
From stpeter@stpeter.im  Thu Aug  1 00:01:36 2013
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From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] Comments on draft-ietf-stox-presence-00
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On 8/1/13 3:17 AM, Michael Lundberg wrote:
>  Peter,
> 
> The suggest text below makes sense to me.  In my opinion, this helps
> more clearly defining the presence mapping from XMPP to SIP for values
> such as 'away'.
> 
> I think there needs to be similar text for the second table (currently
> labeled Table 6), which states that the gateway SHOULD map the <show
> xmlns='jabber:client'> element in the PIDF XML to the <show/> element
> in the XMPP presence stanza.  This would only be for SIP clients that
> support that particular namespace, but it at least provides a
> recommended mapping in the opposite direction for clients that do
> support that namespace.

Yes, I think I already expressed agreement with that intent, so you'll
definitely see text about it in the next version of the spec (if no one
objects on the list, of course).

Peter


From yana@sip-communicator.org  Thu Aug  1 00:22:26 2013
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Subject: Re: [Stox] draft slides for core, presence, im, chat
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Hi Peter, Saul,

Thanks for integrating the last discussions in your slides. They're =
updated.

Yana

On Aug 1, 2013, at 8:49 AM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:

> OK, I have updated my slides:
>=20
> https://stpeter.im/files/ietf87-stox-core-presence.pdf
>=20
> Sorry it took me a bit longer than promised -- I wanted to incorporate
> some results of the list traffic yesterday.

>=20
> Note that I have not included all issues in the slides, only the =
issues
> that in my judgment merit a bit of face-to-face discussion.
>=20
> Peter
>=20
> On 7/31/13 6:39 AM, Peter Saint-Andre wrote:
>> This is just a warning that I'll be providing updated slides to =
address
>> some points that Sa=FAl made in his reviews. I'll try to do that as =
early
>> today as possible.
>>=20
>> On 7/30/13 2:59 PM, Yana Stamcheva wrote:
>>> Hi Peter,
>>>=20
>>> Thanks! Your new slides are uploaded.
>>>=20
>>> Yana
>>>=20
>>> On Jul 30, 2013, at 2:15 PM, Peter Saint-Andre <stpeter@stpeter.im> =
wrote:
>>>=20
>>>> After a hallway discussion with Robert Sparks, I have updated these
>>>> slides. I will post to the list about the issue he raised.
>>>>=20
>>>> On 7/24/13 8:09 PM, Peter Saint-Andre wrote:
>>>>> Draft slides for discussion of the -core, -presence, -im, and =
-chat
>>>>> specfications are here:
>>>>>=20
>>>>> https://stpeter.im/files/ietf87-stox-core-presence.pdf
>>>>>=20
>>>>> As you can see, there are very few open issues. If folks provide
>>>>> further feedback about these specs in the next few days, I'll =
update
>>>>> the slides accordingly. If not, mine will be a very short =
presentation
>>>>> and we'll have more time to discuss the -groupchat and -media =
I-Ds. :-)
>>>>>=20
>>>>> Peter
>>>>>=20


From saul@ag-projects.com  Thu Aug  1 00:42:48 2013
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>=20
>> Also, the priority in PIDF is a
>> float between 0 and 1, is it the same for XMPP?
>=20
> Hmm. We had text about that in RFC 3922, and I suggest we copy that to
> this draft (adjusting as necessary)...
>=20
>   An XMPP presence stanza MAY contain a <priority/> child element =
whose
>   value is an integer between -128 and +127.  The value of this =
element
>   MAY be mapped to the 'priority' attribute of the <contact/> child of
>   the PIDF <tuple/> element.  If the value of the XMPP <priority/>
>   element is negative, an XMPP-CPIM gateway MUST NOT map the value. =
The
>   range of allowable values for the PIDF 'priority' attribute is any
>   decimal number from zero to one inclusive, with a maximum of three
>   decimal places.  If an XMPP-CPIM gateway maps these values, it =
SHOULD
>   treat XMPP <priority>0</priority> as PIDF priority=3D'0' and XMPP
>   <priority>127</priority> as PIDF priority=3D'1', mapping =
intermediate
>   values appropriately so that they are unique (e.g., XMPP priority 1
>   to PIDF priority 0.007, XMPP priority 2 to PIDF priority 0.015, and
>   so on up through mapping XMPP priority 126 to PIDF priority 0.992;
>   note that this is an example only, and that the exact mapping shall
>   be determined by the XMPP-CPIM gateway).
>=20
> That's twice in one day I have looked at RFC 3922. ;-)
>=20

Heh :-) I agree, that text would clarify this.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Hi Markus,

On Jul 31, 2013, at 6:24 PM, Markus.Isomaki@nokia.com wrote:

> Hi Saul, Peter,
>=20
> Sa=FAl Ibarra Corretg=E9 wrote:
>>>=20
>>> Well, the sender might really really want to know if the message was
>>> delivered to the recipient. :-)
>>>=20
>>=20
>> Unfortunately there is no way to do that with a SIP MESSAGE. Unless =
we
>> invent a special payload for acknowledging reception of a given SIP
>> MESSAGE, but I think that is really out of the scope of what we are =
doing in
>> the IM document.
>>=20
>=20
> Please take a look at RFC 5438 =
(http://datatracker.ietf.org/doc/rfc5438/) and see if it applies to what =
would be required.
>=20

I had completely forgotten about this one, thanks for the pointer! =
Indeed this could be used to map receipts for SIP MESSAGE. Now, this is =
in a different spec to that which defines SIP MESSAGE, whereas MSRP =
reporting is baked into the protocol itself. Personally, I would define =
th empaling for MSRP, because REPORT is part of the protocol, and leave =
it out from IM, at least for the moment.

Peter added this to his slides, so I guess we'll discuss it in today's =
meeting.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On 8/1/13 10:01 AM, Saúl Ibarra Corretgé wrote:
> Hi Markus,
> 
> On Jul 31, 2013, at 6:24 PM, Markus.Isomaki@nokia.com wrote:
> 
>> Hi Saul, Peter,
>>
>> Saúl Ibarra Corretgé wrote:
>>>>
>>>> Well, the sender might really really want to know if the message was
>>>> delivered to the recipient. :-)
>>>>
>>>
>>> Unfortunately there is no way to do that with a SIP MESSAGE. Unless we
>>> invent a special payload for acknowledging reception of a given SIP
>>> MESSAGE, but I think that is really out of the scope of what we are doing in
>>> the IM document.
>>>
>>
>> Please take a look at RFC 5438 (http://datatracker.ietf.org/doc/rfc5438/) and see if it applies to what would be required.
>>
> 
> I had completely forgotten about this one, thanks for the pointer! Indeed this could be used to map receipts for SIP MESSAGE. Now, this is in a different spec to that which defines SIP MESSAGE, whereas MSRP reporting is baked into the protocol itself. Personally, I would define th empaling for MSRP, because REPORT is part of the protocol, and leave it out from IM, at least for the moment.
> 
> Peter added this to his slides, so I guess we'll discuss it in today's meeting.

I didn't add RFC 5438 to my slides, though, because I too had forgotten
about it.

One piece of data that would be helpful is to know how widely RFC 5438
is implemented, in comparison to RFC 4975. IMHO what we're trying to do
in this WG is to describe practical interworking between SIP and XMPP.
If a particular SIP or XMPP method is not widely implemented, I say that
we don't talk about it because it's purely theoretical (e.g., I removed
everything about XEP-0155). Thus some input on that point would be
helpful here.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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Hi,

Peter Saint-Andre wrote:
>=20
> I didn't add RFC 5438 to my slides, though, because I too had forgotten a=
bout
> it.
>=20
> One piece of data that would be helpful is to know how widely RFC 5438 is
> implemented, in comparison to RFC 4975. IMHO what we're trying to do in
> this WG is to describe practical interworking between SIP and XMPP.
> If a particular SIP or XMPP method is not widely implemented, I say that =
we
> don't talk about it because it's purely theoretical (e.g., I removed ever=
ything
> about XEP-0155). Thus some input on that point would be helpful here.
>=20

I agree with the principle of including only methods that are supported by =
real client implementations, especially related to specs that are a few yea=
rs old already.=20

I have not personally heard about RFC 5438 implementations. So unless someo=
ne has other info, I'm fine with excluding it from the stox-im spec.

Markus=20

From saul@ag-projects.com  Thu Aug  1 01:55:33 2013
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Subject: Re: [Stox] Review on -im
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On Aug 1, 2013, at 10:52 AM, <Markus.Isomaki@nokia.com> wrote:

> Hi,
>=20
> Peter Saint-Andre wrote:
>>=20
>> I didn't add RFC 5438 to my slides, though, because I too had =
forgotten about
>> it.
>>=20
>> One piece of data that would be helpful is to know how widely RFC =
5438 is
>> implemented, in comparison to RFC 4975. IMHO what we're trying to do =
in
>> this WG is to describe practical interworking between SIP and XMPP.
>> If a particular SIP or XMPP method is not widely implemented, I say =
that we
>> don't talk about it because it's purely theoretical (e.g., I removed =
everything
>> about XEP-0155). Thus some input on that point would be helpful here.
>>=20
>=20
> I agree with the principle of including only methods that are =
supported by real client implementations, especially related to specs =
that are a few years old already.=20
>=20
> I have not personally heard about RFC 5438 implementations. So unless =
someone has other info, I'm fine with excluding it from the stox-im =
spec.
>=20

FWIW, I haven't seen any myself either.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Thu Aug  1 01:57:55 2013
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Cc: stox@ietf.org, Markus.Isomaki@nokia.com
Subject: Re: [Stox] Review on -im
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On 8/1/13 10:55 AM, Saúl Ibarra Corretgé wrote:
> 
> On Aug 1, 2013, at 10:52 AM, <Markus.Isomaki@nokia.com> wrote:
> 
>> Hi,
>>
>> Peter Saint-Andre wrote:
>>>
>>> I didn't add RFC 5438 to my slides, though, because I too had forgotten about
>>> it.
>>>
>>> One piece of data that would be helpful is to know how widely RFC 5438 is
>>> implemented, in comparison to RFC 4975. IMHO what we're trying to do in
>>> this WG is to describe practical interworking between SIP and XMPP.
>>> If a particular SIP or XMPP method is not widely implemented, I say that we
>>> don't talk about it because it's purely theoretical (e.g., I removed everything
>>> about XEP-0155). Thus some input on that point would be helpful here.
>>>
>>
>> I agree with the principle of including only methods that are supported by real client implementations, especially related to specs that are a few years old already. 
>>
>> I have not personally heard about RFC 5438 implementations. So unless someone has other info, I'm fine with excluding it from the stox-im spec.
>>
> 
> FWIW, I haven't seen any myself either.

Thanks for the data points!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From christer.holmberg@ericsson.com  Fri Aug  2 03:14:01 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "stox@ietf.org" <stox@ietf.org>
Thread-Topic: Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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Subject: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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Hi,

As I indicated at the STOX session, the SIP MESSAGE examples in the draft c=
ontain a Contact header field, which is explicitly forbidden.

Section 4 of RFC 3248 says:

	"MESSAGE requests do not initiate dialogs.  User Agents MUST NOT=20
	insert Contact header fields into MESSAGE requests."

My question in the meeting was whether this is simply an editorial bug, or =
whether the Contact header field is actually used for something (there is n=
othing in the normative text), e.g. to ensure that
MESSAGE requests are routed to the correct user agents (someone mentioned t=
hat GRUU might be needed).

Regards,

Christer



From christer.holmberg@ericsson.com  Fri Aug  2 03:16:13 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "stox@ietf.org" <stox@ietf.org>
Thread-Topic: Media Sessions (draft-ietf-stox-media-01) and forking
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Subject: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,



As I indicated at the STOX session (and was later repeated by Jonathan Lenn=
ox), the draft need to have a story on SIP forking, e.g. if what happens if=
 the INVITE sent from the interworking node gets forked, and multiple early=
 dialogs are created.



Regards,



Christer


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<p class=3D"MsoPlainText"><span lang=3D"EN-US">Hi,<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black"><o:p>&=
nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black">As I i=
ndicated at the STOX session (and was later repeated by Jonathan Lennox), t=
he draft need to have a story on SIP forking, e.g. if what happens if the I=
NVITE sent from the interworking node
 gets forked, and multiple early dialogs are created.<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black"><o:p>&=
nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black">Regard=
s,<o:p></o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black"><o:p>&=
nbsp;</o:p></span></p>
<p class=3D"MsoPlainText"><span lang=3D"EN-US" style=3D"color:black">Christ=
er<o:p></o:p></span></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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From ag@ag-projects.com  Fri Aug  2 03:26:46 2013
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi Christer,

I would suggest that best practice is for the SIP traffic to be routed =
to the SIP Proxy/Registrar/Presence Agent of the given domain behind the =
gateway. That element has more proper capabilities for handling forking =
or any other SIP features.=20

Otherwise you will move all SIP functionality into the XMPP gateway and =
must document that whole SIP universe as running inside that gateway, =
which makes little sense.

Regards,=20
Adrian

On Aug 2, 2013, at 12:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> As I indicated at the STOX session (and was later repeated by Jonathan =
Lennox), the draft need to have a story on SIP forking, e.g. if what =
happens if the INVITE sent from the interworking node gets forked, and =
multiple early dialogs are created.
> =20
> Regards,
> =20
> Christer
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><base href=3D"x-msg://76/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; ">Hi =
Christer,<div><br></div><div>I would suggest that best practice is for =
the SIP traffic to be routed to the SIP Proxy/Registrar/Presence Agent =
of the given domain behind the gateway. That element has more proper =
capabilities for handling forking or any other SIP =
features.&nbsp;<div><br></div><div>Otherwise you will move all SIP =
functionality into the XMPP gateway and must document that whole SIP =
universe as running inside that gateway, which makes little =
sense.<div><br></div><div>Regards,&nbsp;</div><div>Adrian</div><div><br><d=
iv><div>On Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.=
com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
lang=3D"FI" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif; "><span =
lang=3D"EN-US">Hi,<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span =
lang=3D"EN-US" style=3D"">&nbsp;</span></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span =
lang=3D"EN-US" style=3D"">As I indicated at the STOX session (and was =
later repeated by Jonathan Lennox), the draft need to have a story on =
SIP forking, e.g. if what happens if the INVITE sent from the =
interworking node gets forked, and multiple early dialogs are =
created.<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" =
style=3D"">&nbsp;</span></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; "><span lang=3D"EN-US" =
style=3D"">Regards,<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span =
lang=3D"EN-US" style=3D"">&nbsp;</span></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span =
lang=3D"EN-US" style=3D"">Christer<o:p></o:p></span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif; =
"><o:p>&nbsp;</o:p></div></div>___________________________________________=
____<br>stox mailing list<br><a href=3D"mailto:stox@ietf.org" =
style=3D"color: purple; text-decoration: underline; =
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adrian Georgescu <ag@ag-projects.com>
Thread-Topic: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi Adrian,

>I would suggest that best practice is for the SIP traffic to be routed to =
the SIP Proxy/Registrar/Presence Agent of the given domain behind the gatew=
ay. That element has more proper capabilities for handling forking or any o=
ther SIP features.

Ok, but how are you going to ensure that such element exists?

Regards,

Christer





Otherwise you will move all SIP functionality into the XMPP gateway and mus=
t document that whole SIP universe as running inside that gateway, which ma=
kes little sense.

Regards,
Adrian

On Aug 2, 2013, at 12:16 PM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:


Hi,

As I indicated at the STOX session (and was later repeated by Jonathan Lenn=
ox), the draft need to have a story on SIP forking, e.g. if what happens if=
 the INVITE sent from the interworking node gets forked, and multiple early=
 dialogs are created.

Regards,

Christer

_______________________________________________
stox mailing list
stox@ietf.org<mailto:stox@ietf.org>
https://www.ietf.org/mailman/listinfo/stox


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</head>
<body lang=3D"FI" 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">Hi Adrian,<o:p></o:p></sp=
an></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">&gt;</s=
pan><span lang=3D"EN-US">I would suggest that best practice is for the SIP =
traffic to be routed to the SIP Proxy/Registrar/Presence Agent of the given=
 domain behind the gateway.
</span>That element has more proper capabilities for handling forking or an=
y other SIP features.&nbsp;<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok, but ho=
w are you going to ensure that such element exists?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal">Otherwise you will move all SIP functionality into t=
he XMPP gateway and must document that whole SIP universe as running inside=
 that gateway, which makes little sense.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Hi,</span><span style=3D=
"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"><=
o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">As I indicated at the ST=
OX session (and was later repeated by Jonathan Lennox), the draft need to h=
ave a story on SIP forking, e.g. if what happens if the INVITE
 sent from the interworking node gets forked, and multiple early dialogs ar=
e created.</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&=
quot;,&quot;sans-serif&quot;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Regards,</span><span sty=
le=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quo=
t;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><span style=
=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;=
"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Christer</span><span sty=
le=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quo=
t;"><o:p></o:p></span></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;<o:p></o:p></span></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org"><span style=3D"color:purple">stox@ietf.org=
</span></a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox"><span style=3D"color=
:purple">https://www.ietf.org/mailman/listinfo/stox</span></a><o:p></o:p></=
span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</div>
</body>
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From: Adrian Georgescu <ag@ag-projects.com>
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On Aug 2, 2013, at 12:30 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi Adrian,
> =20
> >I would suggest that best practice is for the SIP traffic to be =
routed to the SIP Proxy/Registrar/Presence Agent of the given domain =
behind the gateway. That element has more proper capabilities for =
handling forking or any other SIP features.=20
> =20
> Ok, but how are you going to ensure that such element exists?

The gateway knows, it queries the DNS for the domain SIP service =
records. If no proper records exist, the call flow will stop as there is =
nothing to rote the calls flows to.=20

Such gateway, its very existence, implies that there is both a SIP =
Server and and XMPP server for each domain. It is a cross protocol =
federation, the domain must exists otherwise there is nothing to gateway =
to and there is no reason to deploy it.=20

Adrian

> =20
> Regards,
> =20
> Christer
> =20
> =20
> =20
> =20
> =20
> Otherwise you will move all SIP functionality into the XMPP gateway =
and must document that whole SIP universe as running inside that =
gateway, which makes little sense.
> =20
> Regards,=20
> Adrian
> =20
> On Aug 2, 2013, at 12:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>=20
> Hi,
> =20
> As I indicated at the STOX session (and was later repeated by Jonathan =
Lennox), the draft need to have a story on SIP forking, e.g. if what =
happens if the INVITE sent from the interworking node gets forked, and =
multiple early dialogs are created.
> =20
> Regards,
> =20
> Christer
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


--Apple-Mail=_F55209F2-AABA-41D0-91D0-D260918B4EC7
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"><base href=3D"x-msg://76/"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Aug 2, 2013, =
at 12:30 PM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.=
com</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
lang=3D"FI" link=3D"blue" vlink=3D"purple" style=3D"font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><div =
class=3D"WordSection1" style=3D"page: WordSection1; "><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">Hi =
Adrian,<o:p></o:p></span></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"color: rgb(31, 73, 125); =
">&gt;</span><span lang=3D"EN-US">I would suggest that best practice is =
for the SIP traffic to be routed to the SIP Proxy/Registrar/Presence =
Agent of the given domain behind the gateway.<span =
class=3D"Apple-converted-space">&nbsp;</span></span>That element has =
more proper capabilities for handling forking or any other SIP =
features.&nbsp;<o:p></o:p></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Ok, but how =
are you going to ensure that such element =
exists?</span></div></div></div></div></div></blockquote><div><br></div><d=
iv>The gateway knows, it queries the DNS for the domain SIP service =
records. If no proper records exist, the call flow will stop as there is =
nothing to rote the calls flows to.&nbsp;</div><div><br></div><div>Such =
gateway, its very existence, implies that there is both a SIP Server and =
and XMPP server for each domain. It is a cross protocol federation, the =
domain must exists otherwise there is nothing to gateway to and there is =
no reason to deploy =
it.&nbsp;</div><div><br></div><div>Adrian</div><div><br></div><blockquote =
type=3D"cite"><div lang=3D"FI" link=3D"blue" vlink=3D"purple" =
style=3D"font-family: Helvetica; font-size: medium; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><div class=3D"WordSection1" style=3D"page: WordSection1; =
"><div><div><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p></o:p></span></div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Regards,<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Christer<o:p></o:p></span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span></div><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">&nbsp;</span></div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span></div></div><div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; ">Otherwise you =
will move all SIP functionality into the XMPP gateway and must document =
that whole SIP universe as running inside that gateway, which makes =
little sense.<o:p></o:p></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
">Regards,&nbsp;<o:p></o:p></div></div><div><div style=3D"margin: 0cm =
0cm 0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
">Adrian<o:p></o:p></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div><div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; ">On =
Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" style=3D"color: purple; =
text-decoration: underline; ">christer.holmberg@ericsson.com</a>&gt; =
wrote:<o:p></o:p></div></div><div style=3D"margin: 0cm 0cm 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
"><br><br><o:p></o:p></div><div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; ">Hi,</span><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; "><o:p></o:p></span></div></div><div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; ">&nbsp;</span><span style=3D"font-size:=
 11pt; font-family: Calibri, sans-serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; ">As I indicated at the STOX session (and was later repeated =
by Jonathan Lennox), the draft need to have a story on SIP forking, e.g. =
if what happens if the INVITE sent from the interworking node gets =
forked, and multiple early dialogs are created.</span><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; ">&nbsp;</span><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; "><o:p></o:p></span></div></div><div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; ">Regards,</span><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; ">&nbsp;</span><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; "><o:p></o:p></span></div></div><div><div =
style=3D"margin: 0cm 0cm 0.0001pt; font-size: 12pt; font-family: 'Times =
New Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; ">Christer</span><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
"><o:p></o:p></span></div></div><div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 11pt; font-family: Calibri, sans-serif; =
">&nbsp;<o:p></o:p></span></div></div><div style=3D"margin: 0cm 0cm =
0.0001pt; font-size: 12pt; font-family: 'Times New Roman', serif; =
"><span style=3D"font-size: 13.5pt; font-family: Helvetica, sans-serif; =
">_______________________________________________<br>stox mailing =
list<br><a href=3D"mailto:stox@ietf.org" style=3D"color: purple; =
text-decoration: underline; "><span style=3D"color: purple; =
">stox@ietf.org</span></a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/stox" style=3D"color: =
purple; text-decoration: underline; "><span style=3D"color: purple; =
">https://www.ietf.org/mailman/listinfo/stox</span></a><o:p></o:p></span><=
/div></div></div><div style=3D"margin: 0cm 0cm 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div></div></div></div>_________________________=
______________________<br>stox mailing list<br><a =
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ilman/listinfo/stox</div></blockquote></div><br></body></html>=

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From saul@ag-projects.com  Fri Aug  2 03:39:28 2013
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,

On Aug 2, 2013, at 12:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
> =20
> As I indicated at the STOX session (and was later repeated by Jonathan =
Lennox), the draft need to have a story on SIP forking, e.g. if what =
happens if the INVITE sent from the interworking node gets forked, and =
multiple early dialogs are created.

I'll propose text but in general terms, since forking doesn't exist in =
XMPP, the gateway will need to make sure only one reply is relayed to =
the XMPP side. Also, there dis very little text about early media in =
XEP-166, I'll look more into that.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From saul@ag-projects.com  Fri Aug  2 03:41:38 2013
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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Hi Christer,

On Aug 2, 2013, at 12:13 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> As I indicated at the STOX session, the SIP MESSAGE examples in the =
draft contain a Contact header field, which is explicitly forbidden.
>=20
> Section 4 of RFC 3248 says:
>=20
> 	"MESSAGE requests do not initiate dialogs.  User Agents MUST NOT=20=

> 	insert Contact header fields into MESSAGE requests."
>=20
> My question in the meeting was whether this is simply an editorial =
bug, or whether the Contact header field is actually used for something =
(there is nothing in the normative text), e.g. to ensure that
> MESSAGE requests are routed to the correct user agents (someone =
mentioned that GRUU might be needed).
>=20

I was the one who mentioned GRUU. We do need it if we want to know who =
sent the message exactly, but since the -im spec just deals with pager =
mode messages maybe we can live with just strangulating the bare JID. =
Peter?


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From christer.holmberg@ericsson.com  Fri Aug  2 03:50:38 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Thread-Topic: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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Hi,

If you want to signal who sent the message, I believe we can find another m=
echanism for that.=20

The GRUU is not really intended to indicate user identity anyway, and IF yo=
u want to use GRUU I believe you need to define how the gateway gets assign=
ed the GRUU.

Regards,

Christer

-----Alkuper=E4inen viesti-----
L=E4hett=E4j=E4: Sa=FAl Ibarra Corretg=E9 [mailto:saul@ag-projects.com]=20
L=E4hetetty: 2. elokuuta 2013 13:42
Vastaanottaja: Christer Holmberg
Kopio: stox@ietf.org
Aihe: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact =
header field

Hi Christer,

On Aug 2, 2013, at 12:13 PM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:

> Hi,
>=20
> As I indicated at the STOX session, the SIP MESSAGE examples in the draft=
 contain a Contact header field, which is explicitly forbidden.
>=20
> Section 4 of RFC 3248 says:
>=20
> 	"MESSAGE requests do not initiate dialogs.  User Agents MUST NOT=20
> 	insert Contact header fields into MESSAGE requests."
>=20
> My question in the meeting was whether this is simply an editorial=20
> bug, or whether the Contact header field is actually used for something (=
there is nothing in the normative text), e.g. to ensure that MESSAGE reques=
ts are routed to the correct user agents (someone mentioned that GRUU might=
 be needed).
>=20

I was the one who mentioned GRUU. We do need it if we want to know who sent=
 the message exactly, but since the -im spec just deals with pager mode mes=
sages maybe we can live with just strangulating the bare JID. Peter?


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From saul@ag-projects.com  Fri Aug  2 03:54:11 2013
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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On Aug 2, 2013, at 12:50 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> If you want to signal who sent the message, I believe we can find =
another mechanism for that.=20
>=20

I guess we are open to suggestions :-)

> The GRUU is not really intended to indicate user identity anyway, and =
IF you want to use GRUU I believe you need to define how the gateway =
gets assigned the GRUU.
>=20

-core defines how we use GRUU which is required, in a nutshell, a GRUU =
is translated to a full JID and vice versa. Now, IIRC GRUUs are used =
just in the Contact header (which we can't use here) and not in the =46rom=
 header, for instance.


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On Aug 2, 2013, at 12:50 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
> If you want to signal who sent the message, I believe we can find =
another mechanism for that.=20
>=20

One option would be to mandate the use of CPIM and set the =46rom header =
of the CPIM envelope to the GRUU.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From internet-drafts@ietf.org  Sat Aug  3 00:19:02 2013
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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Addresses and=
 Error Conditions
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-01.txt
	Pages           : 15
	Date            : 2013-08-03

Abstract:
   As a foundation for the definition of bidirectional protocol mappings
   between the Session Initiation Protocol (SIP) and the Extensible
   Messaging and Presence Protocol (XMPP), this document specifies the
   architectural assumptions underlying such mappings as well as the
   mapping of addresses and error conditions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-core

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From stpeter@stpeter.im  Sat Aug  3 00:54:09 2013
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On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
> On 7/30/13 5:33 PM, Saúl Ibarra Corretgé wrote:
>> 
>> - Using ID-123kdklejd doesn't seem to work as a valid xs:ID, TID-1234
>> does work though, so we could use TID- as the prefix for tuple
>> identifiers in examples and such.
> 
> I will double-check that against the XML specification.

Hmm. Here is what I see in the XML schema datatype specification:

   The ·value space· of ID is the set of all strings that ·match· the
   NCName production in [Namespaces in XML].

Where the NCName production is defined as follows:

   NCName 	::= 	(Letter | '_') (NCNameChar)*

What software are you using for validation? I'm not seeing a constraint
that an xs:ID needs to begin with three letters.

Peter

-- 
Peter Saint-Andre
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On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
> On 7/30/13 5:33 PM, Saúl Ibarra Corretgé wrote:
>> Hi,
>>
>> Here is my review of the -presence document:
>>
>> - Sec 3.3.2 suggests that if the subscription is maintained but we
>> have no presence state, a PIDF would be generated with a basic status
>> of closed, but what would be the tuple ID, if we don't know any
>> resource for this user anymore? 

Actually, in this scenario the gateway does know the status because the
XMPP user is still online -- it's just that the SIP user's subscription
to the XMPP user's status has expired.

>> We could also send an empty NOTIFY,
>> which would achieve the same goal.
> 
> The empty NOTIFY sounds more consistent with the spirit of SIP presence.

I've looked into this further. Here is what I find in RFC 3856 (Section
6.6.2):

   If the resource is not in a meaningful state, RFC 3265 [2] allows the
   body of the initial NOTIFY to be empty.  In the case of presence,
   that NOTIFY MAY contain a presence document.  This document would
   indicate whatever presence state the subscriber has been authorized
   to see; it is interpreted by the subscriber as the current presence
   state of the presentity.  For pending subscriptions, the state of the
   presentity SHOULD include some kind of textual note that indicates a
   pending status.

And in RFC 3265:

3.1.6.2. Confirmation of Subscription Creation/Refreshing

   Upon successfully accepting or refreshing a subscription, notifiers
   MUST send a NOTIFY message immediately to communicate the current
   resource state to the subscriber.  This NOTIFY message is sent on the
   same dialog as created by the SUBSCRIBE response.  If the resource
   has no meaningful state at the time that the SUBSCRIBE message is
   processed, this NOTIFY message MAY contain an empty or neutral body.

Everything there is about an initial NOTIFY confirming a presence
subscription. However, Section 3.3.2 of the stox-presence spec discusses
what to do if the SIP user's subscription expires:

   It is the responsibility of the SIMPLE-XMPP gateway to properly
   handle the difference between short-lived SIP presence subscriptions
   and long-lived XMPP presence subscriptions.  The gateway has two
   options when the SIP user's subscription expires...

It seems to me that there's a false assumption here, because in general
doesn't the SIP user agent initiate a refresh to maintain the
subscription (as long as it's online)? So I think there are two cases here:

(1) SIP user agent is online and generates a refresh by re-subscribing.
In this case, the empty NOTIFY might make sense (although you could
argue that it's not truly an "initial NOTIFY").

(2) SIP user agent goes offline and the subscription truly expires. In
this case, the handling we currently describe seems generally correct to me.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-01.txt
	Pages           : 21
	Date            : 2013-08-03

Abstract:
   This document defines a bi-directional protocol mapping for the
   exchange of presence information between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-presence

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-presence-01

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http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-presence-01


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From stpeter@stpeter.im  Sat Aug  3 02:29:31 2013
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Date: Sat, 03 Aug 2013 11:29:21 +0200
From: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-presence-01.txt
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This version addresses *most* of the feedback received. There's still an
open issue about empty SIP NOTIFY messages, but once Saúl replies in
that thread we can figure out what's correct.

Further feedback is of course welcome. :-)

On 8/3/13 11:26 AM, 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 SIP-TO-XMPP Working Group of the IETF.
> 
> 	Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Presence
> 	Author(s)       : Peter Saint-Andre
>                           Avshalom Houri
>                           Joe Hildebrand
> 	Filename        : draft-ietf-stox-presence-01.txt
> 	Pages           : 21
> 	Date            : 2013-08-03
> 
> Abstract:
>    This document defines a bi-directional protocol mapping for the
>    exchange of presence information between the Session Initiation
>    Protocol (SIP) and the Extensible Messaging and Presence Protocol
>    (XMPP).
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-presence
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-presence-01
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-presence-01
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> 


From christer.holmberg@ericsson.com  Sun Aug  4 09:55:24 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adrian Georgescu <ag@ag-projects.com>
Thread-Topic: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,

There for sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) i=
n the network, but how can you guarantee that it will handle forking? A for=
king proxy will simply forward 18x responses forming the early dialogs towa=
rds the gateway, which then will have to handle them.

Regards,

Christer

L=E4hett=E4j=E4: Adrian Georgescu [mailto:ag@ag-projects.com]
L=E4hetetty: 2. elokuuta 2013 13:36
Vastaanottaja: Christer Holmberg
Kopio: stox@ietf.org
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking


On Aug 2, 2013, at 12:30 PM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:


Hi Adrian,

>I would suggest that best practice is for the SIP traffic to be routed to =
the SIP Proxy/Registrar/Presence Agent of the given domain behind the gatew=
ay. That element has more proper capabilities for handling forking or any o=
ther SIP features.

Ok, but how are you going to ensure that such element exists?

The gateway knows, it queries the DNS for the domain SIP service records. I=
f no proper records exist, the call flow will stop as there is nothing to r=
ote the calls flows to.

Such gateway, its very existence, implies that there is both a SIP Server a=
nd and XMPP server for each domain. It is a cross protocol federation, the =
domain must exists otherwise there is nothing to gateway to and there is no=
 reason to deploy it.

Adrian


Regards,

Christer





Otherwise you will move all SIP functionality into the XMPP gateway and mus=
t document that whole SIP universe as running inside that gateway, which ma=
kes little sense.

Regards,
Adrian

On Aug 2, 2013, at 12:16 PM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:



Hi,

As I indicated at the STOX session (and was later repeated by Jonathan Lenn=
ox), the draft need to have a story on SIP forking, e.g. if what happens if=
 the INVITE sent from the interworking node gets forked, and multiple early=
 dialogs are created.

Regards,

Christer

_______________________________________________
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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,<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 lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">There for =
sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) in the netw=
ork, but how can you guarantee that it will handle forking?
 A forking proxy will simply forward 18x responses forming the early dialog=
s towards the gateway, which then will have to handle them.<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> Adrian Georgescu [mailto:ag@ag-projects.com]
<br>
<b>L=E4hetetty:</b> 2. elokuuta 2013 13:36<br>
<b>Vastaanottaja:</b> Christer Holmberg<br>
<b>Kopio:</b> stox@ietf.org<br>
<b>Aihe:</b> Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forki=
ng<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Aug 2, 2013, at 12:30 PM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt; wrote:<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<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 Adrian,</span><o:p></o=
:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">&gt;</s=
pan><span lang=3D"EN-US">I would suggest that best practice is for the SIP =
traffic to be routed to the SIP Proxy/Registrar/Presence Agent of the given=
 domain behind the gateway.<span class=3D"apple-converted-space">&nbsp;</sp=
an></span>That
 element has more proper capabilities for handling forking or any other SIP=
 features.&nbsp;<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok, but ho=
w are you going to ensure that such element exists?</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">The gateway knows, it queries the DNS for the domain=
 SIP service records. If no proper records exist, the call flow will stop a=
s there is nothing to rote the calls flows to.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Such gateway, its very existence, implies that there=
 is both a SIP Server and and XMPP server for each domain. It is a cross pr=
otocol federation, the domain must exists otherwise there is nothing to gat=
eway to and there is no reason to
 deploy it.&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</=
span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</=
span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Otherwise you will move all SIP functionality into t=
he XMPP gateway and must document that whole SIP universe as running inside=
 that gateway, which makes little sense.<o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Regards,&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com"><span style=3D"color:purpl=
e">christer.holmberg@ericsson.com</span></a>&gt; wrote:<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Hi,</span><o:p></o:p></p=
>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">As I indicated at the ST=
OX session (and was later repeated by Jonathan Lennox), the draft need to h=
ave a story on SIP forking, e.g. if what happens if the INVITE
 sent from the interworking node gets forked, and multiple early dialogs ar=
e created.</span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Regards,</span><o:p></o:=
p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Christer</span><o:p></o:=
p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org"><span style=3D"color:purple">stox@ietf.org=
</span></a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox"><span style=3D"color=
:purple">https://www.ietf.org/mailman/listinfo/stox</span></a></span><o:p><=
/o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox">https://www.ietf.org=
/mailman/listinfo/stox</a><o:p></o:p></span></p>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

--_000_7594FB04B1934943A5C02806D1A2204B1C41C17EESESSMB209erics_--

From christer.holmberg@ericsson.com  Sun Aug  4 09:58:05 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: =?iso-8859-1?Q?Sa=FAl_Ibarra_Corretg=E9?= <saul@ag-projects.com>
Thread-Topic: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,
=20
>> As I indicated at the STOX session (and was later repeated by Jonathan L=
ennox), the draft need to have a story on SIP forking, e.g. if what happens=
 if the INVITE sent from the interworking node gets forked, and multiple ea=
rly dialogs are created.
>
> I'll propose text but in general terms, since forking doesn't exist in XM=
PP, the gateway will need to make sure only one reply is relayed to the XMP=
P side. Also, there dis very little text about early media in XEP-166, I'll=
 look more into that.

