[Spce-user] webRTC in production

[H Yavari]

Hi,I installed the m3.6.1 and now I can registered my sip user from browser. But now when I create a call, I receive "User Denied Media Access"and call not established.
I changed the "srtp_transcoding" to Force RTP but error not changed. 
How can I solve this issue?Thanks.

Regards,H.Yavari
   
Jssip (like http://tryit.jssip.net/) work with wss URLs too. Use the
ones I specified earlier in the thread.

Andreas

On 11/19/2014 12:32 PM, H Yavari wrote:
> Hi,
> 
> Thanks. But my problem is that clients like jssip only support ws://<ip>.
> after upgrading to mr3.6.1, what address I should use for WS URI ????
> (WS URI is mandatory field)
> 
> 
> Regards,
> H. Yavari
> 
> ------------------------------------------------------------------------
> **
> Hi,
> 
> Use the latest version, which is mr3.6.1. Older version don't support
> the /wss/sip url.
> 
> Andreas
> 


   
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sipwise.com/pipermail/spce-user_lists.sipwise.com/attachments/20141120/4cc0ed60/attachment-0001.html>