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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi Peter,<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 hope I’m answering the right question:<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 something like the STUN connectivity check or
authorization is required to be implemented in the browser to prevent the Javascript
application from sending RTP or some other UDP data to any arbitrary IP
addresses and port. If the browser only allows the sending after a successful connectivity
check it can ensure (to some extent) that the other peer also actually wants to
receive this traffic. That code needs to reside in the browser so that the
browser can act as the gatekeeper.<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>
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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"'> ext Peter Musgrave
[mailto:peter.musgrave@magorcorp.com] <br>
<b>Sent:</b> 09 February, 2011 22:16<br>
<b>To:</b> Adam Roach<br>
<b>Cc:</b> Isomaki Markus (Nokia-CIC/Espoo); rtc-web@alvestrand.no;
dispatch@ietf.org; xavier.marjou@orange-ftgroup.com<br>
<b>Subject:</b> Re: [dispatch] [RTW] 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>I also agree with the "make RTP just work" -
signal where and how you want.. <o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>Sorry if I am being dense here - do we not need ICE
connectivity checks to finish before RTP can be sent? Or is the point that this
will take too long to get into browser code and that ICE in Javascript with
STUN in the browser gets rolled out faster?<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>Peter Musgrave<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>On 2011-02-09, at 3:07 PM, Adam Roach wrote:<o:p></o:p></p>
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<br>
<o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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)</span><span style='font-size:13.5pt;
font-family:"Helvetica","sans-serif"'><o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
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