Unfortunately my XMPP knowledge is more or less zero, so at this point I ca=
n't really suggest any solution. However, if early media does not exist in =
XMPP, and the gateway only needs to care about (at least from a media persp=
ective) the 200 OK, there probably is no problem :)

Regards,

Christer





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From: Adrian Georgescu <ag@ag-projects.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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The gateway lookups the SIP server for a given domain in the DNS same as =
any other device would do like a SIP phone configured with credentials =
under that domain. Is the job of that server to properly implement all =
SIP functions like forking proxy, handle nat traversal or registrar to =
name a few common features.

You ask what shall the world do if the SIP server of given domain does =
not implemented forking or other feature. My answer is the gateway must =
not care, is the problem of whomever manages that domain and its =
applications to handle such features.

Regards,
Adrian



On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:

> Hi,
> =20
> There for sure may be a SIP server (a proxy, B2BUA and/or the remote =
UAS) in the network, but how can you guarantee that it will handle =
forking? A forking proxy will simply forward 18x responses forming the =
early dialogs towards the gateway, which then will have to handle them.
> =20
> Regards,
> =20
> Christer
> =20
> L=E4hett=E4j=E4: Adrian Georgescu [mailto:ag@ag-projects.com]=20
> L=E4hetetty: 2. elokuuta 2013 13:36
> Vastaanottaja: Christer Holmberg
> Kopio: stox@ietf.org
> Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
> =20
> =20
> On Aug 2, 2013, at 12:30 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>=20
> Hi Adrian,
> =20
> >I would suggest that best practice is for the SIP traffic to be =
routed to the SIP Proxy/Registrar/Presence Agent of the given domain =
behind the gateway. That element has more proper capabilities for =
handling forking or any other SIP features.=20
> =20
> Ok, but how are you going to ensure that such element exists?
> =20
> The gateway knows, it queries the DNS for the domain SIP service =
records. If no proper records exist, the call flow will stop as there is =
nothing to rote the calls flows to.=20
> =20
> Such gateway, its very existence, implies that there is both a SIP =
Server and and XMPP server for each domain. It is a cross protocol =
federation, the domain must exists otherwise there is nothing to gateway =
to and there is no reason to deploy it.=20
> =20
> Adrian
> =20
> =20
> Regards,
> =20
> Christer
> =20
> =20
> =20
> =20
> =20
> Otherwise you will move all SIP functionality into the XMPP gateway =
and must document that whole SIP universe as running inside that =
gateway, which makes little sense.
> =20
> Regards,=20
> Adrian
> =20
> On Aug 2, 2013, at 12:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>=20
>=20
>=20
> Hi,
> =20
> As I indicated at the STOX session (and was later repeated by Jonathan =
Lennox), the draft need to have a story on SIP forking, e.g. if what =
happens if the INVITE sent from the interworking node gets forked, and =
multiple early dialogs are created.
> =20
> Regards,
> =20
> Christer
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
> =20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox


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<html><head><base href=3D"x-msg://76/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; ">The gateway lookups the SIP server for a given =
domain in the DNS same as any other device would do like a SIP phone =
configured with credentials under that domain. Is the job of that server =
to properly implement all SIP functions like forking proxy, handle nat =
traversal or registrar to name a few common =
features.<div><br></div><div>You ask what shall the world do if the SIP =
server of given domain does not implemented forking or other feature. My =
answer is the gateway must not care, is the problem of whomever manages =
that domain and its applications to handle such =
features.</div><div><br></div><div>Regards,</div><div>Adrian</div><div><br=
><div><br><div><br><div><div>On Aug 4, 2013, at 6:55 PM, Christer =
Holmberg wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"FI" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Hi,<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">There for sure may be a SIP server (a proxy, B2BUA =
and/or the remote UAS) in the network, but how can you guarantee that it =
will handle forking? A forking proxy will simply forward 18x responses =
forming the early dialogs towards the gateway, which then will have to =
handle them.<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Regards,<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Christer<o:p></o:p></span></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div><div =
style=3D"border-right-style: none; border-bottom-style: none; =
border-left-style: none; border-width: initial; border-color: initial; =
border-top-style: solid; border-top-color: rgb(181, 196, 223); =
border-top-width: 1pt; padding-top: 3pt; padding-right: 0cm; =
padding-bottom: 0cm; padding-left: 0cm; "><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><b><span =
style=3D"font-size: 10pt; font-family: Tahoma, sans-serif; =
">L=E4hett=E4j=E4:</span></b><span style=3D"font-size: 10pt; =
font-family: Tahoma, sans-serif; "><span =
class=3D"Apple-converted-space">&nbsp;</span>Adrian Georgescu =
[mailto:ag@ag-projects.com]<span =
class=3D"Apple-converted-space">&nbsp;</span><br><b>L=E4hetetty:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>2. elokuuta 2013 =
13:36<br><b>Vastaanottaja:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Christer =
Holmberg<br><b>Kopio:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:stox@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">stox@ietf.org</a><br><b>Aihe:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Re: [Stox] Media Sessions =
(draft-ietf-stox-media-01) and =
forking<o:p></o:p></span></div></div></div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div><div style=3D"margin-top: 0cm; margin-right: =
0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div><div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">On Aug 2, 2013, at 12:30 =
PM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" style=3D"color: blue; =
text-decoration: underline; ">christer.holmberg@ericsson.com</a>&gt; =
wrote:<o:p></o:p></div></div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><br><br><o:p></o:p></div><div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Hi =
Adrian,</span><o:p></o:p></div></div><div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div></div><div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"color: rgb(31, 73, 125); ">&gt;</span><span =
lang=3D"EN-US">I would suggest that best practice is for the SIP traffic =
to be routed to the SIP Proxy/Registrar/Presence Agent of the given =
domain behind the gateway.<span =
class=3D"apple-converted-space">&nbsp;</span></span>That element has =
more proper capabilities for handling forking or any other SIP =
features.&nbsp;<o:p></o:p></div></div><div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Ok, but how are you going to =
ensure that such element =
exists?</span><o:p></o:p></div></div></div></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">The gateway knows, it queries the DNS for the domain =
SIP service records. If no proper records exist, the call flow will stop =
as there is nothing to rote the calls flows =
to.&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Such gateway, its very =
existence, implies that there is both a SIP Server and and XMPP server =
for each domain. It is a cross protocol federation, the domain must =
exists otherwise there is nothing to gateway to and there is no reason =
to deploy it.&nbsp;<o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">Adrian<o:p></o:p></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><blockquote style=3D"margin-top: 5pt; =
margin-bottom: 5pt; "><div><div><div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span lang=3D"EN-US" =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">&nbsp;</span><o:p></o:p></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Regards,</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">Christer</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
lang=3D"EN-US" style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
">&nbsp;</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">Otherwise you will move all SIP functionality into the =
XMPP gateway and must document that whole SIP universe as running inside =
that gateway, which makes little =
sense.<o:p></o:p></div></div><div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div></div><div><div><div style=3D"margin-top: =
0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
">Regards,&nbsp;<o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">Adrian<o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">&nbsp;<o:p></o:p></div></div><div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; ">On Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com" style=3D"color: blue; =
text-decoration: underline; "><span style=3D"color: purple; =
">christer.holmberg@ericsson.com</span></a>&gt; =
wrote:<o:p></o:p></div></div></div><div><div style=3D"margin-top: 0cm; =
margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><br><br><br><o:p></o:p></div></div><div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">Hi,</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">&nbsp;</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; ">As I indicated at the STOX session =
(and was later repeated by Jonathan Lennox), the draft need to have a =
story on SIP forking, e.g. if what happens if the INVITE sent from the =
interworking node gets forked, and multiple early dialogs are =
created.</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">&nbsp;</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">Regards,</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">&nbsp;</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span lang=3D"EN-US" style=3D"font-size: 11pt; =
font-family: Calibri, sans-serif; =
">Christer</span><o:p></o:p></div></div></div><div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; ">&nbsp;</span><o:p></o:p></div></div></div><div><div =
style=3D"margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 13.5pt; font-family: =
Helvetica, sans-serif; =
">_______________________________________________<br>stox mailing =
list<br><a href=3D"mailto:stox@ietf.org" style=3D"color: blue; =
text-decoration: underline; "><span style=3D"color: purple; =
">stox@ietf.org</span></a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/stox" style=3D"color: =
blue; text-decoration: underline; "><span style=3D"color: purple; =
">https://www.ietf.org/mailman/listinfo/stox</span></a></span><o:p></o:p><=
/div></div></div></div><div><div style=3D"margin-top: 0cm; margin-right: =
0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 12pt; =
font-family: 'Times New Roman', serif; =
">&nbsp;<o:p></o:p></div></div></div></div></div><div style=3D"margin-top:=
 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 13.5pt; font-family: Helvetica, sans-serif; =
">_______________________________________________<br>stox mailing =
list<br><a href=3D"mailto:stox@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">stox@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/stox" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/stox</a><o:p></o:p></span></div></=
div></blockquote></div><div style=3D"margin-top: 0cm; margin-right: 0cm; =
margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 12pt; font-family: =
'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div>___________________________________________=
____<br>stox mailing list<br><a href=3D"mailto:stox@ietf.org" =
style=3D"color: blue; text-decoration: underline; =
">stox@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/stox" style=3D"color: =
blue; text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/stox</a><br></div></span></blockqu=
ote></div><br></div></div></div></body></html>=

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From christer.holmberg@ericsson.com  Sun Aug  4 11:40:04 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adrian Georgescu <ag@ag-projects.com>
Thread-Topic: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
Thread-Index: Ac6PaRwphn/mcGjdScuZ2ITH5v377///4akA///dspCAACTvAP/8UCBAgAdFoQD//8jcoA==
Date: Sun, 4 Aug 2013 18:39:57 +0000
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,

I am not sure I understand.

The server may implement SIP functions like forking, and fork the INVITE. B=
UT, the gateway will still get the 18x responses for the early dialogs - ju=
st like a SIP phone.

Regards,

Christer

L=E4hett=E4j=E4: Adrian Georgescu [mailto:ag@ag-projects.com]
L=E4hetetty: 4. elokuuta 2013 20:20
Vastaanottaja: Christer Holmberg
Kopio: stox@ietf.org
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking

The gateway lookups the SIP server for a given domain in the DNS same as an=
y other device would do like a SIP phone configured with credentials under =
that domain. Is the job of that server to properly implement all SIP functi=
ons like forking proxy, handle nat traversal or registrar to name a few com=
mon features.

You ask what shall the world do if the SIP server of given domain does not =
implemented forking or other feature. My answer is the gateway must not car=
e, is the problem of whomever manages that domain and its applications to h=
andle such features.

Regards,
Adrian



On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:


Hi,

There for sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) i=
n the network, but how can you guarantee that it will handle forking? A for=
king proxy will simply forward 18x responses forming the early dialogs towa=
rds the gateway, which then will have to handle them.

Regards,

Christer

L=E4hett=E4j=E4: Adrian Georgescu [mailto:ag@ag-projects.com]
L=E4hetetty: 2. elokuuta 2013 13:36
Vastaanottaja: Christer Holmberg
Kopio: stox@ietf.org<mailto:stox@ietf.org>
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking


On Aug 2, 2013, at 12:30 PM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:



Hi Adrian,

>I would suggest that best practice is for the SIP traffic to be routed to =
the SIP Proxy/Registrar/Presence Agent of the given domain behind the gatew=
ay. That element has more proper capabilities for handling forking or any o=
ther SIP features.

Ok, but how are you going to ensure that such element exists?

The gateway knows, it queries the DNS for the domain SIP service records. I=
f no proper records exist, the call flow will stop as there is nothing to r=
ote the calls flows to.

Such gateway, its very existence, implies that there is both a SIP Server a=
nd and XMPP server for each domain. It is a cross protocol federation, the =
domain must exists otherwise there is nothing to gateway to and there is no=
 reason to deploy it.

Adrian


Regards,

Christer





Otherwise you will move all SIP functionality into the XMPP gateway and mus=
t document that whole SIP universe as running inside that gateway, which ma=
kes little sense.

Regards,
Adrian

On Aug 2, 2013, at 12:16 PM, Christer Holmberg <christer.holmberg@ericsson.=
com<mailto:christer.holmberg@ericsson.com>> wrote:




Hi,

As I indicated at the STOX session (and was later repeated by Jonathan Lenn=
ox), the draft need to have a story on SIP forking, e.g. if what happens if=
 the INVITE sent from the interworking node gets forked, and multiple early=
 dialogs are created.

Regards,

Christer

_______________________________________________
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https://www.ietf.org/mailman/listinfo/stox

_______________________________________________
stox mailing list
stox@ietf.org<mailto:stox@ietf.org>
https://www.ietf.org/mailman/listinfo/stox

_______________________________________________
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stox@ietf.org<mailto:stox@ietf.org>
https://www.ietf.org/mailman/listinfo/stox


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</head>
<body lang=3D"FI" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,<o:p></=
o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I am not s=
ure I understand.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The server=
 may implement SIP functions like forking, and fork the INVITE. BUT, the ga=
teway will still get the 18x responses for the early dialogs
 &#8211; just like a SIP phone.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer<o=
:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span styl=
e=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;=
"> Adrian Georgescu [mailto:ag@ag-projects.com]
<br>
<b>L=E4hetetty:</b> 4. elokuuta 2013 20:20<br>
<b>Vastaanottaja:</b> Christer Holmberg<br>
<b>Kopio:</b> stox@ietf.org<br>
<b>Aihe:</b> Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forki=
ng<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The gateway lookups the SIP server for a given domai=
n in the DNS same as any other device would do like a SIP phone configured =
with credentials under that domain. Is the job of that server to properly i=
mplement all SIP functions like forking
 proxy, handle nat traversal or registrar to name a few common features.<o:=
p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">You ask what shall the world do if the SIP server of=
 given domain does not implemented forking or other feature. My answer is t=
he gateway must not care, is the problem of whomever manages that domain an=
d its applications to handle such
 features.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<div>
<p class=3D"MsoNormal">On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:=
<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><br>
<br>
<o:p></o:p></p>
<div>
<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,</span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p><=
/p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">There for =
sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) in the netw=
ork, but how can you guarantee that it will handle forking?
 A forking proxy will simply forward 18x responses forming the early dialog=
s towards the gateway, which then will have to handle them.</span><o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</=
span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</=
span><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm;border-width:initial;border-color:initial">
<div>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">L=E4hett=E4j=E4:</span></b><span clas=
s=3D"apple-converted-space"><span style=3D"font-size:10.0pt;font-family:&qu=
ot;Tahoma&quot;,&quot;sans-serif&quot;">&nbsp;</span></span><span style=3D"=
font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Adr=
ian
 Georgescu [<a href=3D"mailto:ag@ag-projects.com">mailto:ag@ag-projects.com=
</a>]<span class=3D"apple-converted-space">&nbsp;</span><br>
<b>L=E4hetetty:</b><span class=3D"apple-converted-space">&nbsp;</span>2. el=
okuuta 2013 13:36<br>
<b>Vastaanottaja:</b><span class=3D"apple-converted-space">&nbsp;</span>Chr=
ister Holmberg<br>
<b>Kopio:</b><span class=3D"apple-converted-space">&nbsp;</span><a href=3D"=
mailto:stox@ietf.org">stox@ietf.org</a><br>
<b>Aihe:</b><span class=3D"apple-converted-space">&nbsp;</span>Re: [Stox] M=
edia Sessions (draft-ietf-stox-media-01) and forking</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Aug 2, 2013, at 12:30 PM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt; wrote:<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<o:p></o:p></p>
</div>
<div>
<div>
<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 Adrian,</span><o:p></o=
:p></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"color:#1F497D">&gt;</s=
pan><span lang=3D"EN-US">I would suggest that best practice is for the SIP =
traffic to be routed to the SIP Proxy/Registrar/Presence Agent of the given=
 domain behind the gateway.<span class=3D"apple-converted-space">&nbsp;</sp=
an></span>That
 element has more proper capabilities for handling forking or any other SIP=
 features.&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Ok, but ho=
w are you going to ensure that such element exists?</span><o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">The gateway knows, it queries the DNS for the domain=
 SIP service records. If no proper records exist, the call flow will stop a=
s there is nothing to rote the calls flows to.&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Such gateway, its very existence, implies that there=
 is both a SIP Server and and XMPP server for each domain. It is a cross pr=
otocol federation, the domain must exists otherwise there is nothing to gat=
eway to and there is no reason to
 deploy it.&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Regards,</=
span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Christer</=
span><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</sp=
an><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Otherwise you will move all SIP functionality into t=
he XMPP gateway and must document that whole SIP universe as running inside=
 that gateway, which makes little sense.<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Regards,&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">Adrian<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal">On Aug 2, 2013, at 12:16 PM, Christer Holmberg &lt;<=
a href=3D"mailto:christer.holmberg@ericsson.com"><span style=3D"color:purpl=
e">christer.holmberg@ericsson.com</span></a>&gt; wrote:<o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><br>
<br>
<br>
<br>
<o:p></o:p></p>
</div>
</div>
<div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Hi,</span><o:p></o:p></p=
>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">As I indicated at the ST=
OX session (and was later repeated by Jonathan Lennox), the draft need to h=
ave a story on SIP forking, e.g. if what happens if the INVITE
 sent from the interworking node gets forked, and multiple early dialogs ar=
e created.</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Regards,</span><o:p></o:=
p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p>=
</p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:11.0pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;">Christer</span><o:p></o:=
p></p>
</div>
</div>
</div>
<div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org"><span style=3D"color:purple">stox@ietf.org=
</span></a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox"><span style=3D"color=
:purple">https://www.ietf.org/mailman/listinfo/stox</span></a></span><o:p><=
/o:p></p>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
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</blockquote>
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<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><span style=3D"font-size:13.5pt;font-family:&quot;He=
lvetica&quot;,&quot;sans-serif&quot;">_____________________________________=
__________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox">https://www.ietf.org=
/mailman/listinfo/stox</a><o:p></o:p></span></p>
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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The gateway would care to translate the final response. Getting more 
then one 180 is not causing any harm nor does those need to be 
translated into different events on XMPP side.

Adrian

On 08/04/2013 08:39 PM, Christer Holmberg wrote:
>
> Hi,
>
> I am not sure I understand.
>
> The server may implement SIP functions like forking, and fork the 
> INVITE. BUT, the gateway will still get the 18x responses for the 
> early dialogs -- just like a SIP phone.
>
> Regards,
>
> Christer
>
> *Lähettäjä:*Adrian Georgescu [mailto:ag@ag-projects.com]
> *Lähetetty:* 4. elokuuta 2013 20:20
> *Vastaanottaja:* Christer Holmberg
> *Kopio:* stox@ietf.org
> *Aihe:* Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
>
> The gateway lookups the SIP server for a given domain in the DNS same 
> as any other device would do like a SIP phone configured with 
> credentials under that domain. Is the job of that server to properly 
> implement all SIP functions like forking proxy, handle nat traversal 
> or registrar to name a few common features.
>
> You ask what shall the world do if the SIP server of given domain does 
> not implemented forking or other feature. My answer is the gateway 
> must not care, is the problem of whomever manages that domain and its 
> applications to handle such features.
>
> Regards,
>
> Adrian
>
> On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:
>
>
>
> Hi,
>
> There for sure may be a SIP server (a proxy, B2BUA and/or the remote 
> UAS) in the network, but how can you guarantee that it will handle 
> forking? A forking proxy will simply forward 18x responses forming the 
> early dialogs towards the gateway, which then will have to handle them.
>
> Regards,
>
> Christer
>
> *Lähettäjä:*Adrian Georgescu [mailto:ag@ag-projects.com]
> *Lähetetty:*2. elokuuta 2013 13:36
> *Vastaanottaja:*Christer Holmberg
> *Kopio:*stox@ietf.org <mailto:stox@ietf.org>
> *Aihe:*Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
>
> On Aug 2, 2013, at 12:30 PM, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>
>
>
> Hi Adrian,
>
> >I would suggest that best practice is for the SIP traffic to be routed 
> to the SIP Proxy/Registrar/Presence Agent of the given domain behind 
> the gateway.That element has more proper capabilities for handling 
> forking or any other SIP features.
>
> Ok, but how are you going to ensure that such element exists?
>
> The gateway knows, it queries the DNS for the domain SIP service 
> records. If no proper records exist, the call flow will stop as there 
> is nothing to rote the calls flows to.
>
> Such gateway, its very existence, implies that there is both a SIP 
> Server and and XMPP server for each domain. It is a cross protocol 
> federation, the domain must exists otherwise there is nothing to 
> gateway to and there is no reason to deploy it.
>
> Adrian
>
>     Regards,
>
>     Christer
>
>     Otherwise you will move all SIP functionality into the XMPP
>     gateway and must document that whole SIP universe as running
>     inside that gateway, which makes little sense.
>
>     Regards,
>
>     Adrian
>
>     On Aug 2, 2013, at 12:16 PM, Christer Holmberg
>     <christer.holmberg@ericsson.com
>     <mailto:christer.holmberg@ericsson.com>> wrote:
>
>
>
>
>
>     Hi,
>
>     As I indicated at the STOX session (and was later repeated by
>     Jonathan Lennox), the draft need to have a story on SIP forking,
>     e.g. if what happens if the INVITE sent from the interworking node
>     gets forked, and multiple early dialogs are created.
>
>     Regards,
>
>     Christer
>
>     _______________________________________________
>     stox mailing list
>     stox@ietf.org <mailto:stox@ietf.org>
>     https://www.ietf.org/mailman/listinfo/stox
>
>     _______________________________________________
>     stox mailing list
>     stox@ietf.org <mailto:stox@ietf.org>
>     https://www.ietf.org/mailman/listinfo/stox
>
> _______________________________________________
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> https://www.ietf.org/mailman/listinfo/stox
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    <div class="moz-cite-prefix">The gateway would care to translate the
      final response. Getting more then one 180 is not causing any harm
      nor does those need to be translated into different events on XMPP
      side.<br>
      <br>
      Adrian <br>
      <br>
      On 08/04/2013 08:39 PM, Christer Holmberg wrote:<br>
    </div>
    <blockquote
cite="mid:7594FB04B1934943A5C02806D1A2204B1C41C208@ESESSMB209.ericsson.se"
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">I am not sure I understand.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">The server may implement SIP functions like
            forking, and fork the INVITE. BUT, the gateway will still
            get the 18x responses for the early dialogs &#8211; just like a
            SIP phone.<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-US"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">L&auml;hett&auml;j&auml;:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                Adrian Georgescu [<a class="moz-txt-link-freetext" href="mailto:ag@ag-projects.com">mailto:ag@ag-projects.com</a>]
                <br>
                <b>L&auml;hetetty:</b> 4. elokuuta 2013 20:20<br>
                <b>Vastaanottaja:</b> Christer Holmberg<br>
                <b>Kopio:</b> <a class="moz-txt-link-abbreviated" href="mailto:stox@ietf.org">stox@ietf.org</a><br>
                <b>Aihe:</b> Re: [Stox] Media Sessions
                (draft-ietf-stox-media-01) and forking<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">The gateway lookups the SIP server for a
          given domain in the DNS same as any other device would do like
          a SIP phone configured with credentials under that domain. Is
          the job of that server to properly implement all SIP functions
          like forking proxy, handle nat traversal or registrar to name
          a few common features.<o:p></o:p></p>
        <div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        </div>
        <div>
          <p class="MsoNormal">You ask what shall the world do if the
            SIP server of given domain does not implemented forking or
            other feature. My answer is the gateway must not care, is
            the problem of whomever manages that domain and its
            applications to handle such features.<o:p></o:p></p>
        </div>
        <div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        </div>
        <div>
          <p class="MsoNormal">Regards,<o:p></o:p></p>
        </div>
        <div>
          <p class="MsoNormal">Adrian<o:p></o:p></p>
        </div>
        <div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <div>
            <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
            <div>
              <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
              <div>
                <div>
                  <p class="MsoNormal">On Aug 4, 2013, at 6:55 PM,
                    Christer Holmberg wrote:<o:p></o:p></p>
                </div>
                <p class="MsoNormal"><br>
                  <br>
                  <o:p></o:p></p>
                <div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi,</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">There for sure may be a SIP server
                        (a proxy, B2BUA and/or the remote UAS) in the
                        network, but how can you guarantee that it will
                        handle forking? A forking proxy will simply
                        forward 18x responses forming the early dialogs
                        towards the gateway, which then will have to
                        handle them.</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">Regards,</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">Christer</span><o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                        lang="EN-US">&nbsp;</span><o:p></o:p></p>
                  </div>
                  <div>
                    <div style="border:none;border-top:solid #B5C4DF
                      1.0pt;padding:3.0pt 0cm 0cm
                      0cm;border-width:initial;border-color:initial">
                      <div>
                        <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">L&auml;hett&auml;j&auml;:</span></b><span
                            class="apple-converted-space"><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">&nbsp;</span></span><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">Adrian

                            Georgescu [<a moz-do-not-send="true"
                              href="mailto:ag@ag-projects.com">mailto:ag@ag-projects.com</a>]<span
                              class="apple-converted-space">&nbsp;</span><br>
                            <b>L&auml;hetetty:</b><span
                              class="apple-converted-space">&nbsp;</span>2.
                            elokuuta 2013 13:36<br>
                            <b>Vastaanottaja:</b><span
                              class="apple-converted-space">&nbsp;</span>Christer
                            Holmberg<br>
                            <b>Kopio:</b><span
                              class="apple-converted-space">&nbsp;</span><a
                              moz-do-not-send="true"
                              href="mailto:stox@ietf.org">stox@ietf.org</a><br>
                            <b>Aihe:</b><span
                              class="apple-converted-space">&nbsp;</span>Re:
                            [Stox] Media Sessions
                            (draft-ietf-stox-media-01) and forking</span><o:p></o:p></p>
                      </div>
                    </div>
                  </div>
                  <div>
                    <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                  </div>
                  <div>
                    <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                  </div>
                  <div>
                    <div>
                      <div>
                        <p class="MsoNormal">On Aug 2, 2013, at 12:30
                          PM, Christer Holmberg &lt;<a
                            moz-do-not-send="true"
                            href="mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt;
                          wrote:<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <p class="MsoNormal"><br>
                        <br>
                        <br>
                        <o:p></o:p></p>
                    </div>
                    <div>
                      <div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi
                              Adrian,</span><o:p></o:p></p>
                        </div>
                      </div>
                      <div>
                        <div>
                          <div>
                            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                          </div>
                        </div>
                      </div>
                      <div>
                        <div>
                          <div>
                            <p class="MsoNormal"><span
                                style="color:#1F497D" lang="EN-US">&gt;</span><span
                                lang="EN-US">I would suggest that best
                                practice is for the SIP traffic to be
                                routed to the SIP
                                Proxy/Registrar/Presence Agent of the
                                given domain behind the gateway.<span
                                  class="apple-converted-space">&nbsp;</span></span>That

                              element has more proper capabilities for
                              handling forking or any other SIP
                              features.&nbsp;<o:p></o:p></p>
                          </div>
                        </div>
                        <div>
                          <div>
                            <div>
                              <p class="MsoNormal"><span
                                  style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
                            </div>
                          </div>
                          <div>
                            <div>
                              <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                  lang="EN-US">Ok, but how are you going
                                  to ensure that such element exists?</span><o:p></o:p></p>
                            </div>
                          </div>
                        </div>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">The gateway knows, it
                          queries the DNS for the domain SIP service
                          records. If no proper records exist, the call
                          flow will stop as there is nothing to rote the
                          calls flows to.&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">Such gateway, its very
                          existence, implies that there is both a SIP
                          Server and and XMPP server for each domain. It
                          is a cross protocol federation, the domain
                          must exists otherwise there is nothing to
                          gateway to and there is no reason to deploy
                          it.&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">Adrian<o:p></o:p></p>
                      </div>
                    </div>
                    <div>
                      <div>
                        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                      </div>
                    </div>
                    <blockquote
                      style="margin-top:5.0pt;margin-bottom:5.0pt">
                      <div>
                        <div>
                          <div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">Regards,</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">Christer</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
                                    lang="EN-US">&nbsp;</span><o:p></o:p></p>
                              </div>
                            </div>
                          </div>
                          <div>
                            <div>
                              <div>
                                <p class="MsoNormal">Otherwise you will
                                  move all SIP functionality into the
                                  XMPP gateway and must document that
                                  whole SIP universe as running inside
                                  that gateway, which makes little
                                  sense.<o:p></o:p></p>
                              </div>
                            </div>
                            <div>
                              <div>
                                <div>
                                  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                                </div>
                              </div>
                            </div>
                            <div>
                              <div>
                                <div>
                                  <p class="MsoNormal">Regards,&nbsp;<o:p></o:p></p>
                                </div>
                              </div>
                            </div>
                            <div>
                              <div>
                                <div>
                                  <p class="MsoNormal">Adrian<o:p></o:p></p>
                                </div>
                              </div>
                            </div>
                            <div>
                              <div>
                                <div>
                                  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                                </div>
                              </div>
                              <div>
                                <div>
                                  <div>
                                    <div>
                                      <p class="MsoNormal">On Aug 2,
                                        2013, at 12:16 PM, Christer
                                        Holmberg &lt;<a
                                          moz-do-not-send="true"
                                          href="mailto:christer.holmberg@ericsson.com"><span
                                            style="color:purple">christer.holmberg@ericsson.com</span></a>&gt;
                                        wrote:<o:p></o:p></p>
                                    </div>
                                  </div>
                                </div>
                                <div>
                                  <div>
                                    <p class="MsoNormal"><br>
                                      <br>
                                      <br>
                                      <br>
                                      <o:p></o:p></p>
                                  </div>
                                </div>
                                <div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">Hi,</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">&nbsp;</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">As I indicated
                                            at the STOX session (and was
                                            later repeated by Jonathan
                                            Lennox), the draft need to
                                            have a story on SIP forking,
                                            e.g. if what happens if the
                                            INVITE sent from the
                                            interworking node gets
                                            forked, and multiple early
                                            dialogs are created.</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">&nbsp;</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">Regards,</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">&nbsp;</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                                            lang="EN-US">Christer</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <div>
                                        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;">&nbsp;</span><o:p></o:p></p>
                                      </div>
                                    </div>
                                  </div>
                                  <div>
                                    <div>
                                      <p class="MsoNormal"><span
style="font-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;">_______________________________________________<br>
                                          stox mailing list<br>
                                          <a moz-do-not-send="true"
                                            href="mailto:stox@ietf.org"><span
                                              style="color:purple">stox@ietf.org</span></a><br>
                                          <a moz-do-not-send="true"
                                            href="https://www.ietf.org/mailman/listinfo/stox"><span
                                              style="color:purple">https://www.ietf.org/mailman/listinfo/stox</span></a></span><o:p></o:p></p>
                                    </div>
                                  </div>
                                </div>
                              </div>
                              <div>
                                <div>
                                  <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                                </div>
                              </div>
                            </div>
                          </div>
                        </div>
                        <div>
                          <p class="MsoNormal"><span
style="font-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;">_______________________________________________<br>
                              stox mailing list<br>
                              <a moz-do-not-send="true"
                                href="mailto:stox@ietf.org">stox@ietf.org</a><br>
                              <a moz-do-not-send="true"
                                href="https://www.ietf.org/mailman/listinfo/stox">https://www.ietf.org/mailman/listinfo/stox</a></span><o:p></o:p></p>
                        </div>
                      </div>
                    </blockquote>
                  </div>
                  <div>
                    <p class="MsoNormal">&nbsp;<o:p></o:p></p>
                  </div>
                  <p class="MsoNormal"><span
style="font-size:13.5pt;font-family:&quot;Helvetica&quot;,&quot;sans-serif&quot;">_______________________________________________<br>
                      stox mailing list<br>
                      <a moz-do-not-send="true"
                        href="mailto:stox@ietf.org">stox@ietf.org</a><br>
                      <a moz-do-not-send="true"
                        href="https://www.ietf.org/mailman/listinfo/stox">https://www.ietf.org/mailman/listinfo/stox</a><o:p></o:p></span></p>
                </div>
              </div>
              <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
            </div>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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From fippo@goodadvice.pages.de  Sun Aug  4 11:57:22 2013
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Date: Sun, 04 Aug 2013 20:57:03 +0200
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Subject: [Stox] media: mapping a=fmtp to jingle
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The main problem of mapping a=fmtp to jingle and vice versa is that 
jingle has assumed that fmtp is always key-value separated by colons.
See http://xmpp.org/extensions/xep-0167.html#format at the bottom of 
section 4. Several implementors apparently noticed that the text is 
wrong and created workarounds.

I suspect the reasonable advice for the gateway is to translate lines 
with key-value pairs separated by semicolons as described in XEP-0167 
and pick one of the following ways to deal with non-key-value lines:
1) map to <parameter name='' value='thestuff'/> (used by gmail afaik)
2) map to <param name='thestuff' value=''/>
3) map to <param name='attributes'>thestuff</param>
    (XEP-0293/0294 recommend something similar)

How sensitive are SDP implementations in parsing different key-value 
variants (e.g. "key=value;key2=value" vs "key=value; key2=value2"?


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adrian Georgescu <ag@ag-projects.com>
Thread-Topic: VS: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,

> The gateway would care to translate the final response. Getting more then=
 one 180 is not causing any harm nor does those need to be translated into =
different events on XMPP side.

If that's the case, then you should have to problem :) I would suggest to h=
ave some text about that, though.

And, I assume that means that possible early media from the SIP side won't =
be forwarded into the XMPP network either, or?

