[Spce-user] webRTC in production

[H Yavari]

Hi Thomas,
I using : "srtp_transcoding" = "Force RTP"  and "rtcp_feedback" = "Force AVP" other things are default of sipwise.Thanks for reply.

Regards,H. Yavari
    Hello Yavari,what are your current settings in „NAT and Media Flow Control“ for your sip user ?jssip should actually work.BRThomas
Am 20.11.2014 um 09:35 schrieb H Yavari <hyavari at rocketmail.com>:



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





  
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