Hi Vlad,
yes, it show connected.
I'm trying to call an external SIP URI.
It works only inside @opensips.org?
Thanks
Stefano
Il 05/07/2013 14.14, Vlad Paiu ha scritto:
Hello,
So when you go to 'web calls' and hit the 'Login' button, the app
successfully registers your SIP account against opensips.org and shows
'Connected' ?
If yes, the username that you are trying to call, is it also logged in
on the website, in the 'web calls' section ? If it's using a regular
soft/hard phone, does the phone have webRTC capabilities ?
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 07/04/2013 07:35 PM, Stefano Pisani wrote:
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
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