Regards,

Christer



On 08/04/2013 08:39 PM, Christer Holmberg wrote:
Hi,
=A0
I am not sure I understand.
=A0
The server may implement SIP functions like forking, and fork the INVITE. B=
UT, the gateway will still get the 18x responses for the early dialogs - ju=
st like a SIP phone.
=A0
Regards,
=A0
Christer
=A0
L=E4hett=E4j=E4: Adrian Georgescu [mailto:ag@ag-projects.com]=20
L=E4hetetty: 4. elokuuta 2013 20:20
Vastaanottaja: Christer Holmberg
Kopio: stox@ietf.org
Aihe: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
=A0
The gateway lookups the SIP server for a given domain in the DNS same as an=
y other device would do like a SIP phone configured with credentials under =
that domain. Is the job of that server to properly implement all SIP functi=
ons like forking proxy, handle nat traversal or registrar to name a few com=
mon features.
=A0
You ask what shall the world do if the SIP server of given domain does not =
implemented forking or other feature. My answer is the gateway must not car=
e, is the problem of whomever manages that domain and its applications to h=
andle such features.
=A0
Regards,
Adrian
=A0
=A0
=A0
On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:



Hi,
=A0
There for sure may be a SIP server (a proxy, B2BUA and/or the remote UAS) i=
n the network, but how can you guarantee that it will handle forking? A for=
king proxy will simply forward 18x responses forming the early dialogs towa=
rds the gateway, which then will have to handle them.
=A0
Regards,
=A0
Christer
=A0
L=E4hett=E4j=E4:=A0Adrian Georgescu [mailto:ag@ag-projects.com]=A0
L=E4hetetty:=A02. elokuuta 2013 13:36
Vastaanottaja:=A0Christer Holmberg
Kopio:=A0stox@ietf.org
Aihe:=A0Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
=A0
=A0
On Aug 2, 2013, at 12:30 PM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:




Hi Adrian,
=A0
>I would suggest that best practice is for the SIP traffic to be routed to =
the SIP Proxy/Registrar/Presence Agent of the given domain behind the gatew=
ay.=A0That element has more proper capabilities for handling forking or any=
 other SIP features.=A0
=A0
Ok, but how are you going to ensure that such element exists?
=A0
The gateway knows, it queries the DNS for the domain SIP service records. I=
f no proper records exist, the call flow will stop as there is nothing to r=
ote the calls flows to.=A0
=A0
Such gateway, its very existence, implies that there is both a SIP Server a=
nd and XMPP server for each domain. It is a cross protocol federation, the =
domain must exists otherwise there is nothing to gateway to and there is no=
 reason to deploy it.=A0
=A0
Adrian
=A0
=A0
Regards,
=A0
Christer
=A0
=A0
=A0
=A0
=A0
Otherwise you will move all SIP functionality into the XMPP gateway and mus=
t document that whole SIP universe as running inside that gateway, which ma=
kes little sense.
=A0
Regards,=A0
Adrian
=A0
On Aug 2, 2013, at 12:16 PM, Christer Holmberg <christer.holmberg@ericsson.=
com> wrote:





Hi,
=A0
As I indicated at the STOX session (and was later repeated by Jonathan Lenn=
ox), the draft need to have a story on SIP forking, e.g. if what happens if=
 the INVITE sent from the interworking node gets forked, and multiple early=
 dialogs are created.
=A0
Regards,
=A0
Christer
=A0
_______________________________________________
stox mailing list
stox@ietf.org
https://www.ietf.org/mailman/listinfo/stox
=A0
_______________________________________________
stox mailing list
stox@ietf.org
https://www.ietf.org/mailman/listinfo/stox
=A0
_______________________________________________
stox mailing list
stox@ietf.org
https://www.ietf.org/mailman/listinfo/stox
=A0


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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Am 05.08.2013 07:12, schrieb Christer Holmberg:
> Hi,
>
>> The gateway would care to translate the final response. Getting more then one 180 is not causing any harm nor does those need to be translated into different events on XMPP side.
>
> If that's the case, then you should have to problem :) I would suggest to have some text about that, though.
>
> And, I assume that means that possible early media from the SIP side won't be forwarded into the XMPP network either, or?

http://xmpp.org/extensions/xep-0269.html has some suggestions on how to 
do early media. It's not entirely clear to me if the details are correct 
(e.g. I wouldn't use a raw-udp transport when the initiator only 
indicated support for ice-udp) but the general plan seems ok.

From pkyzivat@alum.mit.edu  Tue Aug  6 09:10:43 2013
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On 8/4/13 8:44 PM, Adrian Georgescu wrote:
> The gateway would care to translate the final response. Getting more
> then one 180 is not causing any harm nor does those need to be
> translated into different events on XMPP side.

I think Christer's point is that the behavior you are describing needs 
to be described somewhere.

The typical issue is that forking results in multiple early dialogs. 
More than one may result in early media. To give good results the GW 
needs to pass on some or all of that early media to the XMPP side. If 
more than one has early media, then its difficult to know what to do. 
You might pass *one* of them on to the XMPP side.

Most times *one* of those early dialogs will return a 200, and the 
others will be cancelled (by the gateway). But the one that returns the 
200 may not be the one you were processing early media from. So there 
may need to be a media change.

It is also possible that more than one of those early dialogs will 
return a 200. Then *something* needs to be done with all of those. The 
simplest is to hang up on one of them, though its also possible to 
conference them. So the GW needs to deal with this possibility, even 
though it will be rare.

You can *hope* that some sip server downstream of the GW will handle all 
of this and present the GW with a call flow that has no forking. But you 
can't count on this. There is nothing the GW can do to guarantee that 
will happen.

	Thanks,
	Paul

> Adrian
>
> On 08/04/2013 08:39 PM, Christer Holmberg wrote:
>>
>> Hi,
>>
>> I am not sure I understand.
>>
>> The server may implement SIP functions like forking, and fork the
>> INVITE. BUT, the gateway will still get the 18x responses for the
>> early dialogs – just like a SIP phone.
>>
>> Regards,
>>
>> Christer
>>
>> *Lähettäjä:*Adrian Georgescu [mailto:ag@ag-projects.com]
>> *Lähetetty:* 4. elokuuta 2013 20:20
>> *Vastaanottaja:* Christer Holmberg
>> *Kopio:* stox@ietf.org
>> *Aihe:* Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
>>
>> The gateway lookups the SIP server for a given domain in the DNS same
>> as any other device would do like a SIP phone configured with
>> credentials under that domain. Is the job of that server to properly
>> implement all SIP functions like forking proxy, handle nat traversal
>> or registrar to name a few common features.
>>
>> You ask what shall the world do if the SIP server of given domain does
>> not implemented forking or other feature. My answer is the gateway
>> must not care, is the problem of whomever manages that domain and its
>> applications to handle such features.
>>
>> Regards,
>>
>> Adrian
>>
>> On Aug 4, 2013, at 6:55 PM, Christer Holmberg wrote:
>>
>>
>>
>> Hi,
>>
>> There for sure may be a SIP server (a proxy, B2BUA and/or the remote
>> UAS) in the network, but how can you guarantee that it will handle
>> forking? A forking proxy will simply forward 18x responses forming the
>> early dialogs towards the gateway, which then will have to handle them.
>>
>> Regards,
>>
>> Christer
>>
>> *Lähettäjä:*Adrian Georgescu [mailto:ag@ag-projects.com]
>> *Lähetetty:*2. elokuuta 2013 13:36
>> *Vastaanottaja:*Christer Holmberg
>> *Kopio:*stox@ietf.org <mailto:stox@ietf.org>
>> *Aihe:*Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
>>
>> On Aug 2, 2013, at 12:30 PM, Christer Holmberg
>> <christer.holmberg@ericsson.com
>> <mailto:christer.holmberg@ericsson.com>> wrote:
>>
>>
>>
>>
>> Hi Adrian,
>>
>> >I would suggest that best practice is for the SIP traffic to be routed
>> to the SIP Proxy/Registrar/Presence Agent of the given domain behind
>> the gateway.That element has more proper capabilities for handling
>> forking or any other SIP features.
>>
>> Ok, but how are you going to ensure that such element exists?
>>
>> The gateway knows, it queries the DNS for the domain SIP service
>> records. If no proper records exist, the call flow will stop as there
>> is nothing to rote the calls flows to.
>>
>> Such gateway, its very existence, implies that there is both a SIP
>> Server and and XMPP server for each domain. It is a cross protocol
>> federation, the domain must exists otherwise there is nothing to
>> gateway to and there is no reason to deploy it.
>>
>> Adrian
>>
>>     Regards,
>>
>>     Christer
>>
>>     Otherwise you will move all SIP functionality into the XMPP
>>     gateway and must document that whole SIP universe as running
>>     inside that gateway, which makes little sense.
>>
>>     Regards,
>>
>>     Adrian
>>
>>     On Aug 2, 2013, at 12:16 PM, Christer Holmberg
>>     <christer.holmberg@ericsson.com
>>     <mailto:christer.holmberg@ericsson.com>> wrote:
>>
>>
>>
>>
>>
>>     Hi,
>>
>>     As I indicated at the STOX session (and was later repeated by
>>     Jonathan Lennox), the draft need to have a story on SIP forking,
>>     e.g. if what happens if the INVITE sent from the interworking node
>>     gets forked, and multiple early dialogs are created.
>>
>>     Regards,
>>
>>     Christer
>>
>>     _______________________________________________
>>     stox mailing list
>>     stox@ietf.org <mailto:stox@ietf.org>
>>     https://www.ietf.org/mailman/listinfo/stox
>>
>>     _______________________________________________
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>>     stox@ietf.org <mailto:stox@ietf.org>
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From pkyzivat@alum.mit.edu  Tue Aug  6 09:17:55 2013
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On 8/4/13 6:57 PM, Christer Holmberg wrote:
> Hi,
>
>>> As I indicated at the STOX session (and was later repeated by Jonathan Lennox), the draft need to have a story on SIP forking, e.g. if what happens if the INVITE sent from the interworking node gets forked, and multiple early dialogs are created.
>>
>> I'll propose text but in general terms, since forking doesn't exist in XMPP, the gateway will need to make sure only one reply is relayed to the XMPP side. Also, there dis very little text about early media in XEP-166, I'll look more into that.
>
> Unfortunately my XMPP knowledge is more or less zero, so at this point I can't really suggest any solution.

Me too.

> However, if early media does not exist in XMPP, and the gateway only needs to care about (at least from a media perspective) the 200 OK, there probably is no problem :)

I'm not sure what it means for early media to not exist in XMPP, or the 
implications of that on an XMPP-SIP GW. If the session is being 
originated from XMPP, and media addresses are provided before the call 
is answered, then early media is possible, even if it will be ignored.

The only way I can see to avoid this entirely is for the GW to initiate 
the initial INVITE without an offer, and *not* support 100rel, so that 
the answer won't be sent until the ACK.

	Thanks,
	Paul

> Regards,
>
> Christer
>
>
>
>
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From saul@ag-projects.com  Wed Aug  7 14:44:05 2013
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Subject: Re: [Stox] Review on -presence
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On Aug 3, 2013, at 9:53 AM, Peter Saint-Andre wrote:

> On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
>> On 7/30/13 5:33 PM, Sa=FAl Ibarra Corretg=E9 wrote:
>>>=20
>>> - Using ID-123kdklejd doesn't seem to work as a valid xs:ID, =
TID-1234
>>> does work though, so we could use TID- as the prefix for tuple
>>> identifiers in examples and such.
>>=20
>> I will double-check that against the XML specification.
>=20
> Hmm. Here is what I see in the XML schema datatype specification:
>=20
>   The =B7value space=B7 of ID is the set of all strings that =B7match=B7=
 the
>   NCName production in [Namespaces in XML].
>=20
> Where the NCName production is defined as follows:
>=20
>   NCName 	::=3D 	(Letter | '_') (NCNameChar)*
>=20
> What software are you using for validation? I'm not seeing a =
constraint
> that an xs:ID needs to begin with three letters.
>=20

I tested it with lxml, a Python wrapper for libxml2. I complained =
because 'ID-123' was too ambiguous for an xs:ID. Not sure if it's an =
implementation detail / issue though.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] Review on -presence
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On Aug 3, 2013, at 10:48 AM, Peter Saint-Andre wrote:

> On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
>> On 7/30/13 5:33 PM, Sa=FAl Ibarra Corretg=E9 wrote:
>>> Hi,
>>>=20
>>> Here is my review of the -presence document:
>>>=20
>>> - Sec 3.3.2 suggests that if the subscription is maintained but we
>>> have no presence state, a PIDF would be generated with a basic =
status
>>> of closed, but what would be the tuple ID, if we don't know any
>>> resource for this user anymore?=20
>=20
> Actually, in this scenario the gateway does know the status because =
the
> XMPP user is still online -- it's just that the SIP user's =
subscription
> to the XMPP user's status has expired.
>=20
>>> We could also send an empty NOTIFY,
>>> which would achieve the same goal.
>>=20
>> The empty NOTIFY sounds more consistent with the spirit of SIP =
presence.
>=20
> I've looked into this further. Here is what I find in RFC 3856 =
(Section
> 6.6.2):
>=20
>   If the resource is not in a meaningful state, RFC 3265 [2] allows =
the
>   body of the initial NOTIFY to be empty.  In the case of presence,
>   that NOTIFY MAY contain a presence document.  This document would
>   indicate whatever presence state the subscriber has been authorized
>   to see; it is interpreted by the subscriber as the current presence
>   state of the presentity.  For pending subscriptions, the state of =
the
>   presentity SHOULD include some kind of textual note that indicates a
>   pending status.
>=20
> And in RFC 3265:
>=20
> 3.1.6.2. Confirmation of Subscription Creation/Refreshing
>=20
>   Upon successfully accepting or refreshing a subscription, notifiers
>   MUST send a NOTIFY message immediately to communicate the current
>   resource state to the subscriber.  This NOTIFY message is sent on =
the
>   same dialog as created by the SUBSCRIBE response.  If the resource
>   has no meaningful state at the time that the SUBSCRIBE message is
>   processed, this NOTIFY message MAY contain an empty or neutral body.
>=20
> Everything there is about an initial NOTIFY confirming a presence
> subscription. However, Section 3.3.2 of the stox-presence spec =
discusses
> what to do if the SIP user's subscription expires:
>=20
>   It is the responsibility of the SIMPLE-XMPP gateway to properly
>   handle the difference between short-lived SIP presence subscriptions
>   and long-lived XMPP presence subscriptions.  The gateway has two
>   options when the SIP user's subscription expires...
>=20
> It seems to me that there's a false assumption here, because in =
general
> doesn't the SIP user agent initiate a refresh to maintain the
> subscription (as long as it's online)? So I think there are two cases =
here:
>=20

Yep, you would usually refresh the subscription.

> (1) SIP user agent is online and generates a refresh by =
re-subscribing.
> In this case, the empty NOTIFY might make sense (although you could
> argue that it's not truly an "initial NOTIFY").
>=20

Yep, I think we could say that you MAY send an empty NOTIFY and compose =
and offline state tuple based on the previous state.

> (2) SIP user agent goes offline and the subscription truly expires. In
> this case, the handling we currently describe seems generally correct =
to me.
>=20

Yes.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] media: mapping a=fmtp to jingle
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Hi Philipp!

On Aug 4, 2013, at 8:57 PM, Philipp Hancke wrote:

> The main problem of mapping a=3Dfmtp to jingle and vice versa is that =
jingle has assumed that fmtp is always key-value separated by colons.
> See http://xmpp.org/extensions/xep-0167.html#format at the bottom of =
section 4. Several implementors apparently noticed that the text is =
wrong and created workarounds.
>=20
> I suspect the reasonable advice for the gateway is to translate lines =
with key-value pairs separated by semicolons as described in XEP-0167 =
and pick one of the following ways to deal with non-key-value lines:
> 1) map to <parameter name=3D'' value=3D'thestuff'/> (used by gmail =
afaik)

I'd go with this one.

> 2) map to <param name=3D'thestuff' value=3D''/>
> 3) map to <param name=3D'attributes'>thestuff</param>
>   (XEP-0293/0294 recommend something similar)
>=20
> How sensitive are SDP implementations in parsing different key-value =
variants (e.g. "key=3Dvalue;key2=3Dvalue" vs "key=3Dvalue; key2=3Dvalue2"?=

>=20

I think they usually just treat it as a freeform text.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On Aug 5, 2013, at 7:12 AM, Christer Holmberg wrote:

> Hi,
>=20
>> The gateway would care to translate the final response. Getting more =
then one 180 is not causing any harm nor does those need to be =
translated into different events on XMPP side.
>=20
> If that's the case, then you should have to problem :) I would suggest =
to have some text about that, though.
>=20
> And, I assume that means that possible early media from the SIP side =
won't be forwarded into the XMPP network either, or?
>=20

My suggestion would be not to do it, since there is no counterpart in =
XMPP. Yes, there is XEP-269, but it has been deferred.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On Aug 6, 2013, at 6:10 PM, Paul Kyzivat wrote:

> On 8/4/13 8:44 PM, Adrian Georgescu wrote:
>> The gateway would care to translate the final response. Getting more
>> then one 180 is not causing any harm nor does those need to be
>> translated into different events on XMPP side.
>=20
> I think Christer's point is that the behavior you are describing needs =
to be described somewhere.
>=20

Indeed.

> The typical issue is that forking results in multiple early dialogs. =
More than one may result in early media. To give good results the GW =
needs to pass on some or all of that early media to the XMPP side. If =
more than one has early media, then its difficult to know what to do. =
You might pass *one* of them on to the XMPP side.
>=20

I don't think we can pass the early media, since there is no way to do =
it today. I'd go for leaving it unspecified. There seems to be some =
action going on for rebooting Jingle, so when Jingle has a story for =
early media a new document could be written specifying it. Would this =
work for you? I know it doesn't cover all those mutli-edarly-dialog =
cases, but I'm not sure we can do nay better at this point.

> Most times *one* of those early dialogs will return a 200, and the =
others will be cancelled (by the gateway). But the one that returns the =
200 may not be the one you were processing early media from. So there =
may need to be a media change.
>=20

Not if we don't translate early media ;-)

> It is also possible that more than one of those early dialogs will =
return a 200. Then *something* needs to be done with all of those. The =
simplest is to hang up on one of them, though its also possible to =
conference them. So the GW needs to deal with this possibility, even =
though it will be rare.
>=20

Yeah, I think we could add some text about it, though what to do will be =
implementation specific, right?

> You can *hope* that some sip server downstream of the GW will handle =
all of this and present the GW with a call flow that has no forking. But =
you can't count on this. There is nothing the GW can do to guarantee =
that will happen.
>=20

Right.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On Aug 5, 2013, at 7:56 AM, Philipp Hancke wrote:

> Am 05.08.2013 07:12, schrieb Christer Holmberg:
>> Hi,
>>=20
>>> The gateway would care to translate the final response. Getting more =
then one 180 is not causing any harm nor does those need to be =
translated into different events on XMPP side.
>>=20
>> If that's the case, then you should have to problem :) I would =
suggest to have some text about that, though.
>>=20
>> And, I assume that means that possible early media from the SIP side =
won't be forwarded into the XMPP network either, or?
>=20
> http://xmpp.org/extensions/xep-0269.html has some suggestions on how =
to do early media. It's not entirely clear to me if the details are =
correct (e.g. I wouldn't use a raw-udp transport when the initiator only =
indicated support for ice-udp) but the general plan seems ok.

Can we rely on it given the fact that is deferred? Are there any plans =
to change stuff in there, in "Jingle 2.0"?

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Am 08.08.2013 00:27, schrieb Saúl Ibarra Corretgé:
>
> On Aug 5, 2013, at 7:56 AM, Philipp Hancke wrote:
>
>> Am 05.08.2013 07:12, schrieb Christer Holmberg:
>>> Hi,
>>>
>>>> The gateway would care to translate the final response. Getting more then one 180 is not causing any harm nor does those need to be translated into different events on XMPP side.
>>>
>>> If that's the case, then you should have to problem :) I would suggest to have some text about that, though.
>>>
>>> And, I assume that means that possible early media from the SIP side won't be forwarded into the XMPP network either, or?
>>
>> http://xmpp.org/extensions/xep-0269.html has some suggestions on how to do early media. It's not entirely clear to me if the details are correct (e.g. I wouldn't use a raw-udp transport when the initiator only indicated support for ice-udp) but the general plan seems ok.
>
> Can we rely on it given the fact that is deferred?

Well, it's deferred because of inactivity, i.e. nobody bothered to 
update it or push it to draft.
yate might have an implementation...

> Are there any plans to change stuff in there, in "Jingle 2.0"?

Not that I know of.

From fippo@goodadvice.pages.de  Thu Aug  8 11:33:59 2013
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Subject: [Stox] review of core, chat, groupchat and presence
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I just noticed i reviewed -00 from the draft package... but anyway, 
quite a number of nits. Read them coming from XMPP land.

Comments prefixed with example of section number.
editorial comments:
- inconsistent usage of note. sometimes (Note: ...) other times (-im,
   presence) Note:
- chat and im prefix examples with '|'

chat:
     example f2: call-id: contain jid? how about uniqueness?.
	    Is there any text on the use of <thread/>?
	in -im @example.com callid is used
     	a=lang could be taken xml:lang?
     before example F4: the gateway acknowledges on behalf of juliet?

     example f5: the initial message in F1 should have an id
	attribute which gets mapped in f5. Does a786hjs2 map to
	thread/call id?

     example f6: shouldn't message-id should be different from f5?
         mapping of message-id to id attribute, should be randomly
	generated if not present

     section 4: sessinos -> sessions;

core:
     the draft says:
	     instant messaging and presence applications of XMPP also
	     need to support im: and pres: URIs
	     as specified in [RFC3860] and [RFC3859] respectively
	while this is encouraged in RFC 6121, those are informative
	references.

     4.2 bullet 1: ref to jid escaping xep here instead of in bullet 3
     5.1: <policy-violation/> is missing from the list.

groupchat:
     2: at least in RFC 6120, jid stands for jabber id, not jabber
	identifier.
     3.1: Room Nickname -> lowercase?
     3.1: example f2: conversion->conversation?
          table 1: thread<->call id
     3.2: example f7: possibly use by= on the error element?
	xep 0045 has the conflicting resource in the from
     3.4: "Occupant JID in another room" -- i don't think mediated
	invites work across different rooms.
         example f1: hecate is benvolio? c&p error from 0045
         example f2: contact juliet@juliet.example.com has a wrong host?
         reeived -> received
         example f3 is not shown
         below f4 there is a sip/2.0 200 ok on its own?
     3.5: table 2: xmpp From/To -> from/to
         ex f9: where is juliets own presence in msrp?
         ex f11: where is juliets own presence with code 110?
         ex f14: mapping <message id/> to Message-ID
         	does msrp have a concept of room history?
     3.6.2: do we have an equivalent disco feature for muc?
     3.7: does xmpp still support the <status/> when in muc? In IRC this
	has caused PART spam
     4.1: ex f5: lack of error until when? must wait until receiving 110
	status?
     4.4: presence broadcast before ack? must buffer until receiving 110
	ack?
     4.5.1: use message id instead of guessing?

im:
     table 4: align tables accross documents and put normative version
	to -core.

pres:
     should mention different concepts, long-lived vs short-lived in
	intro, not only in 3.3.2
     3.1: im uri of the form <pres:>?
     3.3.1: if a subscription already exists -> already defined in 6121,
	not sure if this was in 3921 already
     4.3: table 7 CSeq <-> id?! consistency...

From pkyzivat@alum.mit.edu  Fri Aug  9 11:33:50 2013
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Cc: stox@ietf.org
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On 8/8/13 12:18 AM, Saúl Ibarra Corretgé wrote:
>
> On Aug 6, 2013, at 6:10 PM, Paul Kyzivat wrote:
>
>> On 8/4/13 8:44 PM, Adrian Georgescu wrote:
>>> The gateway would care to translate the final response. Getting more
>>> then one 180 is not causing any harm nor does those need to be
>>> translated into different events on XMPP side.
>>
>> I think Christer's point is that the behavior you are describing needs to be described somewhere.
>>
>
> Indeed.
>
>> The typical issue is that forking results in multiple early dialogs. More than one may result in early media. To give good results the GW needs to pass on some or all of that early media to the XMPP side. If more than one has early media, then its difficult to know what to do. You might pass *one* of them on to the XMPP side.
>>
>
> I don't think we can pass the early media, since there is no way to do it today. I'd go for leaving it unspecified. There seems to be some action going on for rebooting Jingle, so when Jingle has a story for early media a new document could be written specifying it. Would this work for you? I know it doesn't cover all those mutli-edarly-dialog cases, but I'm not sure we can do nay better at this point.

This is a hard conversation to have, since I am almost totally ignorant 
of Jingle.

But I think what happens here must depend a bit on the nature of the GW. 
If it terminates media, then it can swallow the early media. But if it 
is only a signaling GW, and doesn't terminate the media, then it will 
only be negotiating using the SDP addresses of the SIP and Jingle 
endpoints. In that case its hard to avoid the early media. But it all 
depends on the sort of signaling you can do in the GW.

	Thanks,
	Paul

>> Most times *one* of those early dialogs will return a 200, and the others will be cancelled (by the gateway). But the one that returns the 200 may not be the one you were processing early media from. So there may need to be a media change.
>>
>
> Not if we don't translate early media ;-)
>
>> It is also possible that more than one of those early dialogs will return a 200. Then *something* needs to be done with all of those. The simplest is to hang up on one of them, though its also possible to conference them. So the GW needs to deal with this possibility, even though it will be rare.
>>
>
> Yeah, I think we could add some text about it, though what to do will be implementation specific, right?
>
>> You can *hope* that some sip server downstream of the GW will handle all of this and present the GW with a call flow that has no forking. But you can't count on this. There is nothing the GW can do to guarantee that will happen.
>>
>
> Right.
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>


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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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9 aug 2013 kl. 20:27 skrev Paul Kyzivat <pkyzivat@alum.mit.edu>:

>>> The typical issue is that forking results in multiple early dialogs. =
More than one may result in early media. To give good results the GW =
needs to pass on some or all of that early media to the XMPP side. If =
more than one has early media, then its difficult to know what to do. =
You might pass *one* of them on to the XMPP side.
>>>=20
>>=20
>> I don't think we can pass the early media, since there is no way to =
do it today. I'd go for leaving it unspecified. There seems to be some =
action going on for rebooting Jingle, so when Jingle has a story for =
early media a new document could be written specifying it. Would this =
work for you? I know it doesn't cover all those mutli-edarly-dialog =
cases, but I'm not sure we can do nay better at this point.
>=20
Just answer the call on the jingle side when you have early media. If =
you haven't got billing on that side, it doesn't matter. If you have =
billing, then don't answer. Easy to document.

/O=

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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>9 aug 2013 kl. 20:27 skrev Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;:</div>=
<br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><blockquote type=3D"cite" style=3D"font-family: monospace; =
font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><blockquote type=3D"cite">The typical issue is that forking results in =
multiple early dialogs. More than one may result in early media. To give =
good results the GW needs to pass on some or all of that early media to =
the XMPP side. If more than one has early media, then its difficult to =
know what to do. You might pass *one* of them on to the XMPP =
side.<br><br></blockquote><br>I don't think we can pass the early media, =
since there is no way to do it today. I'd go for leaving it unspecified. =
There seems to be some action going on for rebooting Jingle, so when =
Jingle has a story for early media a new document could be written =
specifying it. Would this work for you? I know it doesn't cover all =
those mutli-edarly-dialog cases, but I'm not sure we can do nay better =
at this point.<br></blockquote><br =
class=3D"Apple-interchange-newline"></blockquote></div>Just answer the =
call on the jingle side when you have early media. If you haven't got =
billing on that side, it doesn't matter. If you have billing, then don't =
answer. Easy to document.<div><br></div><div>/O</div></body></html>=

--Apple-Mail=_17476E18-D2C3-4CA4-A54E-B873F2B198CF--

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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On 8/10/13 2:41 PM, Olle E. Johansson wrote:
>
> 9 aug 2013 kl. 20:27 skrev Paul Kyzivat <pkyzivat@alum.mit.edu
> <mailto:pkyzivat@alum.mit.edu>>:
>
>>>> The typical issue is that forking results in multiple early dialogs.
>>>> More than one may result in early media. To give good results the GW
>>>> needs to pass on some or all of that early media to the XMPP side.
>>>> If more than one has early media, then its difficult to know what to
>>>> do. You might pass *one* of them on to the XMPP side.
>>>>
>>>
>>> I don't think we can pass the early media, since there is no way to
>>> do it today. I'd go for leaving it unspecified. There seems to be
>>> some action going on for rebooting Jingle, so when Jingle has a story
>>> for early media a new document could be written specifying it. Would
>>> this work for you? I know it doesn't cover all those
>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better at
>>> this point.
>>
> Just answer the call on the jingle side when you have early media. If
> you haven't got billing on that side, it doesn't matter. If you have
> billing, then don't answer. Easy to document.

That doesn't help very often. The case of concern is when the call 
originates from Jingle and an offer is included. This then gets 
gatewayed to SIP, and then forked. At that point each of the forks could 
start sending media back to the media address in the offer.

The Jingle end doesn't have the option to answer the call.

If the GW terminates and relays media, then it is possible for the GW to 
answer the Jingle call, prior to receiving an answer from any fork. Then 
the GW can decide what media to relay.

But that only works if the GW is a media relay. It isn't an option for a 
signaling only GW.

	Thanks,
	Paul


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From: "Olle E. Johansson" <oej@edvina.net>
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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10 aug 2013 kl. 16:33 skrev Paul Kyzivat <pkyzivat@alum.mit.edu>:

> On 8/10/13 2:41 PM, Olle E. Johansson wrote:
>>=20
>> 9 aug 2013 kl. 20:27 skrev Paul Kyzivat <pkyzivat@alum.mit.edu
>> <mailto:pkyzivat@alum.mit.edu>>:
>>=20
>>>>> The typical issue is that forking results in multiple early =
dialogs.
>>>>> More than one may result in early media. To give good results the =
GW
>>>>> needs to pass on some or all of that early media to the XMPP side.
>>>>> If more than one has early media, then its difficult to know what =
to
>>>>> do. You might pass *one* of them on to the XMPP side.
>>>>>=20
>>>>=20
>>>> I don't think we can pass the early media, since there is no way to
>>>> do it today. I'd go for leaving it unspecified. There seems to be
>>>> some action going on for rebooting Jingle, so when Jingle has a =
story
>>>> for early media a new document could be written specifying it. =
Would
>>>> this work for you? I know it doesn't cover all those
>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better at
>>>> this point.
>>>=20
>> Just answer the call on the jingle side when you have early media. If
>> you haven't got billing on that side, it doesn't matter. If you have
>> billing, then don't answer. Easy to document.
>=20
> That doesn't help very often. The case of concern is when the call =
originates from Jingle and an offer is included. This then gets =
gatewayed to SIP, and then forked. At that point each of the forks could =
start sending media back to the media address in the offer.
>=20
> The Jingle end doesn't have the option to answer the call.
>=20
> If the GW terminates and relays media, then it is possible for the GW =
to answer the Jingle call, prior to receiving an answer from any fork. =
Then the GW can decide what media to relay.
>=20
> But that only works if the GW is a media relay. It isn't an option for =
a signaling only GW.
Right, but I haven't seen anyone really interested in "signaling only" =
gw's. Emil said multiple times during our IETF meeting that it will be a =
b2bua anyway.

Early media is just media - it's the billing that's the difference. =
There's no way to handle multiple early media streams "properly" even if =
you're a sip b2bua without Jingle at all...

So if you have a jingle endpoint placing a call to a SIP URI and the =
gateway gets early media from the SIP side, the gateway can answer the =
jingle call and relay the early media. Whether it wants to mix or stay =
with the first stream in the case of multiple early media stream is =
propably up to the implementation - or it's time to specify this =
behaviour.  Personally I would like to ban sending ring tones with 183 - =
if they used 180 for ring tones and 183 for other messages a gateway =
could suspend the ring tones instead of mixing or just ignoring the =
second media stream.

/O=

From ag@ag-projects.com  Sat Aug 10 08:01:10 2013
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From: Adrian Georgescu <ag@ag-projects.com>
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To: "Olle E. Johansson" <oej@edvina.net>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On Aug 10, 2013, at 2:41 PM, "Olle E. Johansson" <oej@edvina.net> wrote:

>=20
> 9 aug 2013 kl. 20:27 skrev Paul Kyzivat <pkyzivat@alum.mit.edu>:
>=20
>>>> The typical issue is that forking results in multiple early =
dialogs. More than one may result in early media. To give good results =
the GW needs to pass on some or all of that early media to the XMPP =
side. If more than one has early media, then its difficult to know what =
to do. You might pass *one* of them on to the XMPP side.
>>>>=20
>>>=20
>>> I don't think we can pass the early media, since there is no way to =
do it today. I'd go for leaving it unspecified. There seems to be some =
action going on for rebooting Jingle, so when Jingle has a story for =
early media a new document could be written specifying it. Would this =
work for you? I know it doesn't cover all those mutli-edarly-dialog =
cases, but I'm not sure we can do nay better at this point.
>>=20
> Just answer the call on the jingle side when you have early media. If =
you haven't got billing on that side, it doesn't matter. If you have =
billing, then don't answer. Easy to document.

It is already documented.

See =
draft-ietf-stox-media-billing-cannot-live-without-multiple-leg-cdrs-no-way=
-without-billing-dtmf-prack-crack-pl-help-01


Adrian

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


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>On Aug 10, 2013, at 2:41 PM, "Olle E. Johansson" &lt;<a =
href=3D"mailto:oej@edvina.net">oej@edvina.net</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><meta =
http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Diso-8859-1"><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>9 aug 2013 kl. 20:27 skrev Paul Kyzivat &lt;<a =
href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;:</div>=
<br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><blockquote type=3D"cite" style=3D"font-family: monospace; =
font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: =
none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; =
"><blockquote type=3D"cite">The typical issue is that forking results in =
multiple early dialogs. More than one may result in early media. To give =
good results the GW needs to pass on some or all of that early media to =
the XMPP side. If more than one has early media, then its difficult to =
know what to do. You might pass *one* of them on to the XMPP =
side.<br><br></blockquote><br>I don't think we can pass the early media, =
since there is no way to do it today. I'd go for leaving it unspecified. =
There seems to be some action going on for rebooting Jingle, so when =
Jingle has a story for early media a new document could be written =
specifying it. Would this work for you? I know it doesn't cover all =
those mutli-edarly-dialog cases, but I'm not sure we can do nay better =
at this point.<br></blockquote><br =
class=3D"Apple-interchange-newline"></blockquote></div>Just answer the =
call on the jingle side when you have early media. If you haven't got =
billing on that side, it doesn't matter. If you have billing, then don't =
answer. Easy to document.</div></blockquote><div><br></div><div>It is =
already documented.</div><div><br></div><div>See =
draft-ietf-stox-media-billing-cannot-live-without-multiple-leg-cdrs-no-way=
-without-billing-dtmf-prack-crack-pl-help-01</div><div><br></div><div><br>=
</div>Adrian</div><div><br><blockquote type=3D"cite"><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; =
"><div><br></div><div>/O</div></div>______________________________________=
_________<br>stox mailing list<br><a =
href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>https://www.ietf.org/ma=
ilman/listinfo/stox<br></blockquote></div><br></body></html>=

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Olle E. Johansson" <oej@edvina.net>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Cc: "stox@ietf.org" <stox@ietf.org>
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Hi,
=20
>>>>>> The typical issue is that forking results in multiple early dialogs.
>>>>>> More than one may result in early media. To give good results the=20
>>>>>> GW needs to pass on some or all of that early media to the XMPP side=
.
>>>>>> If more than one has early media, then its difficult to know what=20
>>>>>> to do. You might pass *one* of them on to the XMPP side.
>>>>>>=20
>>>>>=20
>>>>> I don't think we can pass the early media, since there is no way to=20
>>>>> do it today. I'd go for leaving it unspecified. There seems to be=20
>>>>> some action going on for rebooting Jingle, so when Jingle has a=20
>>>>> story for early media a new document could be written specifying=20
>>>>> it. Would this work for you? I know it doesn't cover all those=20
>>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better at=20
>>>>> this point.
>>>>=20
>>> Just answer the call on the jingle side when you have early media. If=20
>>> you haven't got billing on that side, it doesn't matter. If you have=20
>>> billing, then don't answer. Easy to document.
>>=20
>> That doesn't help very often. The case of concern is when the call origi=
nates from Jingle and an offer is included. This then gets gatewayed to SIP=
, and then forked. At that point each of the forks could start sending medi=
a back to the media address in the offer.
>>=20
>> The Jingle end doesn't have the option to answer the call.
>>=20
>> If the GW terminates and relays media, then it is possible for the GW to=
 answer the Jingle call, prior to receiving an answer from any fork. Then t=
he GW can decide what media to relay.
>>=20
>> But that only works if the GW is a media relay. It isn't an option for a=
 signaling only GW.
> Right, but I haven't seen anyone really interested in "signaling only" gw=
's. Emil said multiple times during our IETF meeting that it will be a b2bu=
a anyway.

A B2BUA does not necessarily control media. Please take a look at http://to=
ols.ietf.org/html/draft-ietf-straw-b2bua-taxonomy-02, which discusses signa=
ling B2BUAs :)

> Early media is just media - it's the billing that's the difference. There=
's no way to handle multiple early media streams "properly" even if you're =
a sip b2bua without Jingle at all...

