[OpenSIPS-Users] WebRTC : Integration with opensips.org free VoIP service & Tutorial
vladpaiu at opensips.org
Fri Jul 5 14:14:57 CEST 2013
So when you go to 'web calls' and hit the 'Login' button, the app
successfully registers your SIP account against opensips.org and shows
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 ?
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 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.
>>  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
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