Hello,
The free VoIP service offered by opensips.org has now been enhanced in
order to support WebRTC calls.
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.
Also, we have added a new tutorial, available at [2], which shows how to
add WebRTC capabilities to any existing OpenSIPS-based deployment.
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.
[1] https://www.opensips.org/account/
[2] http://www.opensips.org/Documentation/Tutorials-WebSocket
Best Regards,
--
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
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