That is correct. The difference is that we are now standardizing the behavi=
or of the GW, so we do need to say something.

> So if you have a jingle endpoint placing a call to a SIP URI and the gate=
way gets early media from the SIP side, the gateway can answer the jingle c=
all and relay the=20
> early media. Whether it wants to mix or stay with the first stream in the=
 case of multiple early media stream is propably up to the implementation -=
 or it's time to=20
> specify this behaviour.  Personally I would like to ban sending ring tone=
s with 183 - if they used 180 for ring tones and 183 for other messages a g=
ateway could suspend=20
> the ring tones instead of mixing or just ignoring the second media stream=
.

If you are going to forward all early media from the SIP side (keep in mind=
 that there may be multiple simultaneous early media streams coming from di=
fferent SIP UASs), you also need to map and forward the SDP information rec=
eived on each SIP early dialog to the Jingle side.

Regards,

Christer


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From: Adrian Georgescu <ag@ag-projects.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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I am wondering how a SIP session with multiple media types like RTP and =
MSRP can be translated into an equivalent XMPP signalling without =
bridging media as well.

Adrian


On Aug 11, 2013, at 12:22 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>>>>>> The typical issue is that forking results in multiple early =
dialogs.
>>>>>>> More than one may result in early media. To give good results =
the=20
>>>>>>> GW needs to pass on some or all of that early media to the XMPP =
side.
>>>>>>> If more than one has early media, then its difficult to know =
what=20
>>>>>>> to do. You might pass *one* of them on to the XMPP side.
>>>>>>>=20
>>>>>>=20
>>>>>> I don't think we can pass the early media, since there is no way =
to=20
>>>>>> do it today. I'd go for leaving it unspecified. There seems to be=20=

>>>>>> some action going on for rebooting Jingle, so when Jingle has a=20=

>>>>>> story for early media a new document could be written specifying=20=

>>>>>> it. Would this work for you? I know it doesn't cover all those=20
>>>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better =
at=20
>>>>>> this point.
>>>>>=20
>>>> Just answer the call on the jingle side when you have early media. =
If=20
>>>> you haven't got billing on that side, it doesn't matter. If you =
have=20
>>>> billing, then don't answer. Easy to document.
>>>=20
>>> That doesn't help very often. The case of concern is when the call =
originates from Jingle and an offer is included. This then gets =
gatewayed to SIP, and then forked. At that point each of the forks could =
start sending media back to the media address in the offer.
>>>=20
>>> The Jingle end doesn't have the option to answer the call.
>>>=20
>>> If the GW terminates and relays media, then it is possible for the =
GW to answer the Jingle call, prior to receiving an answer from any =
fork. Then the GW can decide what media to relay.
>>>=20
>>> But that only works if the GW is a media relay. It isn't an option =
for a signaling only GW.
>> Right, but I haven't seen anyone really interested in "signaling =
only" gw's. Emil said multiple times during our IETF meeting that it =
will be a b2bua anyway.
>=20
> A B2BUA does not necessarily control media. Please take a look at =
http://tools.ietf.org/html/draft-ietf-straw-b2bua-taxonomy-02, which =
discusses signaling B2BUAs :)
>=20
>> Early media is just media - it's the billing that's the difference. =
There's no way to handle multiple early media streams "properly" even if =
you're a sip b2bua without Jingle at all...
>=20
> That is correct. The difference is that we are now standardizing the =
behavior of the GW, so we do need to say something.
>=20
>> So if you have a jingle endpoint placing a call to a SIP URI and the =
gateway gets early media from the SIP side, the gateway can answer the =
jingle call and relay the=20
>> early media. Whether it wants to mix or stay with the first stream in =
the case of multiple early media stream is propably up to the =
implementation - or it's time to=20
>> specify this behaviour.  Personally I would like to ban sending ring =
tones with 183 - if they used 180 for ring tones and 183 for other =
messages a gateway could suspend=20
>> the ring tones instead of mixing or just ignoring the second media =
stream.
>=20
> If you are going to forward all early media from the SIP side (keep in =
mind that there may be multiple simultaneous early media streams coming =
from different SIP UASs), you also need to map and forward the SDP =
information received on each SIP early dialog to the Jingle side.
>=20
> Regards,
>=20
> Christer
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


From stpeter@stpeter.im  Mon Aug 12 13:44:52 2013
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Subject: Re: [Stox] Review on -presence
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On 8/7/13 3:43 PM, Saúl Ibarra Corretgé wrote:
> 
> On Aug 3, 2013, at 9:53 AM, Peter Saint-Andre wrote:
> 
>> On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
>>> On 7/30/13 5:33 PM, Saúl Ibarra Corretgé wrote:
>>>> 
>>>> - Using ID-123kdklejd doesn't seem to work as a valid xs:ID,
>>>> TID-1234 does work though, so we could use TID- as the prefix
>>>> for tuple identifiers in examples and such.
>>> 
>>> I will double-check that against the XML specification.
>> 
>> Hmm. Here is what I see in the XML schema datatype specification:
>> 
>> The ·value space· of ID is the set of all strings that ·match· the 
>> NCName production in [Namespaces in XML].
>> 
>> Where the NCName production is defined as follows:
>> 
>> NCName 	::= 	(Letter | '_') (NCNameChar)*
>> 
>> What software are you using for validation? I'm not seeing a
>> constraint that an xs:ID needs to begin with three letters.
>> 
> 
> I tested it with lxml, a Python wrapper for libxml2. I complained
> because 'ID-123' was too ambiguous for an xs:ID. Not sure if it's an
> implementation detail / issue though.

Sounds like it.

In RFC 6120, it's suggested to make the XMPP resourcepart a UUID.
Something like ID-2E855EFA-DA21-48F1-A6EB-18B5231893AC won't trigger
warnings about ambiguity. :-)

Would it help to change some of the examples?

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Mon Aug 12 13:59:01 2013
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On 8/7/13 3:54 PM, Saúl Ibarra Corretgé wrote:
> 
> On Aug 3, 2013, at 10:48 AM, Peter Saint-Andre wrote:
> 
>> On 8/1/13 12:32 AM, Peter Saint-Andre wrote:
>>> On 7/30/13 5:33 PM, Saúl Ibarra Corretgé wrote:
>>>> Hi,
>>>> 
>>>> Here is my review of the -presence document:
>>>> 
>>>> - Sec 3.3.2 suggests that if the subscription is maintained but
>>>> we have no presence state, a PIDF would be generated with a
>>>> basic status of closed, but what would be the tuple ID, if we
>>>> don't know any resource for this user anymore?
>> 
>> Actually, in this scenario the gateway does know the status because
>> the XMPP user is still online -- it's just that the SIP user's
>> subscription to the XMPP user's status has expired.
>> 
>>>> We could also send an empty NOTIFY, which would achieve the
>>>> same goal.
>>> 
>>> The empty NOTIFY sounds more consistent with the spirit of SIP
>>> presence.
>> 
>> I've looked into this further. Here is what I find in RFC 3856
>> (Section 6.6.2):
>> 
>> If the resource is not in a meaningful state, RFC 3265 [2] allows
>> the body of the initial NOTIFY to be empty.  In the case of
>> presence, that NOTIFY MAY contain a presence document.  This
>> document would indicate whatever presence state the subscriber has
>> been authorized to see; it is interpreted by the subscriber as the
>> current presence state of the presentity.  For pending
>> subscriptions, the state of the presentity SHOULD include some kind
>> of textual note that indicates a pending status.
>> 
>> And in RFC 3265:
>> 
>> 3.1.6.2. Confirmation of Subscription Creation/Refreshing
>> 
>> Upon successfully accepting or refreshing a subscription,
>> notifiers MUST send a NOTIFY message immediately to communicate the
>> current resource state to the subscriber.  This NOTIFY message is
>> sent on the same dialog as created by the SUBSCRIBE response.  If
>> the resource has no meaningful state at the time that the SUBSCRIBE
>> message is processed, this NOTIFY message MAY contain an empty or
>> neutral body.
>> 
>> Everything there is about an initial NOTIFY confirming a presence 
>> subscription. However, Section 3.3.2 of the stox-presence spec
>> discusses what to do if the SIP user's subscription expires:
>> 
>> It is the responsibility of the SIMPLE-XMPP gateway to properly 
>> handle the difference between short-lived SIP presence
>> subscriptions and long-lived XMPP presence subscriptions.  The
>> gateway has two options when the SIP user's subscription
>> expires...
>> 
>> It seems to me that there's a false assumption here, because in
>> general doesn't the SIP user agent initiate a refresh to maintain
>> the subscription (as long as it's online)? So I think there are two
>> cases here:
>> 
> 
> Yep, you would usually refresh the subscription.

Agreed.

>> (1) SIP user agent is online and generates a refresh by
>> re-subscribing. In this case, the empty NOTIFY might make sense
>> (although you could argue that it's not truly an "initial
>> NOTIFY").
>> 
> 
> Yep, I think we could say that you MAY send an empty NOTIFY and
> compose and offline state tuple based on the previous state.

In draft-ietf-stox-presence-01 I added the following paragraph:

   For as long as a SIP user is online and interested in receiving
   presence notifications from the XMPP users, the user's SIP user agent
   is responsible for periodically refreshing the subscription by
   sending an updated SUBSCRIBE request with an appropriate value for
   the Expires header.

However, I did not add an example, so IMHO it would be good to do that
in -02.

>> (2) SIP user agent goes offline and the subscription truly expires.
>> In this case, the handling we currently describe seems generally
>> correct to me.
>> 
> 
> Yes.

Thanks for the validation. :-)

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Mon Aug 12 14:26:45 2013
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Hi Philipp, thanks for the review. I'm splitting up my replies for
easier tracking...

On 8/8/13 12:33 PM, Philipp Hancke wrote:
> I just noticed i reviewed -00 from the draft package... but anyway,
> quite a number of nits. Read them coming from XMPP land.
> 
> Comments prefixed with example of section number.
> editorial comments:
> - inconsistent usage of note. sometimes (Note: ...) other times (-im,
>   presence) Note:
> - chat and im prefix examples with '|'
> 

I'll review the documents for consistency, because I like consistency. :)

> core:
>     the draft says:
>          instant messaging and presence applications of XMPP also
>          need to support im: and pres: URIs
>          as specified in [RFC3860] and [RFC3859] respectively
>     while this is encouraged in RFC 6121, those are informative
>     references.

If I understand your comment correctly, you might agree with the
following change...

OLD
   The XMPP address format is specified in [RFC6122]; as discussed in
   [RFC6121], instant messaging and presence applications of XMPP also
   need to support 'im:' and 'pres:' URIs as specified in [RFC3860] and
   [RFC3859] respectively, although such support might simply involve
   leaving resolution of such addresses up to an XMPP server.

NEW
   The XMPP address format is specified in [RFC6122]; in addition,
   [RFC6121] encourages instant messaging and presence applications of
   XMPP to support 'im:' and 'pres:' URIs as specified in [RFC3860] and
   [RFC3859] respectively, although such support might simply involve
   leaving resolution of such addresses up to an XMPP server.

>     4.2 bullet 1: ref to jid escaping xep here instead of in bullet 3

It seems to me that the reference is appropriate in both places, so
adding it to bullet 1 seems reasonable.

>     5.1: <policy-violation/> is missing from the list.

Good catch. What SIP response code do you suggest mapping that to? The
best option I can see is "400 Bad Request".

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Mon Aug 12 14:55:36 2013
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Subject: [Stox] review of presence (was: Re:  review of core, chat, groupchat and presence)
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On 8/8/13 12:33 PM, Philipp Hancke wrote:

> pres:
>     should mention different concepts, long-lived vs short-lived in
>     intro, not only in 3.3.2

Yes, Ben Campbell mentioned that during the WG session in Berlin, too. I
propose adding the following paragraph to the introduction:

   SIP and XMPP differ significantly in their presence subscription
   models, since SIP subscriptions are short-lived (requiring relatively
   frequent refreshes even during a presence session) whereas XMPP
   subscriptions last across presence sessions until they are explicitly
   cancelled.  This document provides suggestions for bridging the gap
   between these very different models.

>     3.1: im uri of the form <pres:>?

Copy and paste error. Will fix.

>     3.3.1: if a subscription already exists -> already defined in 6121,
>     not sure if this was in 3921 already

RFC 3921 said:

   However, if the user receives a presence stanza of type
   "subscribe" from a contact to whom the user has already granted
   permission to see the user's presence information (e.g., in cases
   when the contact is seeking to resynchronize subscription states),
   the user's server SHOULD auto-reply on behalf of the user.

I suspect that RFC 6121 goes into more detail about that scenario, since
it goes into more detail about everything. ;-)

>     4.3: table 7 CSeq <-> id?! consistency...

That was fixed in -01.

Thanks for the feedback!

Peter

-- 
Peter Saint-Andre
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From yana@sip-communicator.org  Tue Aug 13 17:02:05 2013
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Hi all,

This is a reminder that the review deadline for drafts: =
draft-ietf-stox-core, draft-ietf-stox-presence is approaching (August =
16). So, those of you interested in reviewing please do so before the =
end of the week (I remember at least 6 six people raising hands at the =
meeting in Berlin).

Regards,
Yana=

From pkyzivat@alum.mit.edu  Wed Aug 14 08:53:27 2013
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Subject: Re: [Stox] Review on -presence
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On 8/3/13 10:48 AM, Peter Saint-Andre wrote:

> (1) SIP user agent is online and generates a refresh by re-subscribing.
> In this case, the empty NOTIFY might make sense (although you could
> argue that it's not truly an "initial NOTIFY").

An initial notify is the one directly in response to a SUBSCRIBE, even a 
re-SUBSCRIBE. So I don't think there should be any controversy.

	Thanks,
	Paul

From stpeter@stpeter.im  Wed Aug 14 10:43:12 2013
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On 8/14/13 9:53 AM, Paul Kyzivat wrote:
> On 8/3/13 10:48 AM, Peter Saint-Andre wrote:
> 
>> (1) SIP user agent is online and generates a refresh by re-subscribing.
>> In this case, the empty NOTIFY might make sense (although you could
>> argue that it's not truly an "initial NOTIFY").
> 
> An initial notify is the one directly in response to a SUBSCRIBE, even a
> re-SUBSCRIBE. So I don't think there should be any controversy.

Thanks for the verification.

So under this model, the SIP user agent would send a re-SUBSCRIBE and it
seems to me that the gateway then SHOULD (?) respond with a NOTIFY from
the SIP representation of the XMPP user/resource. There are two cases
for the NOTIFY:

1. If the XMPP user/resource is not in a "meaningful state" (cf. RFC
3856 and RFC 3265), the NOTIFY SHOULD (?) be empty.

2. If the XMPP user/resource is in a meaningful state, the NOTIFY SHOULD
(?) contain data about the user/resource.

Does that sound right?

Peter

-- 
Peter Saint-Andre
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From pkyzivat@alum.mit.edu  Wed Aug 14 10:47:20 2013
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Subject: Re: [Stox] Review on -presence
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On 8/14/13 7:43 PM, Peter Saint-Andre wrote:
> On 8/14/13 9:53 AM, Paul Kyzivat wrote:
>> On 8/3/13 10:48 AM, Peter Saint-Andre wrote:
>>
>>> (1) SIP user agent is online and generates a refresh by re-subscribing.
>>> In this case, the empty NOTIFY might make sense (although you could
>>> argue that it's not truly an "initial NOTIFY").
>>
>> An initial notify is the one directly in response to a SUBSCRIBE, even a
>> re-SUBSCRIBE. So I don't think there should be any controversy.
>
> Thanks for the verification.
>
> So under this model, the SIP user agent would send a re-SUBSCRIBE and it
> seems to me that the gateway then SHOULD (?) respond with a NOTIFY from
> the SIP representation of the XMPP user/resource. There are two cases
> for the NOTIFY:
>
> 1. If the XMPP user/resource is not in a "meaningful state" (cf. RFC
> 3856 and RFC 3265), the NOTIFY SHOULD (?) be empty.

I think this is probably ok.

> 2. If the XMPP user/resource is in a meaningful state, the NOTIFY SHOULD
> (?) contain data about the user/resource.
>
> Does that sound right?

Yeah.

	Thanks,
	Paul


From stpeter@stpeter.im  Wed Aug 14 11:04:03 2013
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Subject: Re: [Stox] Reminder core and presence review deadline
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On 8/13/13 6:01 PM, Yana Stamcheva wrote:
> Hi all,
> 
> This is a reminder that the review deadline for drafts:
> draft-ietf-stox-core, draft-ietf-stox-presence is approaching (August
> 16). So, those of you interested in reviewing please do so before the
> end of the week (I remember at least 6 six people raising hands at
> the meeting in Berlin).

Thanks for the reminder.

How would the chairs like to proceed on updated versions? In my working
copies I have incorporated the results of list discussion, so I can
submit revised I-Ds for core and presence after August 16 or really at
any time. (Once I do that I will process feedback on the next set, i.e.,
im and chat.)

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Wed Aug 14 11:28:18 2013
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Subject: Re: [Stox] Review on -presence
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On 8/14/13 11:47 AM, Paul Kyzivat wrote:
> On 8/14/13 7:43 PM, Peter Saint-Andre wrote:
>> On 8/14/13 9:53 AM, Paul Kyzivat wrote:
>>> On 8/3/13 10:48 AM, Peter Saint-Andre wrote:
>>>
>>>> (1) SIP user agent is online and generates a refresh by re-subscribing.
>>>> In this case, the empty NOTIFY might make sense (although you could
>>>> argue that it's not truly an "initial NOTIFY").
>>>
>>> An initial notify is the one directly in response to a SUBSCRIBE, even a
>>> re-SUBSCRIBE. So I don't think there should be any controversy.
>>
>> Thanks for the verification.
>>
>> So under this model, the SIP user agent would send a re-SUBSCRIBE and it
>> seems to me that the gateway then SHOULD (?) respond with a NOTIFY from
>> the SIP representation of the XMPP user/resource. There are two cases
>> for the NOTIFY:
>>
>> 1. If the XMPP user/resource is not in a "meaningful state" (cf. RFC
>> 3856 and RFC 3265), the NOTIFY SHOULD (?) be empty.
> 
> I think this is probably ok.
> 
>> 2. If the XMPP user/resource is in a meaningful state, the NOTIFY SHOULD
>> (?) contain data about the user/resource.
>>
>> Does that sound right?
> 
> Yeah.

OK, good. Here is proposed text for Section 3.3.2:

   For as long as a SIP user is online and interested in receiving
   presence notifications from the XMPP users, the user's SIP user agent
   is responsible for periodically refreshing the subscription by
   sending an updated SUBSCRIBE request with an appropriate value for
   the Expires header.  In response, the SIMPLE-XMPP gateway SHOULD send
   a SIP NOTIFY to the user agent; if the gateway has meaningful
   information about the availability state of the XMPP user then the
   NOTIFY SHOULD communicate that information (e.g., by containing a
   PIDF body [RFC3863] with the relevant data), whereas if the gateway
   does not have meaningful information about the availability state of
   the XMPP user then the NOTIFY SHOULD be empty as allowed by
   [RFC3265].

That would come before the existing discussion of handling by the
SIMPLE-XMPP gateway.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From pkyzivat@alum.mit.edu  Wed Aug 14 11:48:03 2013
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Subject: Re: [Stox] Review on -presence
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On 8/14/13 8:28 PM, Peter Saint-Andre wrote:

> OK, good. Here is proposed text for Section 3.3.2:
>
>     For as long as a SIP user is online and interested in receiving
>     presence notifications from the XMPP users, the user's SIP user agent
>     is responsible for periodically refreshing the subscription by
>     sending an updated SUBSCRIBE request with an appropriate value for
>     the Expires header.  In response, the SIMPLE-XMPP gateway SHOULD send

S/SHOULD/MUST/

I'm using RFC6665 as my reference, rather than 3265.
See section 4.2.1.2 - the NOTIFY is required.

	Thanks,
	Paul

>     a SIP NOTIFY to the user agent; if the gateway has meaningful
>     information about the availability state of the XMPP user then the
>     NOTIFY SHOULD communicate that information (e.g., by containing a
>     PIDF body [RFC3863] with the relevant data), whereas if the gateway
>     does not have meaningful information about the availability state of
>     the XMPP user then the NOTIFY SHOULD be empty as allowed by
>     [RFC3265].
>
> That would come before the existing discussion of handling by the
> SIMPLE-XMPP gateway.
>
> Peter
>


From stpeter@stpeter.im  Wed Aug 14 12:03:07 2013
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Subject: Re: [Stox] Review on -presence
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On 8/14/13 12:47 PM, Paul Kyzivat wrote:
> On 8/14/13 8:28 PM, Peter Saint-Andre wrote:
> 
>> OK, good. Here is proposed text for Section 3.3.2:
>>
>>     For as long as a SIP user is online and interested in receiving
>>     presence notifications from the XMPP users, the user's SIP user agent
>>     is responsible for periodically refreshing the subscription by
>>     sending an updated SUBSCRIBE request with an appropriate value for
>>     the Expires header.  In response, the SIMPLE-XMPP gateway SHOULD send
> 
> S/SHOULD/MUST/
> 
> I'm using RFC6665 as my reference, rather than 3265.
> See section 4.2.1.2 - the NOTIFY is required.

Duly noted!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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On 8/11/13 12:33 AM, Adrian Georgescu wrote:
> I am wondering how a SIP session with multiple media types like RTP and MSRP can be translated into an equivalent XMPP signalling without bridging media as well.

I suppose the MSRP would indeed have to be terminated by the GW and 
translated into XMPP format. But it could still avoid handling the RTP.

	Thanks,
	Paul

> Adrian
>
>
> On Aug 11, 2013, at 12:22 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>>>>>>>> The typical issue is that forking results in multiple early dialogs.
>>>>>>>> More than one may result in early media. To give good results the
>>>>>>>> GW needs to pass on some or all of that early media to the XMPP side.
>>>>>>>> If more than one has early media, then its difficult to know what
>>>>>>>> to do. You might pass *one* of them on to the XMPP side.
>>>>>>>>
>>>>>>>
>>>>>>> I don't think we can pass the early media, since there is no way to
>>>>>>> do it today. I'd go for leaving it unspecified. There seems to be
>>>>>>> some action going on for rebooting Jingle, so when Jingle has a
>>>>>>> story for early media a new document could be written specifying
>>>>>>> it. Would this work for you? I know it doesn't cover all those
>>>>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better at
>>>>>>> this point.
>>>>>>
>>>>> Just answer the call on the jingle side when you have early media. If
>>>>> you haven't got billing on that side, it doesn't matter. If you have
>>>>> billing, then don't answer. Easy to document.
>>>>
>>>> That doesn't help very often. The case of concern is when the call originates from Jingle and an offer is included. This then gets gatewayed to SIP, and then forked. At that point each of the forks could start sending media back to the media address in the offer.
>>>>
>>>> The Jingle end doesn't have the option to answer the call.
>>>>
>>>> If the GW terminates and relays media, then it is possible for the GW to answer the Jingle call, prior to receiving an answer from any fork. Then the GW can decide what media to relay.
>>>>
>>>> But that only works if the GW is a media relay. It isn't an option for a signaling only GW.
>>> Right, but I haven't seen anyone really interested in "signaling only" gw's. Emil said multiple times during our IETF meeting that it will be a b2bua anyway.
>>
>> A B2BUA does not necessarily control media. Please take a look at http://tools.ietf.org/html/draft-ietf-straw-b2bua-taxonomy-02, which discusses signaling B2BUAs :)
>>
>>> Early media is just media - it's the billing that's the difference. There's no way to handle multiple early media streams "properly" even if you're a sip b2bua without Jingle at all...
>>
>> That is correct. The difference is that we are now standardizing the behavior of the GW, so we do need to say something.
>>
>>> So if you have a jingle endpoint placing a call to a SIP URI and the gateway gets early media from the SIP side, the gateway can answer the jingle call and relay the
>>> early media. Whether it wants to mix or stay with the first stream in the case of multiple early media stream is propably up to the implementation - or it's time to
>>> specify this behaviour.  Personally I would like to ban sending ring tones with 183 - if they used 180 for ring tones and 183 for other messages a gateway could suspend
>>> the ring tones instead of mixing or just ignoring the second media stream.
>>
>> If you are going to forward all early media from the SIP side (keep in mind that there may be multiple simultaneous early media streams coming from different SIP UASs), you also need to map and forward the SDP information received on each SIP early dialog to the Jingle side.
>>
>> Regards,
>>
>> Christer
>>
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>
>
>


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Hi all,

Please find attached the notes of the STOX session recorded by Jean Mahoney=
 and Philipp Hancke. I have also collected the main issues raised and discu=
ssed below as a summary. Some of them have already been discussed on the li=
st after the meeting. You can find more details about each within the minut=
es.

Please let the chairs know if you have any comments on these before we post=
 them for the proceedings.

Regards,
	Markus

----

* Charter

Principles: At the moment consider only features that are mature in both SI=
P and XMPP. Mature means the IETF and/or XSF standards are completed, and t=
here are known and deployed implementations.

Q: Is file transfer on the charter?
A: Not at this point. The XMPP/XSF file transfer spec(s) do not currently m=
eet the matureness criteria. Once they stabilize, SIP mapping can be consid=
ered.

* Core

- Issue: Should something general about forking be defined here or is it pu=
rely a -media issue?
- Issue: Add text about Max-Forwards and Max-Breath. Loop detection in gene=
ral.=20

* IM

- Issue: Contact header is not allowed in SIP MESSAGE, need to fix the draf=
t.=20

* Presence

- Issue: Mapping between XMPP's permanent and SIP's soft-state (refreshed) =
subscriptions.

* Chat

- Issue: Mapping between XMPP ChatStates (XEP-0085) and isComposing event (=
RFC 3994).
- Issue: Mapping between message receipts (XEP-0184) and MSRP REPORT chunks=
 (RFC 4975).
- Consider if the above two are included or possibly split into a forthcomi=
ng advanced-chat draft.
- Issue: Draft says gateway determines MSRP is supported before sending INV=
ITE. How?

* Groupchat

- Issue: Need to define what to do in case of multiple clients. Use and map=
ping of JID, AoR, contact.=20

* Media

- Issue: "Hold" needs to be more carefully defined. What does it actually m=
ean on both sides and how are those meanings mapped. a=3D<direction> does n=
ot always mean "Hold".=20
- Issue: Forking needs to be explained either in -media or -core.
- Issue: a=3Dfmtp mapping.

* Next steps

- Reviews for -core and -presence first by August 16
- Reviews for -im and -chat by August 30
- -groupchat and -media will need further revisions





=20




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Content-Description: notes-stox
Content-Disposition: attachment; filename="notes-stox"; size=8390;
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Thread-Topic: [Stox] Reminder core and presence review deadline
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Hi Peter,

Peter Saint-Andre wrote:
>=20
> Thanks for the reminder.
>=20
> How would the chairs like to proceed on updated versions? In my working
> copies I have incorporated the results of list discussion, so I can submi=
t
> revised I-Ds for core and presence after August 16 or really at any time.

Would you be able to submit a diff or summary of changes that you have alre=
ady planned for -core and -presence to the list so that reviewers who haven=
't yet started can take those into account? Perhaps we should wait until ne=
xt week before publishing -02 not to confuse anyone about which version to =
review right now.=20

Markus

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Subject: Re: [Stox] Reminder core and presence review deadline
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On 8/14/13 1:32 PM, Markus.Isomaki@nokia.com wrote:
> Hi Peter,
> 
> Peter Saint-Andre wrote:
>> 
>> Thanks for the reminder.
>> 
>> How would the chairs like to proceed on updated versions? In my
>> working copies I have incorporated the results of list discussion,
>> so I can submit revised I-Ds for core and presence after August 16
>> or really at any time.
> 
> Would you be able to submit a diff or summary of changes that you
> have already planned for -core and -presence to the list so that
> reviewers who haven't yet started can take those into account?

I use the IETF SVN repository for this work so people can track the
changes there:

https://svn.tools.ietf.org/svn/wg/stox/

All the modifications have been discussed on the list.

> Perhaps we should wait until next week before publishing -02 not to
> confuse anyone about which version to review right now.

Sure. I don't think any of the changes are really big, but publishing
-02 early next week would give people a chance to see the very latest text.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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From: Adrian Georgescu <ag@ag-projects.com>
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References: <7594FB04B1934943A5C02806D1A2204B1C418E51@ESESSMB209.ericsson.se> <8C703C0F-181C-4E36-8A59-727B45C0A1B5@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C418FAC@ESESSMB209.ericsson.se> <B7CBEFB1-2A23-4337-90DD-D8157123A1AD@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C41C17E@ESESSMB209.ericsson.se> <6E2B9398-C9C5-47F1-AC5D-2C7E23AB6E69@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C41C208@ESESSMB209.ericsson.se> <51FEA128.2060806@ag-projects.com> <52011FEA.5040104@alum.mit.edu> <6A3963CE-AD8C-4C93-9E39-9268BA82AF4B@ag-projects.com> <520534A9.9050100@alum.mit.edu> <8A0B7E66-0F4F-41CF-8CC0-E5BD77EFE0F5@edvina.net> <52064F57.80309@alum.mit.edu> <AF343715-E436-44E4-A2DE-062AA0526DB0@edvina.net> <7594FB04B1934943A5C02806D1A2204B1C43CA2E@ESESSMB209.ericsson.se> <4B4DCCFA-EF19-4F93-A740-6A9F5003DB4D@ag-projects.com> <520BD7CB.6020909@alum.mit.edu>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: "stox@ietf.org" <stox@ietf.org>, "Olle E. Johansson" <oej@edvina.net>, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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How would that work when there is one SIP session with two media types, =
RTP audio and MSRP chat, where would such split occur?

Adrian

On Aug 14, 2013, at 9:17 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> On 8/11/13 12:33 AM, Adrian Georgescu wrote:
>> I am wondering how a SIP session with multiple media types like RTP =
and MSRP can be translated into an equivalent XMPP signalling without =
bridging media as well.
>=20
> I suppose the MSRP would indeed have to be terminated by the GW and =
translated into XMPP format. But it could still avoid handling the RTP.
>=20
> 	Thanks,
> 	Paul
>=20
>> Adrian
>>=20
>>=20
>> On Aug 11, 2013, at 12:22 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
>>=20
>>> Hi,
>>>=20
>>>>>>>>> The typical issue is that forking results in multiple early =
dialogs.
>>>>>>>>> More than one may result in early media. To give good results =
the
>>>>>>>>> GW needs to pass on some or all of that early media to the =
XMPP side.
>>>>>>>>> If more than one has early media, then its difficult to know =
what
>>>>>>>>> to do. You might pass *one* of them on to the XMPP side.
>>>>>>>>>=20
>>>>>>>>=20
>>>>>>>> I don't think we can pass the early media, since there is no =
way to
>>>>>>>> do it today. I'd go for leaving it unspecified. There seems to =
be
>>>>>>>> some action going on for rebooting Jingle, so when Jingle has a
>>>>>>>> story for early media a new document could be written =
specifying
>>>>>>>> it. Would this work for you? I know it doesn't cover all those
>>>>>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay =
better at
>>>>>>>> this point.
>>>>>>>=20
>>>>>> Just answer the call on the jingle side when you have early =
media. If
>>>>>> you haven't got billing on that side, it doesn't matter. If you =
have
>>>>>> billing, then don't answer. Easy to document.
>>>>>=20
>>>>> That doesn't help very often. The case of concern is when the call =
originates from Jingle and an offer is included. This then gets =
gatewayed to SIP, and then forked. At that point each of the forks could =
start sending media back to the media address in the offer.
>>>>>=20
>>>>> The Jingle end doesn't have the option to answer the call.
>>>>>=20
>>>>> If the GW terminates and relays media, then it is possible for the =
GW to answer the Jingle call, prior to receiving an answer from any =
fork. Then the GW can decide what media to relay.
>>>>>=20
>>>>> But that only works if the GW is a media relay. It isn't an option =
for a signaling only GW.
>>>> Right, but I haven't seen anyone really interested in "signaling =
only" gw's. Emil said multiple times during our IETF meeting that it =
will be a b2bua anyway.
>>>=20
>>> A B2BUA does not necessarily control media. Please take a look at =
http://tools.ietf.org/html/draft-ietf-straw-b2bua-taxonomy-02, which =
discusses signaling B2BUAs :)
>>>=20
>>>> Early media is just media - it's the billing that's the difference. =
There's no way to handle multiple early media streams "properly" even if =
you're a sip b2bua without Jingle at all...
>>>=20
>>> That is correct. The difference is that we are now standardizing the =
behavior of the GW, so we do need to say something.
>>>=20
>>>> So if you have a jingle endpoint placing a call to a SIP URI and =
the gateway gets early media from the SIP side, the gateway can answer =
the jingle call and relay the
>>>> early media. Whether it wants to mix or stay with the first stream =
in the case of multiple early media stream is propably up to the =
implementation - or it's time to
>>>> specify this behaviour.  Personally I would like to ban sending =
ring tones with 183 - if they used 180 for ring tones and 183 for other =
messages a gateway could suspend
>>>> the ring tones instead of mixing or just ignoring the second media =
stream.
>>>=20
>>> If you are going to forward all early media from the SIP side (keep =
in mind that there may be multiple simultaneous early media streams =
coming from different SIP UASs), you also need to map and forward the =
SDP information received on each SIP early dialog to the Jingle side.
>>>=20
>>> Regards,
>>>=20
>>> Christer
>>>=20
>>> _______________________________________________
>>> stox mailing list
>>> stox@ietf.org
>>> https://www.ietf.org/mailman/listinfo/stox
>>>=20
>>=20
>>=20
>=20
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>=20


From pkyzivat@alum.mit.edu  Wed Aug 14 13:19:59 2013
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References: <7594FB04B1934943A5C02806D1A2204B1C418E51@ESESSMB209.ericsson.se> <8C703C0F-181C-4E36-8A59-727B45C0A1B5@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C418FAC@ESESSMB209.ericsson.se> <B7CBEFB1-2A23-4337-90DD-D8157123A1AD@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C41C17E@ESESSMB209.ericsson.se> <6E2B9398-C9C5-47F1-AC5D-2C7E23AB6E69@ag-projects.com> <7594FB04B1934943A5C02806D1A2204B1C41C208@ESESSMB209.ericsson.se> <51FEA128.2060806@ag-projects.com> <52011FEA.5040104@alum.mit.edu> <6A3963CE-AD8C-4C93-9E39-9268BA82AF4B@ag-projects.com> <520534A9.9050100@alum.mit.edu> <8A0B7E66-0F4F-41CF-8CC0-E5BD77EFE0F5@edvina.net> <52064F57.80309@alum.mit.edu> <AF343715-E436-44E4-A2DE-062AA0526DB0@edvina.net> <7594FB04B1934943A5C02806D1A2204B1C43CA2E@ESESSMB209.ericsson.se> <4B4DCCFA-EF19-4F93-A740-6A9F5003DB4D@ag-projects.com> <520BD7CB.6020909@alum.mit.edu> <AC61A87E-5CF2-4F33-844B-E68396078CC4@ag-projects.com>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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On 8/14/13 9:52 PM, Adrian Georgescu wrote:
> How would that work when there is one SIP session with two media types, RTP audio and MSRP chat, where would such split occur?

It depends a bit on how this should look on the xmpp side. (Not my 
thing.) Assuming you can use a single xmpp session for the voice and the 
chat, then it would look something like:

XMPP                                     SIP
CLIENT               GW                  CLIENT
                               sip
                       X=====================
            xmpp       |
   ####################+
                       |       msrp
                       X.....................

                      rtp
   ------------------------------------------

(Sorry if the above is mangled by different clients.) So the GW 
translates sip signaling to/from XMPP signalling. It also terminates the 
MSRP on the sip side, and translates the content into XMPP signaling in 
the same session. But it hands off the RTP addresses from the xmpp 
signaling to sip signaling so that the RTP media passes e2e.

