<div dir="ltr"><img width="0" height="0" class="mailtrack-img" style="float:right;" alt="" src="https://mailtrack.io/trace/mail/1676647efd99eeceab44e5dc440be7d1e78a150f.png?u=1422671">Hello Bogdan,<div><br></div><div>Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP in order for it to work) is exactly what we're looking for.</div><div><br></div><div>Unfortunately our Alcatel OmniPCX call center is a proprietary system that only allows for a limited number of SIP extensions (served from what appears to be an outdated customised Kamailio 3.2.2 from what I can tell from the headers.</div><div><br></div><div>For our normal internal office use it all works fine.</div><div><br></div><div>However we have 3 customer support lines that are currently routed to 3 extensions via OmniPCX.</div><div><br></div><div>We want to integrate these to our custom web-based CRM and the best way for us to do it is to use something like SIP js to handle and log calls, identify calling parties, bring up customer details etc.</div><div><br></div><div>Since the kamailio version inside OmniPCX does not support ws/webrtc we are looking to set up Opensips in exactly the way you described as a gateway/proxy for everything in order to convert the UDP-only sip extensions to ws+ webRTC capable ones.</div><div><br></div><div>I have used this tutorial <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1">http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1</a> to get what I assume is half the work (for RTP proxying) but I havent figured out the rest yet.</div><div><br></div><div>Best regards,</div><div>Alex</div><div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jun 29, 2017 at 11:43 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<tt>Hi Alex,<br>
<br>
First, some questions regarding the desired topology:<br>
1) the WS end-points should register in OpenSIPS or all the
way into Kamailio ?<br>
2) also, the calls from the WS end-points should be all the
time sent to Kamailio ?<br>
<br>
More or less, what I'm asking is : is OpenSIPS suppose to act as a
gateway from WS to UDP , but pass all the resulting traffic to
Kamailio ?<br>
<br>
Regards,<br>
</tt>
<pre class="m_-6874691586421181616moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.<wbr>com</a>
OpenSIPS Bootcamp 2017, Houston, US
<a class="m_-6874691586421181616moz-txt-link-freetext" href="http://opensips.org/training/OpenSIPS_Bootcamp_2017.html" target="_blank">http://opensips.org/training/<wbr>OpenSIPS_Bootcamp_2017.html</a>
</pre><div><div class="h5">
<div class="m_-6874691586421181616moz-cite-prefix">On 06/28/2017 12:47 PM, Alex
Megalokonomos wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr"><img class="m_-6874691586421181616mailtrack-img" style="float:right" alt="" src="data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7" height="0" width="0">Hello,
<div><br>
</div>
<div>We have the following scenario: our office call center is
an Alcatel OmniPCX Office setup.</div>
<div><br>
</div>
<div>This handles most of our needs and also provides 4 SIP
extensions.</div>
<div><br>
</div>
<div>These are provided by what appears to be a Kamailio SIP
server v 3.2.2 (no webrtc or websockets support)</div>
<div><br>
</div>
<div>What we would like to do is set up an OpenSIPS instance to
handle WebRTC and proxy everything to this Kamailio SIP
server.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div><br>
<div>I think half of the configuration is this : <a href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank"></a><a class="m_-6874691586421181616moz-txt-link-freetext" href="http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1" target="_blank">http://www.opensips.org/<wbr>Documentation/Tutorials-<wbr>WebSocket-2-1</a></div>
<div><br>
</div>
<div>which I've already completed and indeed, clients can
register to opensips and chat/make calls over websockets
between them.</div>
<div><br>
</div>
<div>How do I go about proxying registrations/invites/etc to
the kamailio server instead?</div>
<div><br>
</div>
<div>best regards</div>
</div>
</div>
<br>
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<br>
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</pre>
</span></blockquote>
<br>
</div>
</blockquote></div><br></div>