Insertion of such values takes place as part of the RFC
>    publication process or as the result of inter-SDO liaison activity.'
>
>
> The problem is that the "Standards Action" requirement is defined: 
> "Values are assigned only for Standards Track RFCs approved by the 
> IESG," which implies that inter-SDO liaising isn't sufficient for 
> adding values.
>
> I can't find any relic of this issue being discussed on the mailing 
> list, and have no specific recollection how we arrived at the current 
> text. Keep in mind that the need to expand these reason codes should 
> be very slight (we've only had one in nine years since 3265 was 
> published), and will almost certainly be paired with protocol changes 
> that RFC 5727 would require an RFC for anyway.
>
> Based on those two factors (and especially the second one), I think 
> what we probably want to do is remove the phrase "of as the result of 
> inter-SDO liaison activity," and leave the policy "Standards Action" 
> as currently stated.
>
> Please raise any objections to this proposed course of action on the 
> SIPCORE mailing list. I will make sure the results of any on-list 
> discussion are brought to the attention of the IESG.
>
> /a
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-rfc3265bis-09.txt
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[as a draft author]

This version represents revisions due to IESG feedback.

See 
<https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc3265bis/ballot/> 
for specific IESG comments leading to the changes.

See 
<http://tools.ietf.org/rfcdiff?url2=draft-ietf-sipcore-rfc3265bis-09.txt> for 
the actual draft changes.

Thank you.

/a

On 4/30/12 2:28 PM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.
>
> 	Title           : SIP-Specific Event Notification
> 	Author(s)       : Adam Roach
> 	Filename        : draft-ietf-sipcore-rfc3265bis-09.txt
> 	Pages           : 53
> 	Date            : 2012-04-30
>
>     This document describes an extension to the Session Initiation
>     Protocol (SIP) defined by RFC 3261.  The purpose of this extension is
>     to provide an extensible framework by which SIP nodes can request
>     notification from remote nodes indicating that certain events have
>     occurred.
>
>     Note that the event notification mechanisms defined herein are NOT
>     intended to be a general-purpose infrastructure for all classes of
>     event subscription and notification.
>
>     This document represents a backwards-compatible improvement on the
>     original mechanism described by RFC 3265, taking into account several
>     years of implementation experience.  Accordingly, this document
>     obsoletes RFC 3265.  This document also updates RFC 4660 slightly to
>     accommodate some small changes to the mechanism that were discussed
>     in that document.
>
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-09.txt
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> This Internet-Draft can be retrieved at:
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-rfc3265bis-09.txt
>
> The IETF datatracker page for this Internet-Draft is:
> https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc3265bis/
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: [sipcore] I-D Action: draft-ietf-sipcore-rfc4244bis-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Session Initiation Protocol Core Work=
ing Group of the IETF.

	Title           : An Extension to the Session Initiation Protocol (SIP) fo=
r Request History Information
	Author(s)       : Mary Barnes
                          Francois Audet
                          Shida Schubert
                          Detecon International Gmbh
                          Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-08.txt
	Pages           : 67
	Date            : 2012-04-30

   This document defines a standard mechanism for capturing the history
   information associated with a Session Initiation Protocol (SIP)
   request.  This capability enables many enhanced services by providing
   the information as to how and why a SIP request arrives at a specific
   application or user.  This document defines an optional SIP header
   field, History-Info, for capturing the history information in
   requests.  The document also defines SIP header field parameters for
   the History-Info and Contact header fields to tag the method by which
   the target of a request is determined.  In addition, this
   specification defines a value for the Privacy header field that
   directs the anonymization of values in the History-Info header field
   This document obsoletes RFC 4244.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-rfc4244bis-08.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-sipcore-rfc4244bis-08.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis/