	Thanks,
	Paul

> Adrian
>
> On Aug 14, 2013, at 9:17 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>
>> On 8/11/13 12:33 AM, Adrian Georgescu wrote:
>>> I am wondering how a SIP session with multiple media types like RTP and MSRP can be translated into an equivalent XMPP signalling without bridging media as well.
>>
>> I suppose the MSRP would indeed have to be terminated by the GW and translated into XMPP format. But it could still avoid handling the RTP.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Adrian
>>>
>>>
>>> On Aug 11, 2013, at 12:22 AM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>>>
>>>> Hi,
>>>>
>>>>>>>>>> The typical issue is that forking results in multiple early dialogs.
>>>>>>>>>> More than one may result in early media. To give good results the
>>>>>>>>>> GW needs to pass on some or all of that early media to the XMPP side.
>>>>>>>>>> If more than one has early media, then its difficult to know what
>>>>>>>>>> to do. You might pass *one* of them on to the XMPP side.
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> I don't think we can pass the early media, since there is no way to
>>>>>>>>> do it today. I'd go for leaving it unspecified. There seems to be
>>>>>>>>> some action going on for rebooting Jingle, so when Jingle has a
>>>>>>>>> story for early media a new document could be written specifying
>>>>>>>>> it. Would this work for you? I know it doesn't cover all those
>>>>>>>>> mutli-edarly-dialog cases, but I'm not sure we can do nay better at
>>>>>>>>> this point.
>>>>>>>>
>>>>>>> Just answer the call on the jingle side when you have early media. If
>>>>>>> you haven't got billing on that side, it doesn't matter. If you have
>>>>>>> billing, then don't answer. Easy to document.
>>>>>>
>>>>>> That doesn't help very often. The case of concern is when the call originates from Jingle and an offer is included. This then gets gatewayed to SIP, and then forked. At that point each of the forks could start sending media back to the media address in the offer.
>>>>>>
>>>>>> The Jingle end doesn't have the option to answer the call.
>>>>>>
>>>>>> If the GW terminates and relays media, then it is possible for the GW to answer the Jingle call, prior to receiving an answer from any fork. Then the GW can decide what media to relay.
>>>>>>
>>>>>> But that only works if the GW is a media relay. It isn't an option for a signaling only GW.
>>>>> Right, but I haven't seen anyone really interested in "signaling only" gw's. Emil said multiple times during our IETF meeting that it will be a b2bua anyway.
>>>>
>>>> A B2BUA does not necessarily control media. Please take a look at http://tools.ietf.org/html/draft-ietf-straw-b2bua-taxonomy-02, which discusses signaling B2BUAs :)
>>>>
>>>>> Early media is just media - it's the billing that's the difference. There's no way to handle multiple early media streams "properly" even if you're a sip b2bua without Jingle at all...
>>>>
>>>> That is correct. The difference is that we are now standardizing the behavior of the GW, so we do need to say something.
>>>>
>>>>> So if you have a jingle endpoint placing a call to a SIP URI and the gateway gets early media from the SIP side, the gateway can answer the jingle call and relay the
>>>>> early media. Whether it wants to mix or stay with the first stream in the case of multiple early media stream is propably up to the implementation - or it's time to
>>>>> specify this behaviour.  Personally I would like to ban sending ring tones with 183 - if they used 180 for ring tones and 183 for other messages a gateway could suspend
>>>>> the ring tones instead of mixing or just ignoring the second media stream.
>>>>
>>>> If you are going to forward all early media from the SIP side (keep in mind that there may be multiple simultaneous early media streams coming from different SIP UASs), you also need to map and forward the SDP information received on each SIP early dialog to the Jingle side.
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> _______________________________________________
>>>> stox mailing list
>>>> stox@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/stox
>>>>
>>>
>>>
>>
>> _______________________________________________
>> stox mailing list
>> stox@ietf.org
>> https://www.ietf.org/mailman/listinfo/stox
>>
>
>


From pkyzivat@alum.mit.edu  Wed Aug 14 13:26:21 2013
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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On 8/2/13 12:50 PM, Christer Holmberg wrote:
> Hi,
>
> If you want to signal who sent the message, I believe we can find another mechanism for that.

I don't think so. This is a *gateway* function. We aren't expecting the 
SIP or XMPP endpoints to do anything special. If the SIP user sends a 
MESSAGE, then that normally doesn't contain a URI of the sending 
endpoint. So we have nothing to put into the XMPP.

I could imagine using Reply-To for this purpose. There is nothing to 
*prevent* use of this. But neither is there currently any reason to 
expect that it will be used.

	Thanks,
	Paul

> The GRUU is not really intended to indicate user identity anyway, and IF you want to use GRUU I believe you need to define how the gateway gets assigned the GRUU.
>
> Regards,
>
> Christer
>
> -----Alkuperäinen viesti-----
> Lähettäjä: Saúl Ibarra Corretgé [mailto:saul@ag-projects.com]
> Lähetetty: 2. elokuuta 2013 13:42
> Vastaanottaja: Christer Holmberg
> Kopio: stox@ietf.org
> Aihe: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
>
> Hi Christer,
>
> On Aug 2, 2013, at 12:13 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> As I indicated at the STOX session, the SIP MESSAGE examples in the draft contain a Contact header field, which is explicitly forbidden.
>>
>> Section 4 of RFC 3248 says:
>>
>> 	"MESSAGE requests do not initiate dialogs.  User Agents MUST NOT
>> 	insert Contact header fields into MESSAGE requests."
>>
>> My question in the meeting was whether this is simply an editorial
>> bug, or whether the Contact header field is actually used for something (there is nothing in the normative text), e.g. to ensure that MESSAGE requests are routed to the correct user agents (someone mentioned that GRUU might be needed).
>>
>
> I was the one who mentioned GRUU. We do need it if we want to know who sent the message exactly, but since the -im spec just deals with pager mode messages maybe we can live with just strangulating the bare JID. Peter?
>
>
> Regards,
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


From pkyzivat@alum.mit.edu  Wed Aug 14 13:29:08 2013
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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On 8/2/13 12:54 PM, Saúl Ibarra Corretgé wrote:
>
> On Aug 2, 2013, at 12:50 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>> Hi,
>>
>> If you want to signal who sent the message, I believe we can find another mechanism for that.
>>
>
> I guess we are open to suggestions :-)
>
>> The GRUU is not really intended to indicate user identity anyway, and IF you want to use GRUU I believe you need to define how the gateway gets assigned the GRUU.
>>
>
> -core defines how we use GRUU which is required, in a nutshell, a GRUU is translated to a full JID and vice versa. Now, IIRC GRUUs are used just in the Contact header (which we can't use here) and not in the From header, for instance.

There is nothing to *prevent* use of a gruu in From, but it isn't what 
is normally used.

If you happen to find a gruu in From, then it probably makes sense to 
translate it to a full JID.

	Thanks,
	Paul


From saul@ag-projects.com  Fri Aug 16 01:00:51 2013
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Subject: Re: [Stox] review of core, chat, groupchat and presence
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Hi Philip,

Thanks for the feedback, I hope to look into these issues over the =
weekend and either make changes or come back with more questions :-)

>=20
> groupchat:
>    2: at least in RFC 6120, jid stands for jabber id, not jabber
> 	identifier.
>    3.1: Room Nickname -> lowercase?
>    3.1: example f2: conversion->conversation?
>         table 1: thread<->call id
>    3.2: example f7: possibly use by=3D on the error element?
> 	xep 0045 has the conflicting resource in the from
>    3.4: "Occupant JID in another room" -- i don't think mediated
> 	invites work across different rooms.
>        example f1: hecate is benvolio? c&p error from 0045
>        example f2: contact juliet@juliet.example.com has a wrong host?
>        reeived -> received
>        example f3 is not shown
>        below f4 there is a sip/2.0 200 ok on its own?
>    3.5: table 2: xmpp From/To -> from/to
>        ex f9: where is juliets own presence in msrp?
>        ex f11: where is juliets own presence with code 110?
>        ex f14: mapping <message id/> to Message-ID
>        	does msrp have a concept of room history?
>    3.6.2: do we have an equivalent disco feature for muc?
>    3.7: does xmpp still support the <status/> when in muc? In IRC this
> 	has caused PART spam
>    4.1: ex f5: lack of error until when? must wait until receiving 110
> 	status?
>    4.4: presence broadcast before ack? must buffer until receiving 110
> 	ack?
>    4.5.1: use message id instead of guessing?


--
Sa=FAl Ibarra Corretg=E9
AG Projects




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>=20
> There is nothing to *prevent* use of a gruu in From, but it isn't what =
is normally used.
>=20
> If you happen to find a gruu in From, then it probably makes sense to =
translate it to a full JID.
>=20

I thought it could only be used in the Contact header, but as you said, =
RFC5627 doesn't enforce that. So I guess we should mention this in the =
-im spec. This could also be used if someone writes a spec to map XMPP =
chat messages to SIP MESSAGE in the future.

Thanks!

--
Sa=FAl Ibarra Corretg=E9
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From saul@ag-projects.com  Fri Aug 16 01:17:26 2013
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Subject: Re: [Stox] Review on -presence
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>=20
> OK, good. Here is proposed text for Section 3.3.2:
>=20
>   For as long as a SIP user is online and interested in receiving
>   presence notifications from the XMPP users, the user's SIP user =
agent
>   is responsible for periodically refreshing the subscription by
>   sending an updated SUBSCRIBE request with an appropriate value for
>   the Expires header.  In response, the SIMPLE-XMPP gateway SHOULD =
send
>   a SIP NOTIFY to the user agent; if the gateway has meaningful
>   information about the availability state of the XMPP user then the
>   NOTIFY SHOULD communicate that information (e.g., by containing a
>   PIDF body [RFC3863] with the relevant data), whereas if the gateway
>   does not have meaningful information about the availability state of
>   the XMPP user then the NOTIFY SHOULD be empty as allowed by
>   [RFC3265].
>=20
> That would come before the existing discussion of handling by the
> SIMPLE-XMPP gateway.
>=20

Sounds good, I guess you'll just change the SHOULDs to MUSTs as pointed =
out by Paul.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From saul@ag-projects.com  Fri Aug 16 01:20:27 2013
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To: Diana Cionoiu <diana@null.ro>
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Subject: Re: [Stox] Media Sessions (draft-ietf-stox-media-01) and forking
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Thanks for confirming, Diana.

On Aug 8, 2013, at 11:53 PM, Diana Cionoiu wrote:

> Hello,
>=20
> Yate has an implementation for xep-0269. That was designed talking =
carefully into consideration the way xmpp work.
>=20
> Diana
>=20
> On 08/08/2013 11:31 AM, Philipp Hancke wrote:
>> Am 08.08.2013 00:27, schrieb Sa=FAl Ibarra Corretg=E9:
>>>=20
>>> On Aug 5, 2013, at 7:56 AM, Philipp Hancke wrote:
>>>=20
>>>> Am 05.08.2013 07:12, schrieb Christer Holmberg:
>>>>> Hi,
>>>>>=20
>>>>>> The gateway would care to translate the final response. Getting =
more then one 180 is not causing any harm nor does those need to be =
translated into different events on XMPP side.
>>>>>=20
>>>>> If that's the case, then you should have to problem :) I would =
suggest to have some text about that, though.
>>>>>=20
>>>>> And, I assume that means that possible early media from the SIP =
side won't be forwarded into the XMPP network either, or?
>>>>=20
>>>> http://xmpp.org/extensions/xep-0269.html has some suggestions on =
how to do early media. It's not entirely clear to me if the details are =
correct (e.g. I wouldn't use a raw-udp transport when the initiator only =
indicated support for ice-udp) but the general plan seems ok.
>>>=20
>>> Can we rely on it given the fact that is deferred?
>>=20
>> Well, it's deferred because of inactivity, i.e. nobody bothered to =
update it or push it to draft.
>> yate might have an implementation...
>>=20
>>> Are there any plans to change stuff in there, in "Jingle 2.0"?
>>=20
>> Not that I know of.
>=20

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Fri Aug 16 13:04:44 2013
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On 8/16/13 2:17 AM, Saúl Ibarra Corretgé wrote:
>>
>> OK, good. Here is proposed text for Section 3.3.2:
>>
>>   For as long as a SIP user is online and interested in receiving
>>   presence notifications from the XMPP users, the user's SIP user agent
>>   is responsible for periodically refreshing the subscription by
>>   sending an updated SUBSCRIBE request with an appropriate value for
>>   the Expires header.  In response, the SIMPLE-XMPP gateway SHOULD send
>>   a SIP NOTIFY to the user agent; if the gateway has meaningful
>>   information about the availability state of the XMPP user then the
>>   NOTIFY SHOULD communicate that information (e.g., by containing a
>>   PIDF body [RFC3863] with the relevant data), whereas if the gateway
>>   does not have meaningful information about the availability state of
>>   the XMPP user then the NOTIFY SHOULD be empty as allowed by
>>   [RFC3265].
>>
>> That would come before the existing discussion of handling by the
>> SIMPLE-XMPP gateway.
>>
> 
> Sounds good, I guess you'll just change the SHOULDs to MUSTs as pointed out by Paul.

Yes, that will be in the next version.

Peter

-- 
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From stpeter@stpeter.im  Fri Aug 16 19:28:21 2013
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Subject: [Stox] core: response code mappings
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The SIP Parameters Registry has a list of SIP response codes:

http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7

A number of those are not specified in RFC 3261. Thus the question
arises: for which codes do we need to define mappings? We could define
mappings for all of them, but I wonder if that's advisable. Some of the
additional codes are specified in RFCs that update RFC 3261 (e.g., code
440 from RFC 5393), whereas other codes are specified in "non-core" RFCs
that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
perhaps make sense to map the "core" codes and not the "non-core" codes?

Peter

-- 
Peter Saint-Andre
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From stpeter@stpeter.im  Fri Aug 16 20:25:37 2013
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Subject: Re: [Stox] Reminder core and presence review deadline
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On 8/14/13 1:49 PM, Peter Saint-Andre wrote:
> On 8/14/13 1:32 PM, Markus.Isomaki@nokia.com wrote:
>> Hi Peter,
>>
>> Peter Saint-Andre wrote:
>>>
>>> Thanks for the reminder.
>>>
>>> How would the chairs like to proceed on updated versions? In my
>>> working copies I have incorporated the results of list discussion,
>>> so I can submit revised I-Ds for core and presence after August 16
>>> or really at any time.
>>
>> Would you be able to submit a diff or summary of changes that you
>> have already planned for -core and -presence to the list so that
>> reviewers who haven't yet started can take those into account?
> 
> I use the IETF SVN repository for this work so people can track the
> changes there:
> 
> https://svn.tools.ietf.org/svn/wg/stox/
> 
> All the modifications have been discussed on the list.
> 
>> Perhaps we should wait until next week before publishing -02 not to
>> confuse anyone about which version to review right now.
> 
> Sure. I don't think any of the changes are really big, but publishing
> -02 early next week would give people a chance to see the very latest text.

An update now that August 16 has come and gone: I think presence-02 is
ready to be submitted, but there's an open issue with the core spec
(response code mappings) so I'll wait for conclusions from list
discussion to emerge before submitting core-02.

Peter

-- 
Peter Saint-Andre
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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-02.txt
	Pages           : 21
	Date            : 2013-08-16

Abstract:
   This document defines a bi-directional protocol mapping for the
   exchange of presence information between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-presence

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-presence-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-presence-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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Subject: Re: [Stox] core: response code mappings
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17 aug 2013 kl. 04:28 skrev Peter Saint-Andre <stpeter@stpeter.im>:

> The SIP Parameters Registry has a list of SIP response codes:
>=20
> =
http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-para=
meters-7
>=20
> A number of those are not specified in RFC 3261. Thus the question
> arises: for which codes do we need to define mappings? We could define
> mappings for all of them, but I wonder if that's advisable. Some of =
the
> additional codes are specified in RFCs that update RFC 3261 (e.g., =
code
> 440 from RFC 5393), whereas other codes are specified in "non-core" =
RFCs
> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
> perhaps make sense to map the "core" codes and not the "non-core" =
codes?

I don't really have any good answer. Can just look into the Asterisk =
code and see
that we've implemented a selection... :-)

The important part is to have a generic mapping for 2xx, 3xx, 4xx, 5xx =
and 6xx codes,
so that if a gateway gets an unknown code, it knows how to map it to =
something. In SIP,
an unknown 4xx code is the same as 400 etc.

After years of struggling with translations between SIP and ISDN in the =
Asterisk core,
I know that this is a hard problem. Especially in the case where you =
have SIP -> xmpp -> SIP=20
and want the same error code on the other side. You simply can't please =
everyone when
mapping between two protocols.

/O=

From stpeter@stpeter.im  Sun Aug 18 20:20:17 2013
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Hi Olle, that is helpful feedback.

On 8/17/13 1:28 AM, Olle E. Johansson wrote:
> 
> 17 aug 2013 kl. 04:28 skrev Peter Saint-Andre <stpeter@stpeter.im>:
> 
>> The SIP Parameters Registry has a list of SIP response codes:
>>
>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>
>> A number of those are not specified in RFC 3261. Thus the question
>> arises: for which codes do we need to define mappings? We could define
>> mappings for all of them, but I wonder if that's advisable. Some of the
>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>> 440 from RFC 5393), whereas other codes are specified in "non-core" RFCs
>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>> perhaps make sense to map the "core" codes and not the "non-core" codes?
> 
> I don't really have any good answer. Can just look into the Asterisk code and see
> that we've implemented a selection... :-)
> 
> The important part is to have a generic mapping for 2xx, 3xx, 4xx, 5xx and 6xx codes,
> so that if a gateway gets an unknown code, it knows how to map it to something. In SIP,
> an unknown 4xx code is the same as 400 etc.

True. I think we can provide that information in the stox-core spec
(e.g., in XMPP <bad-request/> is like a default 400). That way, if an
entity encounters a response code that it doesn't understand, at least
it can map it appropriately.

> After years of struggling with translations between SIP and ISDN in the Asterisk core,
> I know that this is a hard problem. Especially in the case where you have SIP -> xmpp -> SIP 
> and want the same error code on the other side. You simply can't please everyone when
> mapping between two protocols.

Agreed. Those round-trip mappings are especially difficult, but it would
be good to look at a few examples because I think we'll end up at
default things like SIP 400 codes in a lot of situations. I'll
investigate the matter soon and post some proposed text to this list.


Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Mon Aug 19 10:19:02 2013
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On 7/29/13 4:49 AM, Peter Saint-Andre wrote:
> On 7/26/13 5:55 PM, Peter Saint-Andre wrote:
>> On 7/26/13 4:50 PM, Michael Lundberg wrote:
>>
>>> One clarifying question: when you say "if the SIP implementation
>>> supports the namespace", do you mean the XMPP-to-SIP gateway or the
>>> SIP user agent?
>>>
>>>> Good question.  It would be benificial if both supported the
>>>> namespace, but only the gateway is probably required to.  If the
>>>> client supports the namespace, then the gateway would just need to map
>>>> between the elements described in this document.
>>>
>>>> If the client doesn't support the namespace, the gateway would most
>>>> likely need to do an additional translation into a namespace the
>>>> client does understand.  In this case, the values might not be the
>>>> same between the two namespaces, and therefore things are 'lost' in
>>>> translation. This is one of the big issues with presence mapping today
>>>> as many implementations have thier own implementation specific
>>>> namespace, which makes it hard to map between different
>>>> implementations.  Both the implementation specific and common
>>>> namespaces could coexist, where the implementation specific namespace
>>>> is used for internal communication and a common, standard namespace
>>>> (e.g., 'jabber:client' ) is used when communicating between different
>>>> implementations.
>>
>> Yes, I think that makes sense. I'm not sure exactly how to translate
>> that into text (especially the part about proprietary / non-standard
>> namespaces), but I'll work something up for consideration by the list.
> 
> Here is some proposed text:
> 
> OLD
>    5.  Some implementations support custom extensions to encapsulate
>        this information; however, there is no need to standardize a PIDF
>        extension for this purpose, since PIDF is already extensible and
>        thus the <show/> element can be included directly, qualified by
>        the 'jabber:client' namespace in the PIDF XML.  The examples in
>        this document illustrate this usage, which is RECOMMENDED.  The
>        most useful values are likely "away" and "dnd", although note
>        that the latter value merely means "busy" and does not imply that
>        a server or client ought to block incoming traffic while the user
>        is in that state.
> 
> NEW
>    5.  Some implementations support custom extensions to encapsulate
>        detailed information about availability; however, there is no
>        need to standardize a PIDF extension for this purpose, since
>        PIDF is already extensible and thus the <show/> element
>        (qualified by the 'jabber:client' namespace) can be included
>        directly in the PIDF XML.  The examples in this document
>        illustrate this usage, which is RECOMMENDED.  The most useful
>        values are likely "away" and "dnd", although note that the
>        latter value merely means "busy" and does not imply that a
>        server or client ought to block incoming traffic while the user
>        is in that state.  Naturally, a gateway can choose to translate
>        a custom extension into an established value of the <show/>
>        element [RFC6121], or translate a <show/> element into a custom
>        extension that the gateway knows is supported by the user agent
>        of the intended recipient.  Unfortunately, this behavior does
>        not guarantee that information will not be lost; to help prevent
>        information loss, a gateway ought to include both the <show/>
>        element and the custom extension if the gateway cannot suitably
>        translate the custom value into a <show/> value.
> 
> Mike, does that text address your concern? Do we need to say more (or
> less) than that?

It seems that I neglected to include this text in the latest version.
I'll submit another revision.

My apologies.

Peter

-- 
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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Presence
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-presence-03.txt
	Pages           : 22
	Date            : 2013-08-19

Abstract:
   This document defines a bi-directional protocol mapping for the
   exchange of presence information between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-presence

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-presence-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-presence-03


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:

> * Core
> 
> - Issue: Should something general about forking be defined here or is it purely a -media issue?

I think this is best left for the stox-media spec.

> - Issue: Add text about Max-Forwards and Max-Breath. Loop detection in general. 

IMHO we could say something like the following in the stox-core spec...

   A gateway between SIP and XMPP (in either direction) effectively acts
   as a SIP proxy server.  Therefore the amplification vulnerability
   described in [RFC5393] can manifest itself, and a gateway SHOULD
   implement the loop-detection methods defined in that specification to
   help mitigate the possibility of amplification attacks.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From saul@ag-projects.com  Mon Aug 19 12:14:10 2013
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Subject: Re: [Stox] core: response code mappings
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On Aug 19, 2013, at 5:20 AM, Peter Saint-Andre wrote:

> Hi Olle, that is helpful feedback.
>=20
> On 8/17/13 1:28 AM, Olle E. Johansson wrote:
>>=20
>> 17 aug 2013 kl. 04:28 skrev Peter Saint-Andre <stpeter@stpeter.im>:
>>=20
>>> The SIP Parameters Registry has a list of SIP response codes:
>>>=20
>>> =
http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-para=
meters-7
>>>=20
>>> A number of those are not specified in RFC 3261. Thus the question
>>> arises: for which codes do we need to define mappings? We could =
define
>>> mappings for all of them, but I wonder if that's advisable. Some of =
the
>>> additional codes are specified in RFCs that update RFC 3261 (e.g., =
code
>>> 440 from RFC 5393), whereas other codes are specified in "non-core" =
RFCs
>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>>> perhaps make sense to map the "core" codes and not the "non-core" =
codes?
>>=20
>> I don't really have any good answer. Can just look into the Asterisk =
code and see
>> that we've implemented a selection... :-)
>>=20
>> The important part is to have a generic mapping for 2xx, 3xx, 4xx, =
5xx and 6xx codes,
>> so that if a gateway gets an unknown code, it knows how to map it to =
something. In SIP,
>> an unknown 4xx code is the same as 400 etc.
>=20
> True. I think we can provide that information in the stox-core spec
> (e.g., in XMPP <bad-request/> is like a default 400). That way, if an
> entity encounters a response code that it doesn't understand, at least
> it can map it appropriately.
>=20

+1

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From pkyzivat@alum.mit.edu  Mon Aug 19 13:03:10 2013
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Subject: Re: [Stox] core: response code mappings
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On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
> The SIP Parameters Registry has a list of SIP response codes:
>
> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>
> A number of those are not specified in RFC 3261. Thus the question
> arises: for which codes do we need to define mappings? We could define
> mappings for all of them, but I wonder if that's advisable. Some of the
> additional codes are specified in RFCs that update RFC 3261 (e.g., code
> 440 from RFC 5393), whereas other codes are specified in "non-core" RFCs
> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
> perhaps make sense to map the "core" codes and not the "non-core" codes?

I know this is almost a non-answer, but...

The codes that are defined in other RFCs are there to support features 
that are introduced in those RFCs. If there is a mapping of that feature 
to XMPP, then there should be a mapping of the code.

If there is no mapping of the feature, then mapping the code is less 
important, but still perhaps useful in some cases. E.g., it could 
conceivably show up in a Reason code based on signaling that was never 
gatewayed to xmpp. But maybe a default mapping would be sufficient in 
those cases.

	Thanks,
	Paul


From stpeter@stpeter.im  Mon Aug 19 13:24:13 2013
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On 8/19/13 2:02 PM, Paul Kyzivat wrote:
> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>> The SIP Parameters Registry has a list of SIP response codes:
>>
>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>
>>
>> A number of those are not specified in RFC 3261. Thus the question
>> arises: for which codes do we need to define mappings? We could define
>> mappings for all of them, but I wonder if that's advisable. Some of the
>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>> 440 from RFC 5393), whereas other codes are specified in "non-core" RFCs
>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>> perhaps make sense to map the "core" codes and not the "non-core" codes?
> 
> I know this is almost a non-answer, but...
> 
> The codes that are defined in other RFCs are there to support features
> that are introduced in those RFCs. If there is a mapping of that feature
> to XMPP, then there should be a mapping of the code.

That makes sense.

> If there is no mapping of the feature, then mapping the code is less
> important, but still perhaps useful in some cases. E.g., it could
> conceivably show up in a Reason code based on signaling that was never
> gatewayed to xmpp. But maybe a default mapping would be sufficient in
> those cases.

I suspect that the default mapping would be fine.

And in general, there might be more art than science here.

I have written some proposed text for this section, but it's fairly long
so my inclination is to submit a revised I-D and then folks can review
what I've written and post to the list with feedback.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From internet-drafts@ietf.org  Mon Aug 19 13:41:25 2013
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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Architecture,=
 Addresses, and Error Handling
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-02.txt
	Pages           : 16
	Date            : 2013-08-19

Abstract:
   As a foundation for the definition of bidirectional protocol mappings
   between the Session Initiation Protocol (SIP) and the Extensible
   Messaging and Presence Protocol (XMPP), this document specifies the
   architectural assumptions underlying such mappings as well as the
   mapping of addresses and error conditions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-core

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-core-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-02


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From stpeter@stpeter.im  Mon Aug 19 13:42:51 2013
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As previously mentioned.

I view some of this text as provisional, but submitting a revised I-D
seemed to be the best way for folks to review it.

On 8/19/13 2:41 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 SIP-TO-XMPP Working Group of the IETF.
> 
> 	Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Architecture, Addresses, and Error Handling
> 	Author(s)       : Peter Saint-Andre
>                           Avshalom Houri
>                           Joe Hildebrand
> 	Filename        : draft-ietf-stox-core-02.txt
> 	Pages           : 16
> 	Date            : 2013-08-19
> 
> Abstract:
>    As a foundation for the definition of bidirectional protocol mappings
>    between the Session Initiation Protocol (SIP) and the Extensible
>    Messaging and Presence Protocol (XMPP), this document specifies the
>    architectural assumptions underlying such mappings as well as the
>    mapping of addresses and error conditions.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-core
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-core-02
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-core-02
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
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> stox@ietf.org
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> 



From miconda@gmail.com  Mon Aug 19 14:44:14 2013
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Subject: Re: [Stox] core: response code mappings
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On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
> On 8/19/13 2:02 PM, Paul Kyzivat wrote:
>> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>>> The SIP Parameters Registry has a list of SIP response codes:
>>>
>>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>>
>>>
>>> A number of those are not specified in RFC 3261. Thus the question
>>> arises: for which codes do we need to define mappings? We could define
>>> mappings for all of them, but I wonder if that's advisable. Some of the
>>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>>> 440 from RFC 5393), whereas other codes are specified in "non-core" RFCs
>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>>> perhaps make sense to map the "core" codes and not the "non-core" codes?
>> I know this is almost a non-answer, but...
>>
>> The codes that are defined in other RFCs are there to support features
>> that are introduced in those RFCs. If there is a mapping of that feature
>> to XMPP, then there should be a mapping of the code.
> That makes sense.
>
>> If there is no mapping of the feature, then mapping the code is less
>> important, but still perhaps useful in some cases. E.g., it could
>> conceivably show up in a Reason code based on signaling that was never
>> gatewayed to xmpp. But maybe a default mapping would be sufficient in
>> those cases.
> I suspect that the default mapping would be fine.
>
> And in general, there might be more art than science here.
>
> I have written some proposed text for this section, but it's fairly long
> so my inclination is to submit a revised I-D and then folks can review
> what I've written and post to the list with feedback.
Looking at the latest draft, it seems to bring confusion on mapping from 
xmpp-to-sip on 400 code. I read this thread and kind of understood that 
any not-obvious-mapping similar-to-a-4xx case would be handled same as 
400 in SIP, but typically the 400 response is given for a malformed sip 
requests (like broken grammar).

After quick read, <conflict/> would map more suggestively to 403 
Forbidden. Maybe <policy-violation> and <undefined-condition/> could get 
other mappings as well.

Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From stpeter@stpeter.im  Mon Aug 19 15:07:17 2013
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On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
> 
> On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
>> On 8/19/13 2:02 PM, Paul Kyzivat wrote:
>>> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>>>> The SIP Parameters Registry has a list of SIP response codes:
>>>>
>>>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>>>
>>>>
>>>>
>>>> A number of those are not specified in RFC 3261. Thus the question
>>>> arises: for which codes do we need to define mappings? We could define
>>>> mappings for all of them, but I wonder if that's advisable. Some of the
>>>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>>>> 440 from RFC 5393), whereas other codes are specified in "non-core"
>>>> RFCs
>>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>>>> perhaps make sense to map the "core" codes and not the "non-core"
>>>> codes?
>>> I know this is almost a non-answer, but...
>>>
>>> The codes that are defined in other RFCs are there to support features
>>> that are introduced in those RFCs. If there is a mapping of that feature
>>> to XMPP, then there should be a mapping of the code.
>> That makes sense.
>>
>>> If there is no mapping of the feature, then mapping the code is less
>>> important, but still perhaps useful in some cases. E.g., it could
>>> conceivably show up in a Reason code based on signaling that was never
>>> gatewayed to xmpp. But maybe a default mapping would be sufficient in
>>> those cases.
>> I suspect that the default mapping would be fine.
>>
>> And in general, there might be more art than science here.
>>
>> I have written some proposed text for this section, but it's fairly long
>> so my inclination is to submit a revised I-D and then folks can review
>> what I've written and post to the list with feedback.
> Looking at the latest draft, it seems to bring confusion on mapping from
> xmpp-to-sip on 400 code. I read this thread and kind of understood that
> any not-obvious-mapping similar-to-a-4xx case would be handled same as
> 400 in SIP, but typically the 400 response is given for a malformed sip
> requests (like broken grammar).

So your suggestion is that we avoid mapping XMPP error conditions to SIP
400 unless the XMPP condition is caused by a malformed request?

> After quick read, <conflict/> would map more suggestively to 403
> Forbidden. 

As I said, it's perhaps more art than science. But mapping <conflict/>
to 403 seems reasonable if we accept your suggestion about not mapping
to 400 unless we have a good reason to do so.

> Maybe <policy-violation> and <undefined-condition/> could get
> other mappings as well.

Well, depending on the policy being violated, <policy-violation/> might
map to 413, 414, 513, or even 406/606.

Given that <undefined-condition/> is something of a catch-all, it's not
clear to me what we'd map it to in a consistent way. That would probably
depend on the application-specific element contained in the error stanza.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On 8/16/13 2:14 AM, Saúl Ibarra Corretgé wrote:
>> 
>> There is nothing to *prevent* use of a gruu in From, but it isn't
>> what is normally used.
>> 
>> If you happen to find a gruu in From, then it probably makes sense
>> to translate it to a full JID.
>> 
> 
> I thought it could only be used in the Contact header, but as you
> said, RFC5627 doesn't enforce that. So I guess we should mention this
> in the -im spec. This could also be used if someone writes a spec to
> map XMPP chat messages to SIP MESSAGE in the future.
> 
> Thanks!

And in this instance the SIP From header would be added by the gateway,
right?

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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19 aug 2013 kl. 20:02 skrev Peter Saint-Andre <stpeter@stpeter.im>:

> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>=20
>> * Core
>>=20
>> - Issue: Should something general about forking be defined here or is =
it purely a -media issue?
>=20
> I think this is best left for the stox-media spec.
Do remember to handle the case where one sip INVITE can get multiple 200 =
OK. It's a corner case, but it does happen.
And it's signalling.

>=20
>> - Issue: Add text about Max-Forwards and Max-Breath. Loop detection =
in general.=20
>=20
> IMHO we could say something like the following in the stox-core =
spec...
>=20
>   A gateway between SIP and XMPP (in either direction) effectively =
acts
>   as a SIP proxy server.  Therefore the amplification vulnerability
b2bua, not a proxy. There's a draft in the straw wg mandating b2bua's to =
support
max-forwards and max-breadth

>   described in [RFC5393] can manifest itself, and a gateway SHOULD
>   implement the loop-detection methods defined in that specification =
to
>   help mitigate the possibility of amplification attacks.

Agree. How are these headers handled in XMPP?

/O=

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To: Peter Saint-Andre <stpeter@stpeter.im>
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Cc: stox@ietf.org, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On Aug 20, 2013, at 5:20 AM, Peter Saint-Andre wrote:

> On 8/16/13 2:14 AM, Sa=FAl Ibarra Corretg=E9 wrote:
>>>=20
>>> There is nothing to *prevent* use of a gruu in From, but it isn't
>>> what is normally used.
>>>=20
>>> If you happen to find a gruu in From, then it probably makes sense
>>> to translate it to a full JID.
>>>=20
>>=20
>> I thought it could only be used in the Contact header, but as you
>> said, RFC5627 doesn't enforce that. So I guess we should mention this
>> in the -im spec. This could also be used if someone writes a spec to
>> map XMPP chat messages to SIP MESSAGE in the future.
>>=20
>> Thanks!
>=20
> And in this instance the SIP =46rom header would be added by the =
gateway,
> right?
>=20

Yes, by translating the from attribute of the XMPP message stanza. So I =
think we are good. It would be nice to have this in the examples, that =
is, no Contact header, and the GRUU in the From.


