[Spce-user] webRTC in production

[Thomas Odorfer]

Hello Yavari,
what are your current settings in „NAT and Media Flow Control“ for your sip user ?
jssip should actually work.
BR
Thomas

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
> 
> 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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