[Spce-user] webRTC in production

[Thomas Odorfer]

Actually, I am using the following settings

In order to achieve transcoding/media adaptation with rtpengine, include rtpproxy for calling other VOIP phones/PSTN:
use_rtpproxy:   „Always with rtpptoxy as only ICE candidate“   (alternatively „as additional ICE candidate“ - this might however lead to problems when calling other softphones)

srtp_transcoding:    „Prefer SRTP“          (webrtc mandates DTLS SRTP as media protocol,  your original setting Force RTP prevents this)
rtcp_feedback:  „Prefer AVPF“     (AVPF is webrtc standard for audio/video profile control)

Maybe you try that with jssip first.




Am 20.11.2014 um 10:29 schrieb H Yavari <hyavari at rocketmail.com>:

> 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.
> 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
>> 
> 
> 
> 

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