--
Sa=FAl Ibarra Corretg=E9
AG Projects




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Date: Tue, 20 Aug 2013 10:43:47 +0200
From: Daniel-Constantin Mierla <miconda@gmail.com>
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Cc: stox@ietf.org, Paul Kyzivat <pkyzivat@alum.mit.edu>
Subject: Re: [Stox] core: response code mappings
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On 8/20/13 12:07 AM, Peter Saint-Andre wrote:
> On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
>> On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
>>> On 8/19/13 2:02 PM, Paul Kyzivat wrote:
>>>> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>>>>> The SIP Parameters Registry has a list of SIP response codes:
>>>>>
>>>>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>>>>
>>>>>
>>>>>
>>>>> A number of those are not specified in RFC 3261. Thus the question
>>>>> arises: for which codes do we need to define mappings? We could define
>>>>> mappings for all of them, but I wonder if that's advisable. Some of the
>>>>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>>>>> 440 from RFC 5393), whereas other codes are specified in "non-core"
>>>>> RFCs
>>>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>>>>> perhaps make sense to map the "core" codes and not the "non-core"
>>>>> codes?
>>>> I know this is almost a non-answer, but...
>>>>
>>>> The codes that are defined in other RFCs are there to support features
>>>> that are introduced in those RFCs. If there is a mapping of that feature
>>>> to XMPP, then there should be a mapping of the code.
>>> That makes sense.
>>>
>>>> If there is no mapping of the feature, then mapping the code is less
>>>> important, but still perhaps useful in some cases. E.g., it could
>>>> conceivably show up in a Reason code based on signaling that was never
>>>> gatewayed to xmpp. But maybe a default mapping would be sufficient in
>>>> those cases.
>>> I suspect that the default mapping would be fine.
>>>
>>> And in general, there might be more art than science here.
>>>
>>> I have written some proposed text for this section, but it's fairly long
>>> so my inclination is to submit a revised I-D and then folks can review
>>> what I've written and post to the list with feedback.
>> Looking at the latest draft, it seems to bring confusion on mapping from
>> xmpp-to-sip on 400 code. I read this thread and kind of understood that
>> any not-obvious-mapping similar-to-a-4xx case would be handled same as
>> 400 in SIP, but typically the 400 response is given for a malformed sip
>> requests (like broken grammar).
> So your suggestion is that we avoid mapping XMPP error conditions to SIP
> 400 unless the XMPP condition is caused by a malformed request?
Yes, <bad-request/> qualifies for that. Otherwise, could create 
confusion for implementers of sip endpoints - from sip rfc:
21.4.1 400 Bad Request


    The request could not be understood due to malformed syntax.  The
    Reason-Phrase SHOULD identify the syntax problem in more detail, for
    example, "Missing Call-ID header field".



>
>> After quick read, <conflict/> would map more suggestively to 403
>> Forbidden.
> As I said, it's perhaps more art than science. But mapping <conflict/>
> to 403 seems reasonable if we accept your suggestion about not mapping
> to 400 unless we have a good reason to do so.
>
>> Maybe <policy-violation> and <undefined-condition/> could get
>> other mappings as well.
> Well, depending on the policy being violated, <policy-violation/> might
> map to 413, 414, 513, or even 406/606.

I would find more appropriate 500 than 400 for mapping, also because 
these situations are more server related. 500 is the most generic error 
code in sip from my point of view -- is used even for SIP message 
related situations, such as CSeq value in a request within a dialog is 
lower than previous received one.

The section in the draft seems to be a global mapping, it is no 
difference between types of errors in xmpp -- there can be same error 
tag for streams and stanzas. For example <conflict/> for streams says:

4.9.3.3. conflict


    The server either (1) is closing the existing stream for this entity
    because a new stream has been initiated that conflicts with the
    existing stream, or (2) is refusing a new stream for this entity
    because allowing the new stream would conflict with an existing
    stream (e.g., because the server allows only a certain number of
    connections from the same IP address or allows only one server-to-
    server stream for a given domain pair as a way of helping to ensure
    in-order processing as described under Section 10.1).

The second example above can be associated with 'No more channels' in a 
SIP media server/gateway -- 500 is used in this case (at least Asterisk 
did it :-) ).
>
> Given that <undefined-condition/> is something of a catch-all, it's not
> clear to me what we'd map it to in a consistent way. That would probably
> depend on the application-specific element contained in the error stanza.
>
Indeed, could be application/case-to-case specific mapping, but 
otherwise I think 500 should be the catch-all case in sip side.

Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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    <br>
    <div class="moz-cite-prefix">On 8/20/13 12:07 AM, Peter Saint-Andre
      wrote:<br>
    </div>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">
On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
</pre>
        <blockquote type="cite">
          <pre wrap="">On 8/19/13 2:02 PM, Paul Kyzivat wrote:
</pre>
          <blockquote type="cite">
            <pre wrap="">On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
</pre>
            <blockquote type="cite">
              <pre wrap="">The SIP Parameters Registry has a list of SIP response codes:

<a class="moz-txt-link-freetext" href="http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7">http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7</a>



A number of those are not specified in RFC 3261. Thus the question
arises: for which codes do we need to define mappings? We could define
mappings for all of them, but I wonder if that's advisable. Some of the
additional codes are specified in RFCs that update RFC 3261 (e.g., code
440 from RFC 5393), whereas other codes are specified in "non-core"
RFCs
that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
perhaps make sense to map the "core" codes and not the "non-core"
codes?
</pre>
            </blockquote>
            <pre wrap="">I know this is almost a non-answer, but...

The codes that are defined in other RFCs are there to support features
that are introduced in those RFCs. If there is a mapping of that feature
to XMPP, then there should be a mapping of the code.
</pre>
          </blockquote>
          <pre wrap="">That makes sense.

</pre>
          <blockquote type="cite">
            <pre wrap="">If there is no mapping of the feature, then mapping the code is less
important, but still perhaps useful in some cases. E.g., it could
conceivably show up in a Reason code based on signaling that was never
gatewayed to xmpp. But maybe a default mapping would be sufficient in
those cases.
</pre>
          </blockquote>
          <pre wrap="">I suspect that the default mapping would be fine.

And in general, there might be more art than science here.

I have written some proposed text for this section, but it's fairly long
so my inclination is to submit a revised I-D and then folks can review
what I've written and post to the list with feedback.
</pre>
        </blockquote>
        <pre wrap="">Looking at the latest draft, it seems to bring confusion on mapping from
xmpp-to-sip on 400 code. I read this thread and kind of understood that
any not-obvious-mapping similar-to-a-4xx case would be handled same as
400 in SIP, but typically the 400 response is given for a malformed sip
requests (like broken grammar).
</pre>
      </blockquote>
      <pre wrap="">
So your suggestion is that we avoid mapping XMPP error conditions to SIP
400 unless the XMPP condition is caused by a malformed request?</pre>
    </blockquote>
    Yes, &lt;bad-request/&gt; qualifies for that. Otherwise, could
    create confusion for implementers of sip endpoints - from sip rfc:<br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
    </blockquote>
    21.4.1 400 Bad Request<br>
    <br>
    <br>
    &nbsp;&nbsp; The request could not be understood due to malformed syntax.&nbsp; The<br>
    &nbsp;&nbsp; Reason-Phrase SHOULD identify the syntax problem in more detail,
    for<br>
    &nbsp;&nbsp; example, "Missing Call-ID header field".<br>
    <br>
    <br>
    <br>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">

</pre>
      <blockquote type="cite">
        <pre wrap="">After quick read, &lt;conflict/&gt; would map more suggestively to 403
Forbidden. 
</pre>
      </blockquote>
      <pre wrap="">
As I said, it's perhaps more art than science. But mapping &lt;conflict/&gt;
to 403 seems reasonable if we accept your suggestion about not mapping
to 400 unless we have a good reason to do so.

</pre>
      <blockquote type="cite">
        <pre wrap="">Maybe &lt;policy-violation&gt; and &lt;undefined-condition/&gt; could get
other mappings as well.
</pre>
      </blockquote>
      <pre wrap="">
Well, depending on the policy being violated, &lt;policy-violation/&gt; might
map to 413, 414, 513, or even 406/606.</pre>
    </blockquote>
    <br>
    I would find more appropriate 500 than 400 for mapping, also because
    these situations are more server related. 500 is the most generic
    error code in sip from my point of view -- is used even for SIP
    message related situations, such as CSeq value in a request within a
    dialog is lower than previous received one.<br>
    <br>
    The section in the draft seems to be a global mapping, it is no
    difference between types of errors in xmpp -- there can be same
    error tag for streams and stanzas. For example &lt;conflict/&gt; for
    streams says:<br>
    <br>
    4.9.3.3. conflict<br>
    <br>
    <br>
    &nbsp;&nbsp; The server either (1) is closing the existing stream for this
    entity<br>
    &nbsp;&nbsp; because a new stream has been initiated that conflicts with the<br>
    &nbsp;&nbsp; existing stream, or (2) is refusing a new stream for this entity<br>
    &nbsp;&nbsp; because allowing the new stream would conflict with an existing<br>
    &nbsp;&nbsp; stream (e.g., because the server allows only a certain number of<br>
    &nbsp;&nbsp; connections from the same IP address or allows only one
    server-to-<br>
    &nbsp;&nbsp; server stream for a given domain pair as a way of helping to
    ensure<br>
    &nbsp;&nbsp; in-order processing as described under Section 10.1).<br>
    <br>
    The second example above can be associated with 'No more channels'
    in a SIP media server/gateway -- 500 is used in this case (at least
    Asterisk did it :-) ).<br>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">

Given that &lt;undefined-condition/&gt; is something of a catch-all, it's not
clear to me what we'd map it to in a consistent way. That would probably
depend on the application-specific element contained in the error stanza.

</pre>
    </blockquote>
    Indeed, could be application/case-to-case specific mapping, but
    otherwise I think 500 should be the catch-all case in sip side.<br>
    <br>
    Daniel<br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
</pre>
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20 aug 2013 kl. 10:43 skrev Daniel-Constantin Mierla =
<miconda@gmail.com>:

>=20
> On 8/20/13 12:07 AM, Peter Saint-Andre wrote:
>> On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
>>> On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
>>>> On 8/19/13 2:02 PM, Paul Kyzivat wrote:
>>>>> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>>>>>> The SIP Parameters Registry has a list of SIP response codes:
>>>>>>=20
>>>>>> =
http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-para=
meters-7
>>>>>>=20
>>>>>>=20
>>>>>>=20
>>>>>> A number of those are not specified in RFC 3261. Thus the =
question
>>>>>> arises: for which codes do we need to define mappings? We could =
define
>>>>>> mappings for all of them, but I wonder if that's advisable. Some =
of the
>>>>>> additional codes are specified in RFCs that update RFC 3261 =
(e.g., code
>>>>>> 440 from RFC 5393), whereas other codes are specified in =
"non-core"
>>>>>> RFCs
>>>>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would =
it
>>>>>> perhaps make sense to map the "core" codes and not the "non-core"
>>>>>> codes?
>>>>> I know this is almost a non-answer, but...
>>>>>=20
>>>>> The codes that are defined in other RFCs are there to support =
features
>>>>> that are introduced in those RFCs. If there is a mapping of that =
feature
>>>>> to XMPP, then there should be a mapping of the code.
>>>> That makes sense.
>>>>=20
>>>>> If there is no mapping of the feature, then mapping the code is =
less
>>>>> important, but still perhaps useful in some cases. E.g., it could
>>>>> conceivably show up in a Reason code based on signaling that was =
never
>>>>> gatewayed to xmpp. But maybe a default mapping would be sufficient =
in
>>>>> those cases.
>>>> I suspect that the default mapping would be fine.
>>>>=20
>>>> And in general, there might be more art than science here.
>>>>=20
>>>> I have written some proposed text for this section, but it's fairly =
long
>>>> so my inclination is to submit a revised I-D and then folks can =
review
>>>> what I've written and post to the list with feedback.
>>> Looking at the latest draft, it seems to bring confusion on mapping =
from
>>> xmpp-to-sip on 400 code. I read this thread and kind of understood =
that
>>> any not-obvious-mapping similar-to-a-4xx case would be handled same =
as
>>> 400 in SIP, but typically the 400 response is given for a malformed =
sip
>>> requests (like broken grammar).
>> So your suggestion is that we avoid mapping XMPP error conditions to =
SIP
>> 400 unless the XMPP condition is caused by a malformed request?
> Yes, <bad-request/> qualifies for that. Otherwise, could create =
confusion for implementers of sip endpoints - from sip rfc:
> 21.4.1 400 Bad Request
>=20
>=20
>    The request could not be understood due to malformed syntax.  The
>    Reason-Phrase SHOULD identify the syntax problem in more detail, =
for
>    example, "Missing Call-ID header field".
>=20
>=20
>=20
>>=20
>>> After quick read, <conflict/> would map more suggestively to 403
>>> Forbidden.=20
>> As I said, it's perhaps more art than science. But mapping =
<conflict/>
>> to 403 seems reasonable if we accept your suggestion about not =
mapping
>> to 400 unless we have a good reason to do so.
>>=20
>>> Maybe <policy-violation> and <undefined-condition/> could get
>>> other mappings as well.
>> Well, depending on the policy being violated, <policy-violation/> =
might
>> map to 413, 414, 513, or even 406/606.
>=20
> I would find more appropriate 500 than 400 for mapping, also because =
these situations are more server related. 500 is the most generic error =
code in sip from my point of view -- is used even for SIP message =
related situations, such as CSeq value in a request within a dialog is =
lower than previous received one.
>=20
> The section in the draft seems to be a global mapping, it is no =
difference between types of errors in xmpp -- there can be same error =
tag for streams and stanzas. For example <conflict/> for streams says:
>=20
> 4.9.3.3. conflict
>=20
>=20
>    The server either (1) is closing the existing stream for this =
entity
>    because a new stream has been initiated that conflicts with the
>    existing stream, or (2) is refusing a new stream for this entity
>    because allowing the new stream would conflict with an existing
>    stream (e.g., because the server allows only a certain number of
>    connections from the same IP address or allows only one server-to-
>    server stream for a given domain pair as a way of helping to ensure
>    in-order processing as described under Section 10.1).
>=20
> The second example above can be associated with 'No more channels' in =
a SIP media server/gateway -- 500 is used in this case (at least =
Asterisk did it :-) ).
>>=20
>> Given that <undefined-condition/> is something of a catch-all, it's =
not
>> clear to me what we'd map it to in a consistent way. That would =
probably
>> depend on the application-specific element contained in the error =
stanza.
>>=20
> Indeed, could be application/case-to-case specific mapping, but =
otherwise I think 500 should be the catch-all case in sip side.

Not really.

5xx codes indicates mostly server errors, meaning that the request may =
succeed in another server.
6xx and 4xx errors indicate something wrong related to the request, no =
point in trying elsewhere. RFC 3261:
21.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (for example, adding appropriate authorization).
   However, the same request to a different server might be successful.

21.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred.


I agree that when we have a specific error condition in XMPP we should =
map it properly to an error code in SIP, but we also need some generic =
mapping. Like for the Kamailio-specific 4xx  error codes that only =
Kamailio understands :-)

Section 8.1.3.2 of RFC 3261 clarifies:

8.1.3.2 Unrecognized Responses

   A UAC MUST treat any final response it does not recognize as being
   equivalent to the x00 response code of that class, and MUST be able
   to process the x00 response code for all classes.  For example, if a
   UAC receives an unrecognized response code of 431, it can safely
   assume that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code.  A
   UAC MUST treat any provisional response different than 100 that it
   does not recognize as 183 (Session Progress).  A UAC MUST be able to
   process 100 and 183 responses.

It doesn't mean we should use 400,500,600 as generic codes, but treat =
unknown
the same way as we treat the even ones. If we receive 679, we should act =
as if
we received 600.=20

Does jabber separate "server errors" and "request errors"?

/O



--Apple-Mail=_BF76114B-5282-417D-AB2E-5573546C0C41
Content-Transfer-Encoding: 7bit
Content-Type: text/html;
	charset=iso-8859-1

<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>20 aug 2013 kl. 10:43 skrev Daniel-Constantin Mierla &lt;<a href="mailto:miconda@gmail.com">miconda@gmail.com</a>&gt;:</div><br class="Apple-interchange-newline"><blockquote type="cite">
  
    <meta content="text/html; charset=ISO-8859-1" http-equiv="Content-Type">
  
  <div bgcolor="#FFFFFF" text="#000000">
    <br>
    <div class="moz-cite-prefix">On 8/20/13 12:07 AM, Peter Saint-Andre
      wrote:<br>
    </div>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
</pre>
        <blockquote type="cite">
          <pre wrap="">On 8/19/13 2:02 PM, Paul Kyzivat wrote:
</pre>
          <blockquote type="cite">
            <pre wrap="">On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
</pre>
            <blockquote type="cite">
              <pre wrap="">The SIP Parameters Registry has a list of SIP response codes:

<a class="moz-txt-link-freetext" href="http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7">http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7</a>



A number of those are not specified in RFC 3261. Thus the question
arises: for which codes do we need to define mappings? We could define
mappings for all of them, but I wonder if that's advisable. Some of the
additional codes are specified in RFCs that update RFC 3261 (e.g., code
440 from RFC 5393), whereas other codes are specified in "non-core"
RFCs
that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
perhaps make sense to map the "core" codes and not the "non-core"
codes?
</pre>
            </blockquote>
            <pre wrap="">I know this is almost a non-answer, but...

The codes that are defined in other RFCs are there to support features
that are introduced in those RFCs. If there is a mapping of that feature
to XMPP, then there should be a mapping of the code.
</pre>
          </blockquote>
          <pre wrap="">That makes sense.

</pre>
          <blockquote type="cite">
            <pre wrap="">If there is no mapping of the feature, then mapping the code is less
important, but still perhaps useful in some cases. E.g., it could
conceivably show up in a Reason code based on signaling that was never
gatewayed to xmpp. But maybe a default mapping would be sufficient in
those cases.
</pre>
          </blockquote>
          <pre wrap="">I suspect that the default mapping would be fine.

And in general, there might be more art than science here.

I have written some proposed text for this section, but it's fairly long
so my inclination is to submit a revised I-D and then folks can review
what I've written and post to the list with feedback.
</pre>
        </blockquote>
        <pre wrap="">Looking at the latest draft, it seems to bring confusion on mapping from
xmpp-to-sip on 400 code. I read this thread and kind of understood that
any not-obvious-mapping similar-to-a-4xx case would be handled same as
400 in SIP, but typically the 400 response is given for a malformed sip
requests (like broken grammar).
</pre>
      </blockquote>
      <pre wrap="">So your suggestion is that we avoid mapping XMPP error conditions to SIP
400 unless the XMPP condition is caused by a malformed request?</pre>
    </blockquote>
    Yes, &lt;bad-request/&gt; qualifies for that. Otherwise, could
    create confusion for implementers of sip endpoints - from sip rfc:<br>
    <blockquote type="cite">
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
    </blockquote>
    21.4.1 400 Bad Request<br>
    <br>
    <br>
    &nbsp;&nbsp; The request could not be understood due to malformed syntax.&nbsp; The<br>
    &nbsp;&nbsp; Reason-Phrase SHOULD identify the syntax problem in more detail,
    for<br>
    &nbsp;&nbsp; example, "Missing Call-ID header field".<br>
    <br>
    <br>
    <br>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <pre wrap="">After quick read, &lt;conflict/&gt; would map more suggestively to 403
Forbidden. 
</pre>
      </blockquote>
      <pre wrap="">As I said, it's perhaps more art than science. But mapping &lt;conflict/&gt;
to 403 seems reasonable if we accept your suggestion about not mapping
to 400 unless we have a good reason to do so.

</pre>
      <blockquote type="cite">
        <pre wrap="">Maybe &lt;policy-violation&gt; and &lt;undefined-condition/&gt; could get
other mappings as well.
</pre>
      </blockquote>
      <pre wrap="">Well, depending on the policy being violated, &lt;policy-violation/&gt; might
map to 413, 414, 513, or even 406/606.</pre>
    </blockquote>
    <br>
    I would find more appropriate 500 than 400 for mapping, also because
    these situations are more server related. 500 is the most generic
    error code in sip from my point of view -- is used even for SIP
    message related situations, such as CSeq value in a request within a
    dialog is lower than previous received one.<br>
    <br>
    The section in the draft seems to be a global mapping, it is no
    difference between types of errors in xmpp -- there can be same
    error tag for streams and stanzas. For example &lt;conflict/&gt; for
    streams says:<br>
    <br>
    4.9.3.3. conflict<br>
    <br>
    <br>
    &nbsp;&nbsp; The server either (1) is closing the existing stream for this
    entity<br>
    &nbsp;&nbsp; because a new stream has been initiated that conflicts with the<br>
    &nbsp;&nbsp; existing stream, or (2) is refusing a new stream for this entity<br>
    &nbsp;&nbsp; because allowing the new stream would conflict with an existing<br>
    &nbsp;&nbsp; stream (e.g., because the server allows only a certain number of<br>
    &nbsp;&nbsp; connections from the same IP address or allows only one
    server-to-<br>
    &nbsp;&nbsp; server stream for a given domain pair as a way of helping to
    ensure<br>
    &nbsp;&nbsp; in-order processing as described under Section 10.1).<br>
    <br>
    The second example above can be associated with 'No more channels'
    in a SIP media server/gateway -- 500 is used in this case (at least
    Asterisk did it :-) ).<br>
    <blockquote cite="mid:5212970D.6070608@stpeter.im" type="cite">
      <pre wrap="">
Given that &lt;undefined-condition/&gt; is something of a catch-all, it's not
clear to me what we'd map it to in a consistent way. That would probably
depend on the application-specific element contained in the error stanza.

</pre>
    </blockquote>
    Indeed, could be application/case-to-case specific mapping, but
    otherwise I think 500 should be the catch-all case in sip side.<br></div></blockquote></div><br><div>Not really.</div><div><br></div><div>5xx codes indicates mostly server errors, meaning that the request may succeed in another server.</div><div>6xx and 4xx errors indicate something wrong related to the request, no point in trying elsewhere. RFC 3261:</div><div><pre style="word-wrap: break-word; white-space: pre-wrap; ">21.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (for example, adding appropriate authorization).
   However, the same request to a different server might be successful.</pre><div><br></div></div><div><pre style="word-wrap: break-word; white-space: pre-wrap; ">21.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred.</pre><div><br></div></div><div><br></div><div>I agree that when we have a specific error condition in XMPP we should map it properly to an error code in SIP, but we also need some generic mapping. Like for the Kamailio-specific 4xx &nbsp;error codes that only Kamailio understands :-)</div><div><br></div><div>Section 8.1.3.2 of RFC 3261 clarifies:</div><div><br></div><div><pre style="word-wrap: break-word; white-space: pre-wrap; ">8.1.3.2 Unrecognized Responses

   A UAC MUST treat any final response it does not recognize as being
   equivalent to the x00 response code of that class, and MUST be able
   to process the x00 response code for all classes.  For example, if a
   UAC receives an unrecognized response code of 431, it can safely
   assume that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code.  A
   UAC MUST treat any provisional response different than 100 that it
   does not recognize as 183 (Session Progress).  A UAC MUST be able to
   process 100 and 183 responses.

</pre><pre style="word-wrap: break-word; white-space: pre-wrap; ">It doesn't mean we should use 400,500,600 as generic codes, but treat unknown</pre><pre style="word-wrap: break-word; white-space: pre-wrap; ">the same way as we treat the even ones. If we receive 679, we should act as if</pre><pre style="word-wrap: break-word; white-space: pre-wrap; ">we received 600. </pre><pre style="word-wrap: break-word; white-space: pre-wrap; "><br></pre><pre style="word-wrap: break-word; white-space: pre-wrap; ">Does jabber separate "server errors" and "request errors"?</pre><pre style="word-wrap: break-word; white-space: pre-wrap; "><br></pre><pre style="word-wrap: break-word; white-space: pre-wrap; ">/O</pre></div><div><br></div><div><br></div></body></html>
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On 8/20/13 10:55 AM, Olle E. Johansson wrote:
>
> 20 aug 2013 kl. 10:43 skrev Daniel-Constantin Mierla 
> <miconda@gmail.com <mailto:miconda@gmail.com>>:
>
>>
>> On 8/20/13 12:07 AM, Peter Saint-Andre wrote:
>>> On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
>>>> On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
>>>>> On 8/19/13 2:02 PM, Paul Kyzivat wrote:
>>>>>> On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
>>>>>>> The SIP Parameters Registry has a list of SIP response codes:
>>>>>>>
>>>>>>> http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> A number of those are not specified in RFC 3261. Thus the question
>>>>>>> arises: for which codes do we need to define mappings? We could define
>>>>>>> mappings for all of them, but I wonder if that's advisable. Some of the
>>>>>>> additional codes are specified in RFCs that update RFC 3261 (e.g., code
>>>>>>> 440 from RFC 5393), whereas other codes are specified in "non-core"
>>>>>>> RFCs
>>>>>>> that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
>>>>>>> perhaps make sense to map the "core" codes and not the "non-core"
>>>>>>> codes?
>>>>>> I know this is almost a non-answer, but...
>>>>>>
>>>>>> The codes that are defined in other RFCs are there to support features
>>>>>> that are introduced in those RFCs. If there is a mapping of that feature
>>>>>> to XMPP, then there should be a mapping of the code.
>>>>> That makes sense.
>>>>>
>>>>>> If there is no mapping of the feature, then mapping the code is less
>>>>>> important, but still perhaps useful in some cases. E.g., it could
>>>>>> conceivably show up in a Reason code based on signaling that was never
>>>>>> gatewayed to xmpp. But maybe a default mapping would be sufficient in
>>>>>> those cases.
>>>>> I suspect that the default mapping would be fine.
>>>>>
>>>>> And in general, there might be more art than science here.
>>>>>
>>>>> I have written some proposed text for this section, but it's fairly long
>>>>> so my inclination is to submit a revised I-D and then folks can review
>>>>> what I've written and post to the list with feedback.
>>>> Looking at the latest draft, it seems to bring confusion on mapping from
>>>> xmpp-to-sip on 400 code. I read this thread and kind of understood that
>>>> any not-obvious-mapping similar-to-a-4xx case would be handled same as
>>>> 400 in SIP, but typically the 400 response is given for a malformed sip
>>>> requests (like broken grammar).
>>> So your suggestion is that we avoid mapping XMPP error conditions to SIP
>>> 400 unless the XMPP condition is caused by a malformed request?
>> Yes, <bad-request/> qualifies for that. Otherwise, could create 
>> confusion for implementers of sip endpoints - from sip rfc:
>> 21.4.1 400 Bad Request
>>
>>
>>    The request could not be understood due to malformed syntax.  The
>>    Reason-Phrase SHOULD identify the syntax problem in more detail, for
>>    example, "Missing Call-ID header field".
>>
>>
>>
>>>> After quick read, <conflict/> would map more suggestively to 403
>>>> Forbidden.
>>> As I said, it's perhaps more art than science. But mapping <conflict/>
>>> to 403 seems reasonable if we accept your suggestion about not mapping
>>> to 400 unless we have a good reason to do so.
>>>
>>>> Maybe <policy-violation> and <undefined-condition/> could get
>>>> other mappings as well.
>>> Well, depending on the policy being violated, <policy-violation/> might
>>> map to 413, 414, 513, or even 406/606.
>>
>> I would find more appropriate 500 than 400 for mapping, also because 
>> these situations are more server related. 500 is the most generic 
>> error code in sip from my point of view -- is used even for SIP 
>> message related situations, such as CSeq value in a request within a 
>> dialog is lower than previous received one.
>>
>> The section in the draft seems to be a global mapping, it is no 
>> difference between types of errors in xmpp -- there can be same error 
>> tag for streams and stanzas. For example <conflict/> for streams says:
>>
>> 4.9.3.3. conflict
>>
>>
>>    The server either (1) is closing the existing stream for this entity
>>    because a new stream has been initiated that conflicts with the
>>    existing stream, or (2) is refusing a new stream for this entity
>>    because allowing the new stream would conflict with an existing
>>    stream (e.g., because the server allows only a certain number of
>>    connections from the same IP address or allows only one server-to-
>>    server stream for a given domain pair as a way of helping to ensure
>>    in-order processing as described under Section 10.1).
>>
>> The second example above can be associated with 'No more channels' in 
>> a SIP media server/gateway -- 500 is used in this case (at least 
>> Asterisk did it :-) ).
>>> Given that <undefined-condition/> is something of a catch-all, it's not
>>> clear to me what we'd map it to in a consistent way. That would probably
>>> depend on the application-specific element contained in the error stanza.
>>>
>> Indeed, could be application/case-to-case specific mapping, but 
>> otherwise I think 500 should be the catch-all case in sip side.
>
> Not really.
>
> 5xx codes indicates mostly server errors, meaning that the request may 
> succeed in another server.
> 6xx and 4xx errors indicate something wrong related to the request, no 
> point in trying elsewhere. RFC 3261:
6xx are indeed global, but many of 4xx can succeed on retry or other 
place (e.g., calling to germany can get 404 on a pstn gateway with no 
route to this country but can succeed on other gateway with such route).

Anyhow, it is what I understand from the xmpp error tags I listed above 
(being currently mapped to 400). For example, number of streams 
(4.9.3.3. conflict) is a server related option, has nothing to do with 
the format of the request or processing the request per se -- it is not 
accepted because local server policy/available resources.


> 21.4 Request Failure 4xx
>
>     4xx responses are definite failure responses from a particular
>     server.  The client SHOULD NOT retry the same request without
>     modification (for example, adding appropriate authorization).
>     However, the same request to a different server might be successful.
>
> 21.5 Server Failure 5xx
>
>     5xx responses are failure responses given when a server itself has
>     erred.
>
>
> I agree that when we have a specific error condition in XMPP we should 
> map it properly to an error code in SIP, but we also need some generic 
> mapping. Like for the Kamailio-specific 4xx  error codes that only 
> Kamailio understands :-)
>
> Section 8.1.3.2 of RFC 3261 clarifies:
>
> 8.1.3.2 Unrecognized Responses
>
>     A UAC MUST treat any final response it does not recognize as being
>     equivalent to the x00 response code of that class, and MUST be able
>     to process the x00 response code for all classes.  For example, if a
>     UAC receives an unrecognized response code of 431, it can safely
>     assume that there was something wrong with its request and treat the
>     response as if it had received a 400 (Bad Request) response code.  A
>     UAC MUST treat any provisional response different than 100 that it
>     does not recognize as 183 (Session Progress).  A UAC MUST be able to
>     process 100 and 183 responses.
>
> It doesn't mean we should use 400,500,600 as generic codes, but treat unknown
> the same way as we treat the even ones. If we receive 679, we should act as if
> we received 600.

I would interpret that in more like generic way:
- unknown >=300 is negative response, closing transaction/dialog
- unknown 4xx or 5xx means a retry could work (at same place or 
different one)
- unknown 6xx means giving up completely

So it might be better to define new 'generic' reply codes for each class 
and let the sip endpoint handle them as stated in rfc (and pasted above).

Daniel

> Does jabber separate "server errors" and "request errors"?
> /O
>
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


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    <br>
    <div class="moz-cite-prefix">On 8/20/13 10:55 AM, Olle E. Johansson
      wrote:<br>
    </div>
    <blockquote
      cite="mid:A35C463B-B8E7-4ECC-B300-BCDAEDC274C7@edvina.net"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <br>
      <div>
        <div>20 aug 2013 kl. 10:43 skrev Daniel-Constantin Mierla &lt;<a
            moz-do-not-send="true" href="mailto:miconda@gmail.com">miconda@gmail.com</a>&gt;:</div>
        <br class="Apple-interchange-newline">
        <blockquote type="cite">
          <meta content="text/html; charset=ISO-8859-1"
            http-equiv="Content-Type">
          <div bgcolor="#FFFFFF" text="#000000"> <br>
            <div class="moz-cite-prefix">On 8/20/13 12:07 AM, Peter
              Saint-Andre wrote:<br>
            </div>
            <blockquote cite="mid:5212970D.6070608@stpeter.im"
              type="cite">
              <pre wrap="">On 8/19/13 3:44 PM, Daniel-Constantin Mierla wrote:
</pre>
              <blockquote type="cite">
                <pre wrap="">On 8/19/13 10:23 PM, Peter Saint-Andre wrote:
</pre>
                <blockquote type="cite">
                  <pre wrap="">On 8/19/13 2:02 PM, Paul Kyzivat wrote:
</pre>
                  <blockquote type="cite">
                    <pre wrap="">On 8/16/13 10:28 PM, Peter Saint-Andre wrote:
</pre>
                    <blockquote type="cite">
                      <pre wrap="">The SIP Parameters Registry has a list of SIP response codes:

<a moz-do-not-send="true" class="moz-txt-link-freetext" href="http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7">http://www.iana.org/assignments/sip-parameters/sip-parameters.xml#sip-parameters-7</a>



A number of those are not specified in RFC 3261. Thus the question
arises: for which codes do we need to define mappings? We could define
mappings for all of them, but I wonder if that's advisable. Some of the
additional codes are specified in RFCs that update RFC 3261 (e.g., code
440 from RFC 5393), whereas other codes are specified in "non-core"
RFCs
that don't update RFC 3261 (e.g., code 470 from RFC 5360). Would it
perhaps make sense to map the "core" codes and not the "non-core"
codes?
</pre>
                    </blockquote>
                    <pre wrap="">I know this is almost a non-answer, but...

The codes that are defined in other RFCs are there to support features
that are introduced in those RFCs. If there is a mapping of that feature
to XMPP, then there should be a mapping of the code.
</pre>
                  </blockquote>
                  <pre wrap="">That makes sense.

</pre>
                  <blockquote type="cite">
                    <pre wrap="">If there is no mapping of the feature, then mapping the code is less
important, but still perhaps useful in some cases. E.g., it could
conceivably show up in a Reason code based on signaling that was never
gatewayed to xmpp. But maybe a default mapping would be sufficient in
those cases.
</pre>
                  </blockquote>
                  <pre wrap="">I suspect that the default mapping would be fine.

And in general, there might be more art than science here.

I have written some proposed text for this section, but it's fairly long
so my inclination is to submit a revised I-D and then folks can review
what I've written and post to the list with feedback.
</pre>
                </blockquote>
                <pre wrap="">Looking at the latest draft, it seems to bring confusion on mapping from
xmpp-to-sip on 400 code. I read this thread and kind of understood that
any not-obvious-mapping similar-to-a-4xx case would be handled same as
400 in SIP, but typically the 400 response is given for a malformed sip
requests (like broken grammar).
</pre>
              </blockquote>
              <pre wrap="">So your suggestion is that we avoid mapping XMPP error conditions to SIP
400 unless the XMPP condition is caused by a malformed request?</pre>
            </blockquote>
            Yes, &lt;bad-request/&gt; qualifies for that. Otherwise,
            could create confusion for implementers of sip endpoints -
            from sip rfc:<br>
            <blockquote type="cite">
              <meta http-equiv="content-type" content="text/html;
                charset=ISO-8859-1">
            </blockquote>
            21.4.1 400 Bad Request<br>
            <br>
            <br>
            &nbsp;&nbsp; The request could not be understood due to malformed
            syntax.&nbsp; The<br>
            &nbsp;&nbsp; Reason-Phrase SHOULD identify the syntax problem in more
            detail, for<br>
            &nbsp;&nbsp; example, "Missing Call-ID header field".<br>
            <br>
            <br>
            <br>
            <blockquote cite="mid:5212970D.6070608@stpeter.im"
              type="cite">
              <blockquote type="cite">
                <pre wrap="">After quick read, &lt;conflict/&gt; would map more suggestively to 403
Forbidden. 
</pre>
              </blockquote>
              <pre wrap="">As I said, it's perhaps more art than science. But mapping &lt;conflict/&gt;
to 403 seems reasonable if we accept your suggestion about not mapping
to 400 unless we have a good reason to do so.

</pre>
              <blockquote type="cite">
                <pre wrap="">Maybe &lt;policy-violation&gt; and &lt;undefined-condition/&gt; could get
other mappings as well.
</pre>
              </blockquote>
              <pre wrap="">Well, depending on the policy being violated, &lt;policy-violation/&gt; might
map to 413, 414, 513, or even 406/606.</pre>
            </blockquote>
            <br>
            I would find more appropriate 500 than 400 for mapping, also
            because these situations are more server related. 500 is the
            most generic error code in sip from my point of view -- is
            used even for SIP message related situations, such as CSeq
            value in a request within a dialog is lower than previous
            received one.<br>
            <br>
            The section in the draft seems to be a global mapping, it is
            no difference between types of errors in xmpp -- there can
            be same error tag for streams and stanzas. For example
            &lt;conflict/&gt; for streams says:<br>
            <br>
            4.9.3.3. conflict<br>
            <br>
            <br>
            &nbsp;&nbsp; The server either (1) is closing the existing stream for
            this entity<br>
            &nbsp;&nbsp; because a new stream has been initiated that conflicts
            with the<br>
            &nbsp;&nbsp; existing stream, or (2) is refusing a new stream for this
            entity<br>
            &nbsp;&nbsp; because allowing the new stream would conflict with an
            existing<br>
            &nbsp;&nbsp; stream (e.g., because the server allows only a certain
            number of<br>
            &nbsp;&nbsp; connections from the same IP address or allows only one
            server-to-<br>
            &nbsp;&nbsp; server stream for a given domain pair as a way of helping
            to ensure<br>
            &nbsp;&nbsp; in-order processing as described under Section 10.1).<br>
            <br>
            The second example above can be associated with 'No more
            channels' in a SIP media server/gateway -- 500 is used in
            this case (at least Asterisk did it :-) ).<br>
            <blockquote cite="mid:5212970D.6070608@stpeter.im"
              type="cite">
              <pre wrap="">Given that &lt;undefined-condition/&gt; is something of a catch-all, it's not
clear to me what we'd map it to in a consistent way. That would probably
depend on the application-specific element contained in the error stanza.

