[Spce-user] webRTC in production

[Thomas Odorfer]

That is strange - I will try to test that setup myself (probably not with eyebeam but with xlite - from the same company, but for free)


Am 22.11.2014 um 07:16 schrieb H Yavari <hyavari at rocketmail.com>:

> 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.
>> 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
>>> 
>>> 
>> 
> 
> 
> 

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