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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Xavier and Jean-Francois,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks for putting this together.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Based on the recent list discussion, I would say that quite many
people are leaning towards the architecture you depict in Section 5.2, Figure
2: The session setup protocol is an application specific Javascript
implementation transported over HTTP or WebSocket, while media is running on
standard RTP supported by the browser. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In that model we can’t put many requirements on the
session setup protocol or its interworking with SIP. If the service provider
needs SIP interoperability (to connect to PSTN, to other service providers or
SIP phones), it is indeed THEIR burden to make sure they use something that has
a clean mapping to SIP – for instance, that they can do things like call
hold. On the other hand if the service provider is not interested in SIP interoperability,
they do not have to worry about that. In the IETF there are probably two ways
to address this interworking: a) do nothing and leave it completely to the implementers
and service providers, or b) define some kind of a SIP/BOSH/HTTP or
SIP/WebSocket mapping in the same way that the XMPP folks have done. The XMPP/BOSH
spec does have implementations both on the client/Javascript side as well as on
the server side, so I think that spec has had some value. (At least in a way
that the Javascript library and the BOSH servers can be implemented somewhat
independently.)<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The RTP/media stack on the other hand is definitely in the scope
of the IETF effort. I think we should standardize the RTP use in the browsers
and that would be one step towards interop with SIP phones. The critical thing
seems to be the STUN connectivity check or media authorization part. If we
mandate browsers to get that exchange done before they are allowed to generate
any RTP packets on behalf of the application, this will ruin the possibility of
interop with 99% of existing SIP clients (without some kind of an SBC). DTMF
transport capability may also be relevant interop requirement.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think these are the key issues we should consider wrt. SIP
phone interop.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Markus<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
rtc-web-bounces@alvestrand.no [mailto:rtc-web-bounces@alvestrand.no] <b>On
Behalf Of </b>ext Xavier Marjou<br>
<b>Sent:</b> 09 February, 2011 11:07<br>
<b>To:</b> DISPATCH list<br>
<b>Cc:</b> rtc-web@alvestrand.no; Jean-François Jestin<br>
<b>Subject:</b> [RTW] [dispatch] RTC-Web I-D about interworking between RTC-Web
and SIP-RTP<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi,<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>We have posted a draft about interworking requirements
between RTC-Web and SIP-RTP.<o:p></o:p></p>
<div>
<p class=MsoNormal><a
href="http://www.ietf.org/internet-drafts/draft-marjou-dispatch-rtcweb-sip-rtp-interwk-reqs-00.txt">http://www.ietf.org/internet-drafts/draft-marjou-dispatch-rtcweb-sip-rtp-interwk-reqs-00.txt</a><o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>Cheers,<o:p></o:p></p>
</div>
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<p class=MsoNormal>Xavier and Jean-François<o:p></o:p></p>
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