</pre>
            </blockquote>
            Indeed, could be application/case-to-case specific mapping,
            but otherwise I think 500 should be the catch-all case in
            sip side.<br>
          </div>
        </blockquote>
      </div>
      <br>
      <div>Not really.</div>
      <div><br>
      </div>
      <div>5xx codes indicates mostly server errors, meaning that the
        request may succeed in another server.</div>
      <div>6xx and 4xx errors indicate something wrong related to the
        request, no point in trying elsewhere. RFC 3261:</div>
    </blockquote>
    6xx are indeed global, but many of 4xx can succeed on retry or other
    place (e.g., calling to germany can get 404 on a pstn gateway with
    no route to this country but can succeed on other gateway with such
    route).<br>
    <br>
    Anyhow, it is what I understand from the xmpp error tags I listed
    above (being currently mapped to 400). For example, number of
    streams (4.9.3.3. conflict) is a server related option, has nothing
    to do with the format of the request or processing the request per
    se -- it is not accepted because local server policy/available
    resources.<br>
    <br>
    <br>
    <blockquote
      cite="mid:A35C463B-B8E7-4ECC-B300-BCDAEDC274C7@edvina.net"
      type="cite">
      <div>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">21.4 Request Failure 4xx

   4xx responses are definite failure responses from a particular
   server.  The client SHOULD NOT retry the same request without
   modification (for example, adding appropriate authorization).
   However, the same request to a different server might be successful.</pre>
        <div><br>
        </div>
      </div>
      <div>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">21.5 Server Failure 5xx

   5xx responses are failure responses given when a server itself has
   erred.</pre>
        <div><br>
        </div>
      </div>
      <div><br>
      </div>
      <div>I agree that when we have a specific error condition in XMPP
        we should map it properly to an error code in SIP, but we also
        need some generic mapping. Like for the Kamailio-specific 4xx
        &nbsp;error codes that only Kamailio understands :-)</div>
      <div><br>
      </div>
      <div>Section 8.1.3.2 of RFC 3261 clarifies:</div>
      <div><br>
      </div>
      <div>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">8.1.3.2 Unrecognized Responses

   A UAC MUST treat any final response it does not recognize as being
   equivalent to the x00 response code of that class, and MUST be able
   to process the x00 response code for all classes.  For example, if a
   UAC receives an unrecognized response code of 431, it can safely
   assume that there was something wrong with its request and treat the
   response as if it had received a 400 (Bad Request) response code.  A
   UAC MUST treat any provisional response different than 100 that it
   does not recognize as 183 (Session Progress).  A UAC MUST be able to
   process 100 and 183 responses.

</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">It doesn't mean we should use 400,500,600 as generic codes, but treat unknown</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">the same way as we treat the even ones. If we receive 679, we should act as if</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">we received 600. </pre>
      </div>
    </blockquote>
    <br>
    I would interpret that in more like generic way:<br>
    - unknown &gt;=300 is negative response, closing transaction/dialog<br>
    - unknown 4xx or 5xx means a retry could work (at same place or
    different one)<br>
    - unknown 6xx means giving up completely<br>
    <br>
    So it might be better to define new 'generic' reply codes for each
    class and let the sip endpoint handle them as stated in rfc (and
    pasted above).<br>
    <br>
    Daniel<br>
    <br>
    <blockquote
      cite="mid:A35C463B-B8E7-4ECC-B300-BCDAEDC274C7@edvina.net"
      type="cite">
      <div>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">
</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">Does jabber separate "server errors" and "request errors"?</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">
</pre>
        <pre style="word-wrap: break-word; white-space: pre-wrap; ">/O</pre>
      </div>
      <div><br>
      </div>
      <div><br>
      </div>
    </blockquote>
    <br>
    <pre class="moz-signature" cols="72">-- 
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
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Subject: Re: [Stox] Instant Messaging (draft-ietf-stox-im-00) and SIP Contact header field
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On 8/20/13 1:13 AM, Saúl Ibarra Corretgé wrote:
> 
> On Aug 20, 2013, at 5:20 AM, Peter Saint-Andre wrote:
> 
>> On 8/16/13 2:14 AM, Saúl Ibarra Corretgé wrote:
>>>>
>>>> There is nothing to *prevent* use of a gruu in From, but it isn't
>>>> what is normally used.
>>>>
>>>> If you happen to find a gruu in From, then it probably makes sense
>>>> to translate it to a full JID.
>>>>
>>>
>>> I thought it could only be used in the Contact header, but as you
>>> said, RFC5627 doesn't enforce that. So I guess we should mention this
>>> in the -im spec. This could also be used if someone writes a spec to
>>> map XMPP chat messages to SIP MESSAGE in the future.
>>>
>>> Thanks!
>>
>> And in this instance the SIP From header would be added by the gateway,
>> right?
>>
> 
> Yes, by translating the from attribute of the XMPP message stanza. So I think we are good. It would be nice to have this in the examples, that is, no Contact header, and the GRUU in the From.

Yes, that will be added in the next revision of the stox-im I-D. :-)

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Instant Messa=
ging
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-im-01.txt
	Pages           : 11
	Date            : 2013-08-20

Abstract:
   This document defines a bidirectional protocol mapping for the
   exchange of single instant messages between the Session Initiation
   Protocol (SIP) and the Extensible Messaging and Presence Protocol
   (XMPP).


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-im

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-im-01

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-im-01


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


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Subject: Re: [Stox] core: response code mappings
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On 8/20/13 3:14 AM, Daniel-Constantin Mierla wrote:

> So it might be better to define new 'generic' reply codes for each class
> and let the sip endpoint handle them as stated in rfc (and pasted above).

Well, this WG is not chartered to define anything new, so I think that's
a non-starter. :-)

Peter

-- 
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On 8/20/13 12:32 AM, Olle E. Johansson wrote:
> 
> 19 aug 2013 kl. 20:02 skrev Peter Saint-Andre <stpeter@stpeter.im>:
> 
>> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>> 
>>> * Core
>>> 
>>> - Issue: Should something general about forking be defined here
>>> or is it purely a -media issue?
>> 
>> I think this is best left for the stox-media spec.
> Do remember to handle the case where one sip INVITE can get multiple
> 200 OK. It's a corner case, but it does happen. And it's signalling.

That sounds like fun.

>>> - Issue: Add text about Max-Forwards and Max-Breath. Loop
>>> detection in general.
>> 
>> IMHO we could say something like the following in the stox-core
>> spec...
>> 
>> A gateway between SIP and XMPP (in either direction) effectively
>> acts as a SIP proxy server.  Therefore the amplification
>> vulnerability
> b2bua, not a proxy.

Correct. I'll fix that in the next version of stox-core (which now says
"proxy").

> There's a draft in the straw wg mandating b2bua's to support 
> max-forwards and max-breadth

I assume that is:

https://datatracker.ietf.org/doc/draft-ietf-straw-b2bua-loop-detection/

>> described in [RFC5393] can manifest itself, and a gateway SHOULD 
>> implement the loop-detection methods defined in that specification
>> to help mitigate the possibility of amplification attacks.
> 
> Agree. How are these headers handled in XMPP?

We don't have those headers in XMPP because we think that we don't have
multiple hops to worry about.

Peter

-- 
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Subject: Re: [Stox] core: response code mappings
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On 8/21/13 12:07 AM, Peter Saint-Andre wrote:
> On 8/20/13 3:14 AM, Daniel-Constantin Mierla wrote:
>
>> So it might be better to define new 'generic' reply codes for each class
>> and let the sip endpoint handle them as stated in rfc (and pasted above).
> Well, this WG is not chartered to define anything new, so I think that's
> a non-starter. :-)
3-6xx classes are defined, with rules how to handle those codes unknown 
locally. Not sure is better to misuse an existing one versus just using 
a random unallocated one from the appropriate class. Anyhow, coming back 
to the initial opinion, I find 400 mapping to be confusing in some 
situations, as it is defined for malformed sip requests and it is not 
the case. Would be good to find more appropriate codes.

Daniel

-- 
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On 8/21/13 12:27 AM, Peter Saint-Andre wrote:
> On 8/20/13 12:32 AM, Olle E. Johansson wrote:
>> 19 aug 2013 kl. 20:02 skrev Peter Saint-Andre <stpeter@stpeter.im>:
>>
>>> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>>>
>>>> * Core
>>>>
>>>> - Issue: Should something general about forking be defined here
>>>> or is it purely a -media issue?
>>> I think this is best left for the stox-media spec.
>> Do remember to handle the case where one sip INVITE can get multiple
>> 200 OK. It's a corner case, but it does happen. And it's signalling.
> That sounds like fun.
This should be caught by the gateway, handled locally, like any SIP 
endpoint UA (and how is happening for PSTN gateways). Only one 200ok 
should be propagated to xmpp/jingle side. Which one? A matter of the 
implementation, most I have seen they go with the first response 
received, for the others they send ACK followed quickly by BYE (I mean, 
the good SIP UA implementations, some just crash in such cases :-) ).
Daniel

-- 
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From stpeter@stpeter.im  Tue Aug 20 15:48:52 2013
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On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:

> * Chat
> 
> - Issue: Mapping between XMPP ChatStates (XEP-0085) and isComposing event (RFC 3994).
> - Issue: Mapping between message receipts (XEP-0184) and MSRP REPORT chunks (RFC 4975).
> - Consider if the above two are included or possibly split into a forthcoming advanced-chat draft.

The sense of the room that I perceived was to map the first but not the
second, since I don't think there's a critical mass of XEP-0184
implementations out there in the world. However, I'd be open to defining
the receipt/chunk mapping in more of an informational way in the
stox-chat spec.

What do other folks think?

> - Issue: Draft says gateway determines MSRP is supported before sending INVITE. How?

SIP OPTIONS...?

Peter

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Subject: Re: [Stox] core issues
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On 8/20/13 6:41 PM, Daniel-Constantin Mierla wrote:
>
> On 8/21/13 12:27 AM, Peter Saint-Andre wrote:
>> On 8/20/13 12:32 AM, Olle E. Johansson wrote:
>>> 19 aug 2013 kl. 20:02 skrev Peter Saint-Andre <stpeter@stpeter.im>:
>>>
>>>> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>>>>
>>>>> * Core
>>>>>
>>>>> - Issue: Should something general about forking be defined here
>>>>> or is it purely a -media issue?
>>>> I think this is best left for the stox-media spec.
>>> Do remember to handle the case where one sip INVITE can get multiple
>>> 200 OK. It's a corner case, but it does happen. And it's signalling.
>> That sounds like fun.
> This should be caught by the gateway, handled locally, like any SIP
> endpoint UA (and how is happening for PSTN gateways). Only one 200ok
> should be propagated to xmpp/jingle side. Which one? A matter of the
> implementation, most I have seen they go with the first response
> received, for the others they send ACK followed quickly by BYE (I mean,
> the good SIP UA implementations, some just crash in such cases :-) ).
> Daniel

You can make it sound easy by ignoring the details, but at the expense 
of quality of implementation.

The problem is that you may get multiple early dialogs, with media. At 
that time you don't know which one will answer first. So what do you do 
with the media while awaiting a 2xx?

You can't "pick one" without the risk that it is the wrong one.

You can "pick one" and switch it if it turns out to be the wrong one. 
But unless you relay the media the xmpp side will see a change, and so 
there must be some plan for how that is handled.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Tue Aug 20 16:52:24 2013
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Subject: Re: [Stox] chat issues
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On 8/20/13 6:48 PM, Peter Saint-Andre wrote:
> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>
>> * Chat
>>
>> - Issue: Mapping between XMPP ChatStates (XEP-0085) and isComposing event (RFC 3994).
>> - Issue: Mapping between message receipts (XEP-0184) and MSRP REPORT chunks (RFC 4975).
>> - Consider if the above two are included or possibly split into a forthcoming advanced-chat draft.
>
> The sense of the room that I perceived was to map the first but not the
> second, since I don't think there's a critical mass of XEP-0184
> implementations out there in the world. However, I'd be open to defining
> the receipt/chunk mapping in more of an informational way in the
> stox-chat spec.
>
> What do other folks think?
>
>> - Issue: Draft says gateway determines MSRP is supported before sending INVITE. How?
>
> SIP OPTIONS...?

Don't go there - its a bad plan.
There is no reasonable way to know in advance.

	Thanks,
	Paul


From stpeter@stpeter.im  Tue Aug 20 18:15:25 2013
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On 8/20/13 5:52 PM, Paul Kyzivat wrote:
> On 8/20/13 6:48 PM, Peter Saint-Andre wrote:
>> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>>
>>> - Issue: Draft says gateway determines MSRP is supported before
>>> sending INVITE. How?
>>
>> SIP OPTIONS...?
> 
> Don't go there - its a bad plan.
> There is no reasonable way to know in advance.

Yeah, I know. It's a shame, but it's not something we can fix here...

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From oej@edvina.net  Tue Aug 20 23:52:54 2013
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21 aug 2013 kl. 01:52 skrev Paul Kyzivat <pkyzivat@alum.mit.edu>:

>>> - Issue: Draft says gateway determines MSRP is supported before =
sending INVITE. How?
>>=20
>> SIP OPTIONS...?
>=20
> Don't go there - its a bad plan.
> There is no reasonable way to know in advance.

Agree.=20

/O=

From miconda@gmail.com  Wed Aug 21 00:17:57 2013
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Subject: Re: [Stox] core issues
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On 8/21/13 1:50 AM, Paul Kyzivat wrote:
> On 8/20/13 6:41 PM, Daniel-Constantin Mierla wrote:
>>
>> On 8/21/13 12:27 AM, Peter Saint-Andre wrote:
>>> On 8/20/13 12:32 AM, Olle E. Johansson wrote:
>>>> 19 aug 2013 kl. 20:02 skrev Peter Saint-Andre <stpeter@stpeter.im>:
>>>>
>>>>> On 8/14/13 1:19 PM, Markus.Isomaki@nokia.com wrote:
>>>>>
>>>>>> * Core
>>>>>>
>>>>>> - Issue: Should something general about forking be defined here
>>>>>> or is it purely a -media issue?
>>>>> I think this is best left for the stox-media spec.
>>>> Do remember to handle the case where one sip INVITE can get multiple
>>>> 200 OK. It's a corner case, but it does happen. And it's signalling.
>>> That sounds like fun.
>> This should be caught by the gateway, handled locally, like any SIP
>> endpoint UA (and how is happening for PSTN gateways). Only one 200ok
>> should be propagated to xmpp/jingle side. Which one? A matter of the
>> implementation, most I have seen they go with the first response
>> received, for the others they send ACK followed quickly by BYE (I mean,
>> the good SIP UA implementations, some just crash in such cases :-) ).
>> Daniel
>
> You can make it sound easy by ignoring the details, but at the expense 
> of quality of implementation.
>
> The problem is that you may get multiple early dialogs, with media. At 
> that time you don't know which one will answer first. So what do you 
> do with the media while awaiting a 2xx?
>
> You can't "pick one" without the risk that it is the wrong one.
>
> You can "pick one" and switch it if it turns out to be the wrong one. 
> But unless you relay the media the xmpp side will see a change, and so 
> there must be some plan for how that is handled.
My comments were for the remark related to multiple 200ok, I haven't 
said sending BYE for additional early dialogs.

For multiple early dialogs, the gateway has to maintain all of them 
until a 200ok. It's also implementation decision which of early media 
streams is sent to xmpp/jingle (again, it looks like it's mostly the 
first one from experiences out there).

Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From pkyzivat@alum.mit.edu  Wed Aug 21 12:03:32 2013
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [Stox] core issues
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On 8/21/13 3:17 AM, Daniel-Constantin Mierla wrote:
>
> On 8/21/13 1:50 AM, Paul Kyzivat wrote:

>> You can make it sound easy by ignoring the details, but at the expense
>> of quality of implementation.
>>
>> The problem is that you may get multiple early dialogs, with media. At
>> that time you don't know which one will answer first. So what do you
>> do with the media while awaiting a 2xx?
>>
>> You can't "pick one" without the risk that it is the wrong one.
>>
>> You can "pick one" and switch it if it turns out to be the wrong one.
>> But unless you relay the media the xmpp side will see a change, and so
>> there must be some plan for how that is handled.
> My comments were for the remark related to multiple 200ok, I haven't
> said sending BYE for additional early dialogs.

They are tightly connected issues.

> For multiple early dialogs, the gateway has to maintain all of them
> until a 200ok. It's also implementation decision which of early media
> streams is sent to xmpp/jingle (again, it looks like it's mostly the
> first one from experiences out there).

This is straightforward if the GW terminates/relays media.
But that is a cost.

A signaling-only GW can't easily or thoroughly prevent the multiple 
early media streams from reaching the xmpp client. (It may be able to 
send updates to all but one to stop their media, but that takes some 
time, and not all will support it.)

So either you mandate that the GW terminate the media, or else you have 
to acknowledge the situation and warn the xmpp client to be prepared.

	Thanks,
	Paul

From miconda@gmail.com  Wed Aug 21 12:28:21 2013
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Date: Wed, 21 Aug 2013 21:28:04 +0200
From: Daniel-Constantin Mierla <miconda@gmail.com>
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Subject: Re: [Stox] core issues
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On 8/21/13 9:03 PM, Paul Kyzivat wrote:
> On 8/21/13 3:17 AM, Daniel-Constantin Mierla wrote:
>>
>> On 8/21/13 1:50 AM, Paul Kyzivat wrote:
>
>>> You can make it sound easy by ignoring the details, but at the expense
>>> of quality of implementation.
>>>
>>> The problem is that you may get multiple early dialogs, with media. At
>>> that time you don't know which one will answer first. So what do you
>>> do with the media while awaiting a 2xx?
>>>
>>> You can't "pick one" without the risk that it is the wrong one.
>>>
>>> You can "pick one" and switch it if it turns out to be the wrong one.
>>> But unless you relay the media the xmpp side will see a change, and so
>>> there must be some plan for how that is handled.
>> My comments were for the remark related to multiple 200ok, I haven't
>> said sending BYE for additional early dialogs.
>
> They are tightly connected issues.

If you just look at parallel forking, but handling of them is totally 
different.

>
>> For multiple early dialogs, the gateway has to maintain all of them
>> until a 200ok. It's also implementation decision which of early media
>> streams is sent to xmpp/jingle (again, it looks like it's mostly the
>> first one from experiences out there).
>
> This is straightforward if the GW terminates/relays media.
> But that is a cost.
>
> A signaling-only GW can't easily or thoroughly prevent the multiple 
> early media streams from reaching the xmpp client. (It may be able to 
> send updates to all but one to stop their media, but that takes some 
> time, and not all will support it.)
>
> So either you mandate that the GW terminate the media, or else you 
> have to acknowledge the situation and warn the xmpp client to be 
> prepared.
 From what's being repeated here, the idea is not to change existing 
protcols by enforcing new behaviour/support for features. It would be 
easier and more logical if changes can be made in both sides, but again, 
it's out of defined scope. Thus I guess the gateway should do stripping 
of what is not supported on the other side.

Since the issue is for calls from xmpp to sip, perhaps the multiple 
early media streams issue can be overcome by initiating the SIP side 
call with INVITE without SDP. The gateway needs to hold the SDP for ACK, 
but that should be no problem as I guess it has to be transaction 
stateful at least.

Daniel

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From saul@ag-projects.com  Wed Aug 21 12:36:21 2013
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [Stox] core issues
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On Aug 21, 2013, at 9:03 PM, Paul Kyzivat wrote:

> On 8/21/13 3:17 AM, Daniel-Constantin Mierla wrote:
>>=20
>> On 8/21/13 1:50 AM, Paul Kyzivat wrote:
>=20
>>> You can make it sound easy by ignoring the details, but at the =
expense
>>> of quality of implementation.
>>>=20
>>> The problem is that you may get multiple early dialogs, with media. =
At
>>> that time you don't know which one will answer first. So what do you
>>> do with the media while awaiting a 2xx?
>>>=20
>>> You can't "pick one" without the risk that it is the wrong one.
>>>=20
>>> You can "pick one" and switch it if it turns out to be the wrong =
one.
>>> But unless you relay the media the xmpp side will see a change, and =
so
>>> there must be some plan for how that is handled.
>> My comments were for the remark related to multiple 200ok, I haven't
>> said sending BYE for additional early dialogs.
>=20
> They are tightly connected issues.
>=20
>> For multiple early dialogs, the gateway has to maintain all of them
>> until a 200ok. It's also implementation decision which of early media
>> streams is sent to xmpp/jingle (again, it looks like it's mostly the
>> first one from experiences out there).
>=20
> This is straightforward if the GW terminates/relays media.
> But that is a cost.
>=20
> A signaling-only GW can't easily or thoroughly prevent the multiple =
early media streams from reaching the xmpp client. (It may be able to =
send updates to all but one to stop their media, but that takes some =
time, and not all will support it.)
>=20
> So either you mandate that the GW terminate the media, or else you =
have to acknowledge the situation and warn the xmpp client to be =
prepared.
>=20

I see the subject hasn't changed, so are we talking about the -core =
draft or the -media draft here? The early media situation is indeed =
tricky when RTP is being used, but if we are dealing with chat, there =
should be no problem in keeping a few TCP connections alive until one of =
the dialogs has been confirmed. Also, when chat is in use the GW =
effectively terminates the media.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From mary.ietf.barnes@gmail.com  Wed Aug 21 13:33:16 2013
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Peter Saint-Andre <stpeter@stpeter.im>
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-core-02.txt
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--001a1133b3e4371bf104e47b15b6
Content-Type: text/plain; charset=ISO-8859-1

I've reviewed this version of the document and overall, I think the
document is ready for WGLC/progressing.  I have a minor comment and nit
that could be fixed before or after the document goes through WGLC.  As far
as the error code handling, I think the text you added in the -02 is fine
and I don't have concerns about the mappings that are suggested.

Minor comments/Nits:
- Section 5.3 (1st para, last sentence):  "resource part" -> "resourcepart"

- Section 5.4 & 5.5.  I personally would find a couple of examples,
highlighting some of the rules (e.g., for the disallowed characters, etc.)
of the mapping to be quite helpful. As always, they aren't normative, but I
think it would help cement the rules in some of our brains.

Regards,
Mary


On Mon, Aug 19, 2013 at 3:42 PM, Peter Saint-Andre <stpeter@stpeter.im>wrote:

> As previously mentioned.
>
> I view some of this text as provisional, but submitting a revised I-D
> seemed to be the best way for folks to review it.
>
> On 8/19/13 2:41 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 SIP-TO-XMPP Working Group of the IETF.
> >
> >       Title           : Interworking between the Session Initiation
> Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP):
> Architecture, Addresses, and Error Handling
> >       Author(s)       : Peter Saint-Andre
> >                           Avshalom Houri
> >                           Joe Hildebrand
> >       Filename        : draft-ietf-stox-core-02.txt
> >       Pages           : 16
> >       Date            : 2013-08-19
> >
> > Abstract:
> >    As a foundation for the definition of bidirectional protocol mappings
> >    between the Session Initiation Protocol (SIP) and the Extensible
> >    Messaging and Presence Protocol (XMPP), this document specifies the
> >    architectural assumptions underlying such mappings as well as the
> >    mapping of addresses and error conditions.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-stox-core
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-stox-core-02
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-core-02
> >
> >
> > Please note that it may take a couple of minutes from the time of
> submission
> > until the htmlized version and diff are available at tools.ietf.org.
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > stox mailing list
> > stox@ietf.org
> > https://www.ietf.org/mailman/listinfo/stox
> >
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>

--001a1133b3e4371bf104e47b15b6
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I&#39;ve reviewed this version of the document and=A0overa=
ll, I think the document is ready for WGLC/progressing. =A0I have a minor c=
omment and nit that could be fixed before or after the document goes throug=
h WGLC. =A0As far as the error code handling, I think the text you added in=
 the -02 is fine and I don&#39;t have concerns about the mappings that are =
suggested.=A0<div>
<div><br></div><div>Minor comments/Nits:</div><div>- Section 5.3 (1st para,=
 last sentence): =A0&quot;resource part&quot; -&gt; &quot;resourcepart&quot=
;</div><div><br></div><div>- Section 5.4 &amp; 5.5. =A0I personally would f=
ind a couple of examples, highlighting some of the rules (e.g., for the dis=
allowed characters, etc.) of the mapping to be quite helpful. As always, th=
ey aren&#39;t normative, but I think it would help cement the rules in some=
 of our brains.=A0</div>
</div><div><br></div><div>Regards,</div><div>Mary</div></div><div class=3D"=
gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Aug 19, 2013 at 3:4=
2 PM, Peter Saint-Andre <span dir=3D"ltr">&lt;<a href=3D"mailto:stpeter@stp=
eter.im" target=3D"_blank">stpeter@stpeter.im</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 previously mentioned.<br>
<br>
I view some of this text as provisional, but submitting a revised I-D<br>
seemed to be the best way for folks to review it.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On 8/19/13 2:41 PM, <a href=3D"mailto:internet-drafts@ietf.org">internet-dr=
afts@ietf.org</a> wrote:<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt; =A0This draft is a work item of the SIP-TO-XMPP Working Group of the I=
ETF.<br>
&gt;<br>
&gt; =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : Interworking between the Sessi=
on Initiation Protocol (SIP) and the Extensible Messaging and Presence Prot=
ocol (XMPP): Architecture, Addresses, and Error Handling<br>
&gt; =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Peter Saint-Andre<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Avshalom Houri<br>
&gt; =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Joe Hildebrand<br>
&gt; =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-ietf-stox-core-02.txt<br>
&gt; =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 16<br>
&gt; =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-08-19<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =A0 =A0As a foundation for the definition of bidirectional protocol ma=
ppings<br>
&gt; =A0 =A0between the Session Initiation Protocol (SIP) and the Extensibl=
e<br>
&gt; =A0 =A0Messaging and Presence Protocol (XMPP), this document specifies=
 the<br>
&gt; =A0 =A0architectural assumptions underlying such mappings as well as t=
he<br>
&gt; =A0 =A0mapping of addresses and error conditions.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-stox-core" targ=
et=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-stox-core</a><br>
&gt;<br>
&gt; There&#39;s also a htmlized version available at:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-ietf-stox-core-02" target=
=3D"_blank">http://tools.ietf.org/html/draft-ietf-stox-core-02</a><br>
&gt;<br>
&gt; A diff from the previous version is available at:<br>
&gt; <a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-02"=
 target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-=
02</a><br>
&gt;<br>
&gt;<br>
&gt; Please note that it may take a couple of minutes from the time of subm=
ission<br>
&gt; until the htmlized version and diff are available at <a href=3D"http:/=
/tools.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
&gt;<br>
&gt; Internet-Drafts are also available by anonymous FTP at:<br>
&gt; <a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp:=
//ftp.ietf.org/internet-drafts/</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; stox mailing list<br>
&gt; <a href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/stox" target=3D"_blan=
k">https://www.ietf.org/mailman/listinfo/stox</a><br>
&gt;<br>
<br>
<br>
_______________________________________________<br>
stox mailing list<br>
<a href=3D"mailto:stox@ietf.org">stox@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/stox" target=3D"_blank">ht=
tps://www.ietf.org/mailman/listinfo/stox</a><br>
</div></div></blockquote></div><br></div>

--001a1133b3e4371bf104e47b15b6--

From miconda@gmail.com  Wed Aug 21 14:26:31 2013
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Date: Wed, 21 Aug 2013 23:26:18 +0200
From: Daniel-Constantin Mierla <miconda@gmail.com>
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-im-01.txt
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Few remarks about the SIP MESSAGE examples:
- iirc, the From tag parameter is mandatory in all SIP requests by 
rfc3261, not only for those that may involve a dialog, thus for MESSAGE 
as well (although I couldn't spot it quickly in the rfc3261, at least 
the examples in rfc3428 are with From tag)
- gr is defined as URI parameter (for the GRUU purposes), but in the 
draft it is actually a header parameter. The From URI along with gr 
parameter must be enclosed in between < > to make it paramter for URI

Daniel

On 8/20/13 6:22 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 SIP-TO-XMPP Working Group of the IETF.
>
> 	Title           : Interworking between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP): Instant Messaging
> 	Author(s)       : Peter Saint-Andre
>                            Avshalom Houri
>                            Joe Hildebrand
> 	Filename        : draft-ietf-stox-im-01.txt
> 	Pages           : 11
> 	Date            : 2013-08-20
>
> Abstract:
>     This document defines a bidirectional protocol mapping for the
>     exchange of single instant messages between the Session Initiation
>     Protocol (SIP) and the Extensible Messaging and Presence Protocol
>     (XMPP).
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-stox-im
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-stox-im-01
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-stox-im-01
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox

-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda


From stpeter@stpeter.im  Wed Aug 21 19:24:09 2013
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Hi Mary, thanks for the review!

On 8/21/13 2:33 PM, Mary Barnes wrote:
> I've reviewed this version of the document and overall, I think the
> document is ready for WGLC/progressing.  I have a minor comment and nit
> that could be fixed before or after the document goes through WGLC.  As
> far as the error code handling, I think the text you added in the -02 is
> fine and I don't have concerns about the mappings that are suggested. 
> 
> Minor comments/Nits:
> - Section 5.3 (1st para, last sentence):  "resource part" -> "resourcepart"

Noted.

> - Section 5.4 & 5.5.  I personally would find a couple of examples,
> highlighting some of the rules (e.g., for the disallowed characters,
> etc.) of the mapping to be quite helpful. As always, they aren't
> normative, but I think it would help cement the rules in some of our
> brains. 

Good idea. We'll work those into the next revision.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From internet-drafts@ietf.org  Wed Aug 21 20:36:04 2013
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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 SIP-TO-XMPP Working Group of the IETF.

	Title           : Interworking between the Session Initiation Protocol (SI=
P) and the Extensible Messaging and Presence Protocol (XMPP): Architecture,=
 Addresses, and Error Handling
	Author(s)       : Peter Saint-Andre
                          Avshalom Houri
                          Joe Hildebrand
	Filename        : draft-ietf-stox-core-03.txt
	Pages           : 17
	Date            : 2013-08-21

Abstract:
   As a foundation for the definition of bidirectional protocol mappings
   between the Session Initiation Protocol (SIP) and the Extensible
   Messaging and Presence Protocol (XMPP), this document specifies the
   architectural assumptions underlying such mappings as well as the
   mapping of addresses and error conditions.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-stox-core

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-stox-core-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-stox-core-03


Please note that it may take a couple of minutes from the time of submission
until the htmlized version and diff are available at tools.ietf.org.

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


From stpeter@stpeter.im  Wed Aug 21 20:43:09 2013
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On 8/21/13 3:26 PM, Daniel-Constantin Mierla wrote:
> Few remarks about the SIP MESSAGE examples:
> - iirc, the From tag parameter is mandatory in all SIP requests by
> rfc3261, not only for those that may involve a dialog, thus for MESSAGE
> as well (although I couldn't spot it quickly in the rfc3261, at least
> the examples in rfc3428 are with From tag)
> - gr is defined as URI parameter (for the GRUU purposes), but in the
> draft it is actually a header parameter. The From URI along with gr
> parameter must be enclosed in between < > to make it paramter for URI

So...

OLD
   From: sip:juliet@example.com;gr=balcony

NEW
   From: <sip:juliet@example.com>;tag=12345;gr=balcony

Correct?

Peter

-- 
Peter Saint-Andre
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On Aug 22, 2013, at 5:42 AM, Peter Saint-Andre wrote:

> On 8/21/13 3:26 PM, Daniel-Constantin Mierla wrote:
>> Few remarks about the SIP MESSAGE examples:
>> - iirc, the =46rom tag parameter is mandatory in all SIP requests by
>> rfc3261, not only for those that may involve a dialog, thus for =
MESSAGE
>> as well (although I couldn't spot it quickly in the rfc3261, at least
>> the examples in rfc3428 are with =46rom tag)
>> - gr is defined as URI parameter (for the GRUU purposes), but in the
>> draft it is actually a header parameter. The =46rom URI along with gr
>> parameter must be enclosed in between < > to make it paramter for URI
>=20
> So...
>=20
> OLD
>   From: sip:juliet@example.com;gr=3Dbalcony
>=20
> NEW
>   From: <sip:juliet@example.com>;tag=3D12345;gr=3Dbalcony
>=20
> Correct?
>=20

Nope, it should be:

From: <sip:juliet@example.com;gr=3Dbalcony>;tag=3D12345

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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A few comments based on the document and not discussions on the list :-)

5.1

Separate URI schemas defined by SIP and URI schemas the protocol =
support. SIP supports any URI, and implementations should accept XMPP:, =
URN: and other schemas - even though there's no requirement to support =
it.

The "Tel:" URI should propably be mentioned in this section.

