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<p class="MsoNormal"><a name="_MailEndCompose"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">Curious, why would you want to use OpenSIPS and Kamailio?<o:p></o:p></span></a></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">There both SIP proxies.<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">Robert
<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"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> Users [mailto:users-bounces@lists.opensips.org]
<b>On Behalf Of </b>Alex Megalokonomos<br>
<b>Sent:</b> Wednesday, June 28, 2017 5:47 AM<br>
<b>To:</b> users@lists.opensips.org<br>
<b>Subject:</b> [OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><img id="_x0000_i1025" src="https://mailtrack.io/trace/mail/3c783866054bc5242de3e7df06ab3a49285c0f0b.png?u=1422671">Hello, <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">We have the following scenario: our office call center is an Alcatel OmniPCX Office setup.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">This handles most of our needs and also provides 4 SIP extensions.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no webrtc or websockets support)<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">What we would like to do is set up an OpenSIPS instance to handle WebRTC and proxy everything to this Kamailio SIP server.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">The idea is to allow a web client (using sip js or something similar) to register / make / receive calls as one of the Kamailio extensions.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">I think half of the configuration is this : <a href="https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips.org%2FDocumentation%2FTutorials-WebSocket-2-1&data=02%7C01%7Crmundkowsky%40ets.org%7Cbc057a0583854c1f54eb08d4be0ae688%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636342401422676698&sdata=Y%2BsdaA4hevNYQyWx0zkB4rVvG5glervN54dVx1XArNo%3D&reserved=0">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1</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">which I've already completed and indeed, clients can register to opensips and chat/make calls over websockets between them.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">How do I go about proxying registrations/invites/etc to the kamailio server instead?<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">best regards<o:p></o:p></p>
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