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

[Stefano Pisani]

I tried to call a SIP URI but it do not seems to be working.
I used crome. The connection works but it cannot place the call.

s

Il 04/07/2013 15.55, Vlad Paiu ha scritto:
> 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
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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