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Hello,<br>
<br>
The free VoIP service offered by opensips.org has now been enhanced
in order to support WebRTC calls.<br>
In order to test it, you can login to your account at [1] and go to
'web calls' in the left menu. The integrated client supports both
audio and video calls between two parties. <br>
<br>
Also, we have added a new tutorial, available at [2], which shows
how to add WebRTC capabilities to any existing OpenSIPS-based
deployment.<br>
The tutorial makes use of an OpenSIPS deployment with NAT support,
and adds WebRTC capabilities on top of that by using OverSIP as a WS
to SIP gateway and sipML5 as the web client.<br>
<br>
[1]
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<a href="https://www.opensips.org/account/">https://www.opensips.org/account/</a><br>
[2]
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<a href="http://www.opensips.org/Documentation/Tutorials-WebSocket">http://www.opensips.org/Documentation/Tutorials-WebSocket</a><br>
<br>
Best Regards,<br>
<pre class="moz-signature" cols="72">--
Vlad Paiu
OpenSIPS Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
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