[OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial

Vlad Paiu vladpaiu at opensips.org
Thu Jul 4 15:55:03 CEST 2013


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

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