[OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial
stefano.pisani at omnianet.it
Fri Jul 5 14:25:35 CEST 2013
yes, it show connected.
I'm trying to call an external SIP URI.
It works only inside @opensips.org?
Il 05/07/2013 14.14, Vlad Paiu ha scritto:
> 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 ?
> Vlad Paiu
> OpenSIPS Developer
> 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.
>> Il 04/07/2013 15.55, Vlad Paiu ha scritto:
>>> 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
>>> 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.
>>>  https://www.opensips.org/account/
>>>  http://www.opensips.org/Documentation/Tutorials-WebSocket
>>> Best Regards,
>>> Vlad Paiu
>>> OpenSIPS Developer
>>> Users mailing list
>>> Users at lists.opensips.org
>> Users mailing list
>> Users at lists.opensips.org
> Users mailing list
> Users at lists.opensips.org
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