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

[Vlad Paiu]

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
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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