In theory I would like to use XMPP: uri's in SIP, but I understand few =
implementors and implementations would understand this. ;-) I am not =
sure about the implementation status of IM: and PRES: either, maybe =
that's something we should add to the questionnaire at SIPit. We do test =
support of strange URI schemas, like PIZZADELIVERY:=20

"IDNs are not allowed in the SIP address format"

I think that's a bit too hard. We don't carry IDNs in the protocol, but =
support of it is up to the user agent. Would propose: "IDNs are only =
handled in encoded format in the SIP protocol."=20

5.3

Since SIP is declared to be UTF-8 by default, I'm not sure that the gr =
parameter can only contain ASCII. Im no expert in ABNF parsing, but =
quoted-string seems to my I can quote UTF-8.
quoted-string  =3D  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE
      qdtext         =3D  LWS / %x21 / %x23-5B / %x5D-7E
                        / UTF8-NONASCII
The text in the draft can be parsed to say this, but I would like it to =
be more clear and add the UTF-8 acronym, so it's clear that it's just =
transcoding - quoting - but not translating or stripping.
I would also like to mention that Contacts can include display names, =
like "Olles superduperphone" or "Conference Room Jupiter Phone". I don't =
know if this can be translated into XMPP.
6.1. Table 2
Yes, this is an art form :-)
I notice that 407/401 is translated to different things. I would propose =
that we always translate to 401 or that we add both 401/407 to the =
entries <not-authorized>, <registration-required/> and =
<subscription-required/>. <
401 is for an endpoint delivering a server, a UAS like a registrar or a =
b2bua. 407 is for an intermediary like a proxy. A single request can =
have multiple 407 challenges, but only one 401. After reading mail on =
the mailing list I think we have gateways considering itself an endpoint =
and a possible architecture with a gateway that acts more like a proxy.
I'm not sure about the translation of xmpp "Gone" to SIP "410 Gone" =
either. 410 Gone in sip means that an AOR existed at some point, but is =
no longer in use and no redirection is possible. XMPP gone seems to me =
more like someone that left the chat, but is still alive and fine and =
may show up by the balcony another night.
<policy-violation/> should be translated to 403 Forbidden
<redirect/> to 300 Multiple Choices may be correct, but I would prefer a =
302 or 301. I haven't seen many implementations of 300 for redirects, =
but many using good old 302.
<jid-malformed/> is not a 484. 484 Address Incomplete is used for =
overlapped dialling. Malformed URI should be 400.
<not-allowed/> should translate to 403 Forbidden I guess. Needs review. =
405 indicates that the METHOD is not accepted, like a SIP message using =
PUBLISH to a server that doesn't support PUBLISH.
<remote-server-not-found/> with a translation to 502 is maybe not fully =
wrong, but in real life if the DNS entry is there and we can't reach the =
server, a 408 is sent. If there's no DNS entry at all, a 404 will be =
sent. Depends a bit on the meaning of this XMPP error code.
<remote-server-timeout/> is a 408 Request Timeout.
Section 6.2. table 3
(see above and apply in reverse :-)
Section 8
I'm happy to see loop detection, but I think we need a way to transport =
the values across XMPP for the case we have SIP -> XMPP -> SIP
We don't want to loop around or fork in crazy ways in such a network. I =
can create such a loop easily with Asterisk today. People will end up =
building this kind of gateway constructs. I don't think we can say =
"that's not what we intend" since it will happen and possibly already is =
in use out there. It may be something for another draft, but it will =
have to be approached at some point.
/O


--Apple-Mail=_D31CC5BE-8276-4DBF-ADCB-5054FA25EFBF
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">A few =
comments based on the document and not discussions on the list =
:-)<div><br></div><div>5.1</div><div><br></div><div>Separate URI schemas =
defined by SIP and URI schemas the protocol support. SIP supports any =
URI, and implementations should accept XMPP:, URN: and other schemas - =
even though there's no requirement to support =
it.</div><div><br></div><div>The "Tel:" URI should propably be mentioned =
in this section.</div><div><br></div><div>In theory I would like to use =
XMPP: uri's in SIP, but I understand few implementors and =
implementations would understand this. ;-) I am not sure about the =
implementation status of IM: and PRES: either, maybe that's something we =
should add to the questionnaire at SIPit. We do test support of strange =
URI schemas, like PIZZADELIVERY:&nbsp;</div><div><br></div><div>"<span =
style=3D"font-size: 1em; ">IDNs are not allowed in the&nbsp;</span><span =
style=3D"font-size: 1em; ">SIP address =
format"</span></div><div><br></div><div>I think that's a bit too hard. =
We don't carry IDNs in the protocol, but support of it is up to the user =
agent. Would propose: "IDNs are only handled in encoded format in the =
SIP =
protocol."&nbsp;</div><div><br></div><div>5.3</div><div><br></div><div><sp=
an style=3D"font-size: 1em; ">Since SIP is declared to be UTF-8 by =
default, I'm not sure that the gr parameter can only contain ASCII. Im =
no expert in ABNF parsing, but quoted-string seems to my I can quote =
UTF-8.</span></div><div><pre style=3D"word-wrap: break-word; =
white-space: pre-wrap; ">quoted-string  =3D  SWS DQUOTE *(qdtext / =
quoted-pair ) DQUOTE
      qdtext         =3D  LWS / %x21 / %x23-5B / %x5D-7E
                        / UTF8-NONASCII</pre><pre style=3D"word-wrap: =
break-word; white-space: pre-wrap; ">The text in the draft can be parsed =
to say this, but I would like it to be more clear and add the UTF-8 =
acronym, so it's clear that it's just transcoding - quoting - but not =
translating or stripping.</pre><pre style=3D"word-wrap: break-word; =
white-space: pre-wrap; ">I would also like to mention that Contacts can =
include display names, like "Olles superduperphone" or "Conference Room =
Jupiter Phone". I don't know if this can be translated into =
XMPP.</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">6.1. Table 2</pre><pre style=3D"word-wrap: break-word; white-space: =
pre-wrap; ">Yes, this is an art form :-)</pre><pre style=3D"word-wrap: =
break-word; white-space: pre-wrap; ">I notice that 407/401 is translated =
to different things. I would propose that we always translate to 401 or =
that we add both 401/407 to the entries &lt;not-authorized&gt;, =
&lt;registration-required/&gt; and &lt;subscription-required/&gt;. =
&lt;</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">401 is for an endpoint delivering a server, a UAS like a registrar or =
a b2bua. 407 is for an intermediary like a proxy. A single request can =
have multiple 407 challenges, but only one 401. After reading mail on =
the mailing list I think we have gateways considering itself an endpoint =
and a possible architecture with a gateway that acts more like a =
proxy.</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">I'm not sure about the translation of xmpp "Gone" to SIP "410 Gone" =
either. 410 Gone in sip means that an AOR existed at some point, but is =
no longer in use and no redirection is possible. XMPP gone seems to me =
more like someone that left the chat, but is still alive and fine and =
may show up by the balcony another night.</pre><pre style=3D"word-wrap: =
break-word; white-space: pre-wrap; ">&lt;policy-violation/&gt; should be =
translated to 403 Forbidden</pre><pre style=3D"word-wrap: break-word; =
white-space: pre-wrap; ">&lt;redirect/&gt; to 300 Multiple Choices may =
be correct, but I would prefer a 302 or 301. I haven't seen many =
implementations of 300 for redirects, but many using good old =
302.</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">&lt;jid-malformed/&gt; is not a 484. 484 Address Incomplete is used =
for overlapped dialling. Malformed URI should be 400.</pre><pre =
style=3D"word-wrap: break-word; white-space: pre-wrap; =
">&lt;not-allowed/&gt; should translate to 403 Forbidden I guess. Needs =
review. 405 indicates that the METHOD is not accepted, like a SIP =
message using PUBLISH to a server that doesn't support =
PUBLISH.</pre><pre style=3D"word-wrap: break-word; white-space: =
pre-wrap; ">&lt;remote-server-not-found/&gt; with a translation to 502 =
is maybe not fully wrong, but in real life if the DNS entry is there and =
we can't reach the server, a 408 is sent. If there's no DNS entry at =
all, a 404 will be sent. Depends a bit on the meaning of this XMPP error =
code.</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">&lt;remote-server-timeout/&gt; is a 408 Request Timeout.</pre><pre =
style=3D"word-wrap: break-word; white-space: pre-wrap; ">Section 6.2. =
table 3</pre><pre style=3D"word-wrap: break-word; white-space: pre-wrap; =
">(see above and apply in reverse :-)</pre><pre style=3D"word-wrap: =
break-word; white-space: pre-wrap; ">Section 8</pre><pre =
style=3D"word-wrap: break-word; white-space: pre-wrap; ">I'm happy to =
see loop detection, but I think we need a way to transport the values =
across XMPP for the case we have SIP -&gt; XMPP -&gt; SIP</pre><pre =
style=3D"word-wrap: break-word; white-space: pre-wrap; ">We don't want =
to loop around or fork in crazy ways in such a network. I can create =
such a loop easily with Asterisk today. People will end up building this =
kind of gateway constructs. I don't think we can say "that's not what we =
intend" since it will happen and possibly already is in use out there. =
It may be something for another draft, but it will have to be approached =
at some point.</pre><pre style=3D"word-wrap: break-word; white-space: =
pre-wrap; ">/O</pre><pre style=3D"word-wrap: break-word; white-space: =
pre-wrap; "><br></pre></div></body></html>=

--Apple-Mail=_D31CC5BE-8276-4DBF-ADCB-5054FA25EFBF--

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On 8/22/13 1:10 AM, Saúl Ibarra Corretgé wrote:
> 
> On Aug 22, 2013, at 5:42 AM, Peter Saint-Andre wrote:
> 
>> On 8/21/13 3:26 PM, Daniel-Constantin Mierla wrote:
>>> Few remarks about the SIP MESSAGE examples:
>>> - iirc, the From tag parameter is mandatory in all SIP requests by
>>> rfc3261, not only for those that may involve a dialog, thus for MESSAGE
>>> as well (although I couldn't spot it quickly in the rfc3261, at least
>>> the examples in rfc3428 are with From tag)
>>> - gr is defined as URI parameter (for the GRUU purposes), but in the
>>> draft it is actually a header parameter. The From URI along with gr
>>> parameter must be enclosed in between < > to make it paramter for URI
>>
>> So...
>>
>> OLD
>>   From: sip:juliet@example.com;gr=balcony
>>
>> NEW
>>   From: <sip:juliet@example.com>;tag=12345;gr=balcony
>>
>> Correct?
>>
> 
> Nope, it should be:
> 
> From: <sip:juliet@example.com;gr=balcony>;tag=12345

Noted, thanks!

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From pkyzivat@alum.mit.edu  Thu Aug 22 13:01:32 2013
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Subject: Re: [Stox] I-D Action: draft-ietf-stox-im-01.txt
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What could be simpler? Why would anyone get this wrong? :-)

On 8/22/13 3:10 AM, Saúl Ibarra Corretgé wrote:
>
> On Aug 22, 2013, at 5:42 AM, Peter Saint-Andre wrote:
>
>> On 8/21/13 3:26 PM, Daniel-Constantin Mierla wrote:
>>> Few remarks about the SIP MESSAGE examples:
>>> - iirc, the From tag parameter is mandatory in all SIP requests by
>>> rfc3261, not only for those that may involve a dialog, thus for MESSAGE
>>> as well (although I couldn't spot it quickly in the rfc3261, at least
>>> the examples in rfc3428 are with From tag)
>>> - gr is defined as URI parameter (for the GRUU purposes), but in the
>>> draft it is actually a header parameter. The From URI along with gr
>>> parameter must be enclosed in between < > to make it paramter for URI
>>
>> So...
>>
>> OLD
>>    From: sip:juliet@example.com;gr=balcony
>>
>> NEW
>>    From: <sip:juliet@example.com>;tag=12345;gr=balcony
>>
>> Correct?
>>
>
> Nope, it should be:
>
> From: <sip:juliet@example.com;gr=balcony>;tag=12345
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
> _______________________________________________
> stox mailing list
> stox@ietf.org
> https://www.ietf.org/mailman/listinfo/stox
>


From pkyzivat@alum.mit.edu  Thu Aug 22 13:36:11 2013
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Subject: Re: [Stox] core issues
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On 8/21/13 3:28 PM, Daniel-Constantin Mierla wrote:
>
> On 8/21/13 9:03 PM, Paul Kyzivat wrote:
>> On 8/21/13 3:17 AM, Daniel-Constantin Mierla wrote:
>>>
>>> On 8/21/13 1:50 AM, Paul Kyzivat wrote:
>>
>>>> You can make it sound easy by ignoring the details, but at the expense
>>>> of quality of implementation.
>>>>
>>>> The problem is that you may get multiple early dialogs, with media. At
>>>> that time you don't know which one will answer first. So what do you
>>>> do with the media while awaiting a 2xx?
>>>>
>>>> You can't "pick one" without the risk that it is the wrong one.
>>>>
>>>> You can "pick one" and switch it if it turns out to be the wrong one.
>>>> But unless you relay the media the xmpp side will see a change, and so
>>>> there must be some plan for how that is handled.
>>> My comments were for the remark related to multiple 200ok, I haven't
>>> said sending BYE for additional early dialogs.
>>
>> They are tightly connected issues.
>
> If you just look at parallel forking, but handling of them is totally
> different.
>
>>
>>> For multiple early dialogs, the gateway has to maintain all of them
>>> until a 200ok. It's also implementation decision which of early media
>>> streams is sent to xmpp/jingle (again, it looks like it's mostly the
>>> first one from experiences out there).
>>
>> This is straightforward if the GW terminates/relays media.
>> But that is a cost.
>>
>> A signaling-only GW can't easily or thoroughly prevent the multiple
>> early media streams from reaching the xmpp client. (It may be able to
>> send updates to all but one to stop their media, but that takes some
>> time, and not all will support it.)
>>
>> So either you mandate that the GW terminate the media, or else you
>> have to acknowledge the situation and warn the xmpp client to be
>> prepared.
>  From what's being repeated here, the idea is not to change existing
> protcols by enforcing new behaviour/support for features. It would be
> easier and more logical if changes can be made in both sides, but again,
> it's out of defined scope. Thus I guess the gateway should do stripping
> of what is not supported on the other side.
>
> Since the issue is for calls from xmpp to sip, perhaps the multiple
> early media streams issue can be overcome by initiating the SIP side
> call with INVITE without SDP. The gateway needs to hold the SDP for ACK,
> but that should be no problem as I guess it has to be transaction
> stateful at least.

That is indeed one way to solve the problem. But it will also be 
necessary to *not* support 100rel, or else the problem can still arise.

	Thanks,
	Paul


From pkyzivat@alum.mit.edu  Thu Aug 22 13:41:07 2013
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Subject: Re: [Stox] core issues
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On 8/21/13 3:35 PM, Saúl Ibarra Corretgé wrote:
>
> On Aug 21, 2013, at 9:03 PM, Paul Kyzivat wrote:
>
>> On 8/21/13 3:17 AM, Daniel-Constantin Mierla wrote:
>>>
>>> On 8/21/13 1:50 AM, Paul Kyzivat wrote:
>>
>>>> You can make it sound easy by ignoring the details, but at the expense
>>>> of quality of implementation.
>>>>
>>>> The problem is that you may get multiple early dialogs, with media. At
>>>> that time you don't know which one will answer first. So what do you
>>>> do with the media while awaiting a 2xx?
>>>>
>>>> You can't "pick one" without the risk that it is the wrong one.
>>>>
>>>> You can "pick one" and switch it if it turns out to be the wrong one.
>>>> But unless you relay the media the xmpp side will see a change, and so
>>>> there must be some plan for how that is handled.
>>> My comments were for the remark related to multiple 200ok, I haven't
>>> said sending BYE for additional early dialogs.
>>
>> They are tightly connected issues.
>>
>>> For multiple early dialogs, the gateway has to maintain all of them
>>> until a 200ok. It's also implementation decision which of early media
>>> streams is sent to xmpp/jingle (again, it looks like it's mostly the
>>> first one from experiences out there).
>>
>> This is straightforward if the GW terminates/relays media.
>> But that is a cost.
>>
>> A signaling-only GW can't easily or thoroughly prevent the multiple early media streams from reaching the xmpp client. (It may be able to send updates to all but one to stop their media, but that takes some time, and not all will support it.)
>>
>> So either you mandate that the GW terminate the media, or else you have to acknowledge the situation and warn the xmpp client to be prepared.
>>
>
> I see the subject hasn't changed, so are we talking about the -core draft or the -media draft here? The early media situation is indeed tricky when RTP is being used, but if we are dealing with chat, there should be no problem in keeping a few TCP connections alive until one of the dialogs has been confirmed. Also, when chat is in use the GW effectively terminates the media.

There isn't any way to have a signaling only GW for xmpp-msrp.
So yes, you are right about that. Its only an issue for RTP media.

	Thanks,
	Paul


From saul@ag-projects.com  Fri Aug 23 02:56:09 2013
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Subject: Re: [Stox] core issues
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>>=20
>> I see the subject hasn't changed, so are we talking about the -core =
draft or the -media draft here? The early media situation is indeed =
tricky when RTP is being used, but if we are dealing with chat, there =
should be no problem in keeping a few TCP connections alive until one of =
the dialogs has been confirmed. Also, when chat is in use the GW =
effectively terminates the media.
>=20
> There isn't any way to have a signaling only GW for xmpp-msrp.
> So yes, you are right about that. Its only an issue for RTP media.
>=20

Do, do you (or anyone) still think that any text on this regard is =
needed in the *core* document, or shall we just put that in the media =
one?


Regards,

--
Sa=FAl Ibarra Corretg=E9
AG Projects




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On 8/23/13 3:55 AM, Saúl Ibarra Corretgé wrote:
>>>
>>> I see the subject hasn't changed, so are we talking about the -core draft or the -media draft here? The early media situation is indeed tricky when RTP is being used, but if we are dealing with chat, there should be no problem in keeping a few TCP connections alive until one of the dialogs has been confirmed. Also, when chat is in use the GW effectively terminates the media.
>>
>> There isn't any way to have a signaling only GW for xmpp-msrp.
>> So yes, you are right about that. Its only an issue for RTP media.
>>
> 
> Do, do you (or anyone) still think that any text on this regard is needed in the *core* document, or shall we just put that in the media one?

I think this belongs in the -media document. Let's change the thread
name accordingly. :-)

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



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WFM

On 8/23/13 1:11 PM, Peter Saint-Andre wrote:
> On 8/23/13 3:55 AM, Saúl Ibarra Corretgé wrote:
>>>>
>>>> I see the subject hasn't changed, so are we talking about the -core draft or the -media draft here? The early media situation is indeed tricky when RTP is being used, but if we are dealing with chat, there should be no problem in keeping a few TCP connections alive until one of the dialogs has been confirmed. Also, when chat is in use the GW effectively terminates the media.
>>>
>>> There isn't any way to have a signaling only GW for xmpp-msrp.
>>> So yes, you are right about that. Its only an issue for RTP media.
>>>
>>
>> Do, do you (or anyone) still think that any text on this regard is needed in the *core* document, or shall we just put that in the media one?
>
> I think this belongs in the -media document. Let's change the thread
> name accordingly. :-)
>
> Peter
>


From Markus.Isomaki@nokia.com  Mon Aug 26 05:34:34 2013
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Subject: [Stox] stox-presence-03: a couple of comments
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Hi,

I have a couple of comments on the presence-03 draft:

* Some asymmetry between XMPP-to-SIP (3.2) and SIP-to-XMPP (3.3) directions

- 3.2.1 explains the DNS operations an XMPP domain uses to determine that t=
he foreign domain does not support XMPP but actually supports SIP/SIMPLE. P=
resumably something similar is needed in the reverse direction, but this is=
 not explained in 3.3.1.=20
- The case where the subscription is rejected is covered in 3.3.1 but not i=
n 3.2.1.

*200 OK to NOTIFY is missing in 3.2.1

- 3.2.1 shows a SIP 200 OK response to the SUBSCRIBE request (examples 2 an=
d 3), not to the subsequent NOTIFY (example 4). The 200 OK to NOTIFY is not=
 mapped to XMPP so it may be fine to leave it out but it could be added for=
 clarity.

* SUBSCRIBE with Expires:0?

- In SIP it is possible to one-time query the presence-value by sending a S=
UBSCRIBE with Expires value 0. This triggers just a single NOTIFY. Does tha=
t work nicely through the mapping to XMPP, or is it a special case that wou=
ld need to be separately covered?

Markus=20






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Subject: Re: [Stox] stox-presence-03: a couple of comments
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>=20
>=20
> * SUBSCRIBE with Expires:0?
>=20
> - In SIP it is possible to one-time query the presence-value by =
sending a SUBSCRIBE with Expires value 0. This triggers just a single =
NOTIFY. Does that work nicely through the mapping to XMPP, or is it a =
special case that would need to be separately covered?
>=20

Not sure if we have text about that, but basically it would be the same =
as sending a presence stanza of type 'probe'.

--
Sa=FAl Ibarra Corretg=E9
AG Projects




From stpeter@stpeter.im  Mon Aug 26 15:27:36 2013
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Subject: Re: [Stox] stox-presence-03: a couple of comments
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On 8/26/13 6:53 AM, Saúl Ibarra Corretgé wrote:
>> 
>> 
>> * SUBSCRIBE with Expires:0?
>> 
>> - In SIP it is possible to one-time query the presence-value by
>> sending a SUBSCRIBE with Expires value 0. This triggers just a
>> single NOTIFY. Does that work nicely through the mapping to XMPP,
>> or is it a special case that would need to be separately covered?
>> 
> 
> Not sure if we have text about that, but basically it would be the
> same as sending a presence stanza of type 'probe'.

Yes, one can send a probe, but unless you're already authorized to view
the person's presence (through a long-lived subscription) you won't
actually receive the presence data. I suppose the SIMPLE-XMPP gateway
would keep track of the long-lived subscription and would translate the
SUBSCRIBE with Expires:0 to a probe on the XMPP side. That would work.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Mon Aug 26 15:49:35 2013
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On 8/26/13 6:34 AM, Markus.Isomaki@nokia.com wrote:
> Hi,

Hi Markus, thanks for the feedback.

> I have a couple of comments on the presence-03 draft:
> 
> * Some asymmetry between XMPP-to-SIP (3.2) and SIP-to-XMPP (3.3)
> directions
> 
> - 3.2.1 explains the DNS operations an XMPP domain uses to determine
> that the foreign domain does not support XMPP but actually supports
> SIP/SIMPLE. Presumably something similar is needed in the reverse
> direction, but this is not explained in 3.3.1. - The case where the
> subscription is rejected is covered in 3.3.1 but not in 3.2.1.

Good point. We'll add something about that.

> *200 OK to NOTIFY is missing in 3.2.1
> 
> - 3.2.1 shows a SIP 200 OK response to the SUBSCRIBE request
> (examples 2 and 3), not to the subsequent NOTIFY (example 4). The 200
> OK to NOTIFY is not mapped to XMPP so it may be fine to leave it out
> but it could be added for clarity.

Agreed. Clarity is good. :-)

> * SUBSCRIBE with Expires:0?
> 
> - In SIP it is possible to one-time query the presence-value by
> sending a SUBSCRIBE with Expires value 0. This triggers just a single
> NOTIFY. Does that work nicely through the mapping to XMPP, or is it a
> special case that would need to be separately covered?

See my reply to Saúl. I think it would be reasonable to mention that.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From stpeter@stpeter.im  Mon Aug 26 19:59:14 2013
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Hej Olle, thanks for the review.

On 8/22/13 2:10 AM, Olle E. Johansson wrote:
> A few comments based on the document and not discussions on the list
> :-)
> 
> 5.1
> 
> Separate URI schemas defined by SIP and URI schemas the protocol 
> support. SIP supports any URI, and implementations should accept
> XMPP:, URN: and other schemas - even though there's no requirement to
> support it.

We're focused here on interoperability. I doubt that any SIP software
out there knows what to do with a 'urn:' URI in a SIP request (and I
have no idea what that would mean for communication!).

> The "Tel:" URI should propably be mentioned in this section.

Agreed.

> In theory I would like to use XMPP: uri's in SIP, but I understand
> few implementors and implementations would understand this. ;-) I am
> not sure about the implementation status of IM: and PRES: either,
> maybe that's something we should add to the questionnaire at SIPit.

Sounds like a good idea.

On the XMPP side, I don't think that many IM clients support the 'im'
and 'presence' schemes.

> We do test support of strange URI schemas, like PIZZADELIVERY:
> 
> "IDNs are not allowed in the SIP address format"
> 
> I think that's a bit too hard. We don't carry IDNs in the protocol,
> but support of it is up to the user agent. Would propose: "IDNs are
> only handled in encoded format in the SIP protocol."

By encoded, do you mean percent-encoded as in RFC 3986?

> 5.3
> 
> Since SIP is declared to be UTF-8 by default, I'm not sure that the
> gr parameter can only contain ASCII. Im no expert in ABNF parsing,
> but quoted-string seems to my I can quote UTF-8.
> 
> quoted-string  =  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE qdtext
> =  LWS / %x21 / %x23-5B / %x5D-7E / UTF8-NONASCII

Is it quoted-string or pvalue that we care about here? I think it's
pvalue because RFC 5627 defines a new parameter for SIP URIs, as follows:

   uri-parameter   =/ gr-param
   gr-param        = "gr" ["=" pvalue]   ; defined in RFC 3261

However, pvalue does allow Unicode characters outside the ASCII range,
as long as they are encoded:

pvalue            =  1*paramchar
paramchar         =  param-unreserved / unreserved / escaped

So strictly speaking it's only ASCII characters that allowed directly,
although UTF-8 characters outside the ASCII range can be included if
they are percent-encoded.

> The text in the draft can be parsed to say this, but I would like it
> to be more clear and add the UTF-8 acronym, so it's clear that it's
> just transcoding - quoting - but not translating or stripping.

I prefer the term "percent-encoding" (RFC 3986) to any of those terms.
But yes, the text needs to be clearer.

> I would also like to mention that Contacts can include display names,
> like "Olles superduperphone" or "Conference Room Jupiter Phone". I
> don't know if this can be translated into XMPP.

XMPP doesn't have display names in addresses (as email and SIP do),
although those could be translated into a nickname / handle using XEP-0172:

http://xmpp.org/extensions/xep-0172.html

So adding a note about that seems reasonable.

> 6.1. Table 2
> 
> Yes, this is an art form :-)
> 
> I notice that 407/401 is translated to different things. I would
> propose that we always translate to 401 or that we add both 401/407
> to the entries <not-authorized>, <registration-required/> and
> <subscription-required/>. <
> 
> 401 is for an endpoint delivering a server, a UAS like a registrar or
> a b2bua. 407 is for an intermediary like a proxy. A single request
> can have multiple 407 challenges, but only one 401. After reading
> mail on the mailing list I think we have gateways considering itself
> an endpoint and a possible architecture with a gateway that acts more
> like a proxy.

Proxy or B2BUA? Someone said B2BUA and that sounds right. If that's
right, then we probably would use 401 and not 407.

> I'm not sure about the translation of xmpp "Gone" to SIP "410 Gone"
> either. 410 Gone in sip means that an AOR existed at some point, but
> is no longer in use and no redirection is possible. XMPP gone seems
> to me more like someone that left the chat, but is still alive and
> fine and may show up by the balcony another night.

Oh, there are two kinds of "gone" in XMPP.

One is a chat state:

http://xmpp.org/extensions/xep-0085.html#definitions

The other is a stanza error that maps quite well to SIP 410, I think:

http://xmpp.org/rfcs/rfc6120.html#stanzas-error-conditions-gone

> <policy-violation/> should be translated to 403 Forbidden

That sounds reasonable.

> <redirect/> to 300 Multiple Choices may be correct, but I would
> prefer a 302 or 301. I haven't seen many implementations of 300 for
> redirects, but many using good old 302.

OK.

> <jid-malformed/> is not a 484. 484 Address Incomplete is used for
> overlapped dialling. Malformed URI should be 400.

OK.

> <not-allowed/> should translate to 403 Forbidden I guess. 

I think that's OK, because XMPP <not-allowed/> is defined as:

   The recipient or server does not allow any entity to perform the
   action (e.g., sending to entities at a blacklisted domain)...

> Needs
> review. 405 indicates that the METHOD is not accepted, like a SIP
> message using PUBLISH to a server that doesn't support PUBLISH.

Right, that sounds more like <feature-not-implemented/> in XMPP.

> <remote-server-not-found/> with a translation to 502 is maybe not
> fully wrong, but in real life if the DNS entry is there and we can't
> reach the server, a 408 is sent. If there's no DNS entry at all, a
> 404 will be sent. Depends a bit on the meaning of this XMPP error
> code.

In XMPP the spec doesn't distinguish between those two cases for
<remote-server-not-found/>:

   A remote server or service specified as part or all of the JID of
   the intended recipient does not exist or cannot be resolved (e.g.,
   there is no _xmpp-server._tcp DNS SRV record, the A or AAAA fallback
   resolution fails, or A/AAAA lookups succeed but there is no response
   on the IANA-registered port 5269)...

So I think we need to mention both codes and add some text.

> <remote-server-timeout/> is a 408 Request Timeout.

Sounds right.

> Section 6.2. table 3
> 
> (see above and apply in reverse :-)

Yes. :-)

> Section 8
> 
> I'm happy to see loop detection, but I think we need a way to
> transport the values across XMPP for the case we have SIP -> XMPP ->
> SIP

Hmm.

The only way to do that would be to use SHIM headers in XMPP:

http://xmpp.org/extensions/xep-0131.html

But those aren't widely deployed.

> We don't want to loop around or fork in crazy ways in such a network.
> I can create such a loop easily with Asterisk today. People will end
> up building this kind of gateway constructs. I don't think we can say
> "that's not what we intend" since it will happen and possibly already
> is in use out there. It may be something for another draft, but it
> will have to be approached at some point.

It sounds to me as if we might be defining something new for XMPP, which
it outside of our charter. But perhaps it makes sense to mention
XEP-0131 and sketch out how that would be used.

Peter

-- 
Peter Saint-Andre
https://stpeter.im/



From oej@edvina.net  Mon Aug 26 22:47:42 2013
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27 aug 2013 kl. 04:59 skrev Peter Saint-Andre <stpeter@stpeter.im>:

> Hej Olle, thanks for the review.
> 
> On 8/22/13 2:10 AM, Olle E. Johansson wrote:
>> A few comments based on the document and not discussions on the list
>> :-)
>> 
>> 5.1
>> 
>> Separate URI schemas defined by SIP and URI schemas the protocol 
>> support. SIP supports any URI, and implementations should accept
>> XMPP:, URN: and other schemas - even though there's no requirement to
>> support it.
> 
> We're focused here on interoperability. I doubt that any SIP software
> out there knows what to do with a 'urn:' URI in a SIP request (and I
> have no idea what that would mean for communication!).
Well, they will have to learn, as it is part of the ECRIT standard for
emergency calls.

> 
>> The "Tel:" URI should propably be mentioned in this section.
> 
> Agreed.
> 
>> In theory I would like to use XMPP: uri's in SIP, but I understand
>> few implementors and implementations would understand this. ;-) I am
>> not sure about the implementation status of IM: and PRES: either,
>> maybe that's something we should add to the questionnaire at SIPit.
> 
> Sounds like a good idea.
> 
> On the XMPP side, I don't think that many IM clients support the 'im'
> and 'presence' schemes.
> 
>> We do test support of strange URI schemas, like PIZZADELIVERY:
>> 
>> "IDNs are not allowed in the SIP address format"
>> 
>> I think that's a bit too hard. We don't carry IDNs in the protocol,
>> but support of it is up to the user agent. Would propose: "IDNs are
>> only handled in encoded format in the SIP protocol."
> 
> By encoded, do you mean percent-encoded as in RFC 3986?
Yes.
> 
>> 5.3
>> 
>> Since SIP is declared to be UTF-8 by default, I'm not sure that the
>> gr parameter can only contain ASCII. Im no expert in ABNF parsing,
>> but quoted-string seems to my I can quote UTF-8.
>> 
>> quoted-string  =  SWS DQUOTE *(qdtext / quoted-pair ) DQUOTE qdtext
>> =  LWS / %x21 / %x23-5B / %x5D-7E / UTF8-NONASCII
> 
> Is it quoted-string or pvalue that we care about here? I think it's
> pvalue because RFC 5627 defines a new parameter for SIP URIs, as follows:
> 
>   uri-parameter   =/ gr-param
>   gr-param        = "gr" ["=" pvalue]   ; defined in RFC 3261
> 
> However, pvalue does allow Unicode characters outside the ASCII range,
> as long as they are encoded:
> 
> pvalue            =  1*paramchar
> paramchar         =  param-unreserved / unreserved / escaped
> 
> So strictly speaking it's only ASCII characters that allowed directly,
> although UTF-8 characters outside the ASCII range can be included if
> they are percent-encoded.
Ok. 
> 
>> The text in the draft can be parsed to say this, but I would like it
>> to be more clear and add the UTF-8 acronym, so it's clear that it's
>> just transcoding - quoting - but not translating or stripping.
> 
> I prefer the term "percent-encoding" (RFC 3986) to any of those terms.
> But yes, the text needs to be clearer.
Agreed.

> 
>> I would also like to mention that Contacts can include display names,
>> like "Olles superduperphone" or "Conference Room Jupiter Phone". I
>> don't know if this can be translated into XMPP.
> 
> XMPP doesn't have display names in addresses (as email and SIP do),
> although those could be translated into a nickname / handle using XEP-0172:
> 
> http://xmpp.org/extensions/xep-0172.html
> 
> So adding a note about that seems reasonable.
> 
>> 6.1. Table 2
>> 
>> Yes, this is an art form :-)
>> 
>> I notice that 407/401 is translated to different things. I would
>> propose that we always translate to 401 or that we add both 401/407
>> to the entries <not-authorized>, <registration-required/> and
>> <subscription-required/>. <
>> 
>> 401 is for an endpoint delivering a server, a UAS like a registrar or
>> a b2bua. 407 is for an intermediary like a proxy. A single request
>> can have multiple 407 challenges, but only one 401. After reading
>> mail on the mailing list I think we have gateways considering itself
>> an endpoint and a possible architecture with a gateway that acts more
>> like a proxy.
> 
> Proxy or B2BUA? Someone said B2BUA and that sounds right. If that's
> right, then we probably would use 401 and not 407.
Yes. 401 is correct for b2bua's. (And don't use Asterisk as an example
of the opposite here. I've been trying to change that for years, but
have met resistance :-) )

> 
>> I'm not sure about the translation of xmpp "Gone" to SIP "410 Gone"
>> either. 410 Gone in sip means that an AOR existed at some point, but
>> is no longer in use and no redirection is possible. XMPP gone seems
>> to me more like someone that left the chat, but is still alive and
>> fine and may show up by the balcony another night.
> 
> Oh, there are two kinds of "gone" in XMPP.
> 
> One is a chat state:
> 
> http://xmpp.org/extensions/xep-0085.html#definitions
> 
> The other is a stanza error that maps quite well to SIP 410, I think:
> 
> http://xmpp.org/rfcs/rfc6120.html#stanzas-error-conditions-gone
> 
>> <policy-violation/> should be translated to 403 Forbidden
> 
> That sounds reasonable.
> 
>> <redirect/> to 300 Multiple Choices may be correct, but I would
>> prefer a 302 or 301. I haven't seen many implementations of 300 for
>> redirects, but many using good old 302.
> 
> OK.
> 
>> <jid-malformed/> is not a 484. 484 Address Incomplete is used for
>> overlapped dialling. Malformed URI should be 400.
> 
> OK.
> 
>> <not-allowed/> should translate to 403 Forbidden I guess. 
> 
> I think that's OK, because XMPP <not-allowed/> is defined as:
> 
>   The recipient or server does not allow any entity to perform the
>   action (e.g., sending to entities at a blacklisted domain)...
> 
>> Needs
>> review. 405 indicates that the METHOD is not accepted, like a SIP
>> message using PUBLISH to a server that doesn't support PUBLISH.
> 
> Right, that sounds more like <feature-not-implemented/> in XMPP.
> 
>> <remote-server-not-found/> with a translation to 502 is maybe not
>> fully wrong, but in real life if the DNS entry is there and we can't
>> reach the server, a 408 is sent. If there's no DNS entry at all, a
>> 404 will be sent. Depends a bit on the meaning of this XMPP error
>> code.
> 
> In XMPP the spec doesn't distinguish between those two cases for
> <remote-server-not-found/>:
> 
>   A remote server or service specified as part or all of the JID of
>   the intended recipient does not exist or cannot be resolved (e.g.,
>   there is no _xmpp-server._tcp DNS SRV record, the A or AAAA fallback
>   resolution fails, or A/AAAA lookups succeed but there is no response
>   on the IANA-registered port 5269)...
> 
> So I think we need to mention both codes and add some text.
> 
>> <remote-server-timeout/> is a 408 Request Timeout.
> 
> Sounds right.
> 
>> Section 6.2. table 3
>> 
>> (see above and apply in reverse :-)
> 
> Yes. :-)
> 
>> Section 8
>> 
>> I'm happy to see loop detection, but I think we need a way to
>> transport the values across XMPP for the case we have SIP -> XMPP ->
>> SIP
> 
> Hmm.
> 
> The only way to do that would be to use SHIM headers in XMPP:
> 
> http://xmpp.org/extensions/xep-0131.html
> 
> But those aren't widely deployed.
> 
>> We don't want to loop around or fork in crazy ways in such a network.
>> I can create such a loop easily with Asterisk today. People will end
>> up building this kind of gateway constructs. I don't think we can say
>> "that's not what we intend" since it will happen and possibly already
>> is in use out there. It may be something for another draft, but it
>> will have to be approached at some point.
> 
> It sounds to me as if we might be defining something new for XMPP, which
> it outside of our charter. But perhaps it makes sense to mention
> XEP-0131 and sketch out how that would be used.

Yes, at least put a big warning sign on this door and say "this is out of
scope for this document, but will be in scope for the document that
comes after this" :-)

/O
