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