<div dir="ltr"><div>Thank you, I will take a close look at that.<br><br></div>Rion<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Mar 2, 2015 at 12:51 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</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">
Hello,<br>
<br>
Kamailio can be used to gateway signaling from webrtc/websocket to
other transports like UDP/TCP/TLS. See websocket module for an
example to start with and search of github for more advanced example
by Carlos Ruiz Diaz. Some talks were given at Kamailio World
Conference in the past that shows further examples.<br>
<br>
If you need to do gatewaying of the SRTP to RTP, then you have to
add RTPEngine as well.<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 28/02/15 22:13, Rion Carter wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div dir="ltr">
<div>I'm pretty new to SIP, RTP/SRTP, WebRTC and Websockets, so
I hope this question is coherent. I have a group of SIP
Softphones that need to connect to a
WebRTC/SIP-over-Websockets server. Can Kamailio be configured
to let me do this?</div>
<div><br>
</div>
<div>Any examples, tutorials or documentation would be
appreciated. I'm trying to determine how feasible this task
is. :)</div>
<div><br>
</div>
<div>Thanks!</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - <a href="http://www.kamailioworld.com" target="_blank">http://www.kamailioworld.com</a></pre>
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