[Spce-user] webRTC in production

H Yavari hyavari at rocketmail.com
Sat Nov 22 01:16:19 EST 2014


Hi,
I did this configs,use_rtpproxy:   „Always with rtpptoxy as only ICE candidate“ 
rtcp_feedback:  „Prefer AVPF“ 
srtp_transcoding:    „Prefer SRTP“  

 but the result was not good. the call was being in the "Call Progress" until timeout and called party (soft phone: eyebeam) not went in the ringing state anymore.I attached the RTP logs too. plz help me to solve this issue.

Regards,H.Yavari

    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.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

    


   



  
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