[Spce-user] webRTC in production

H Yavari hyavari at rocketmail.com
Thu Nov 20 03:35:51 EST 2014


Hi,I could to solve this issue. someone that had this problem said that jssip should run on Apache, so I did my test with sipml5 and now call will be established but there is no voice (RTP) and after 30 sec call terminated that I think is because for RTP timeout.So are there any configs that I should do ?Thanks.


Regards,H.Yavari

      From: H Yavari <hyavari at rocketmail.com>
 To: "spce-user at lists.sipwise.com" <spce-user at lists.sipwise.com> 
 Sent: Thursday, 20 November 2014, 10:12:08
 Subject: Re: [Spce-user] webRTC in production
   
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
> 

   
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