[Spce-user] OnSip Sip.Js WebRTC Libraries and SipWise

[Tarak Patel]

HI,



I am using Sipwise CE 3. I have problem configuring sipwise to work with onsip sip.js libraries. So what i have seen and configured is sipwise settings :



1)

RTP -- Plain SDP

SRTP -- Transparent

AVP -- Transparent

WebRTC to WebRTC Fails(Normal Scenario Known to not work)



2)

RTP -- Additional ICE Client

SRTP -- Transparent / Prefer SRTP

AVP -- Transparent / Prefer AVPF

Result -- Call fails on invite



3)

RTP -- AS ICE Client Only

SRTP -- Transparent

AVP -- Transparent

Result -- Call fails on invite



4)

RTP -- AS ICE Client only

SRTP -- Prefer SRTP

AVP -- Prefer AVPF

Result -- Call establishes but fails on initiating media.

I am seeing following error:

Failed to parse SessionDescription. m=audio 30010 RTP/SAVPF media type> 111 103 9 0 8 106 105 13 126 Invalid value: media.(Caller End).



After Above Error, it sends Bye to Receiver.



I have working setup for Kamailio and WebSocket which works perfectly fine with onsip sip.js webrtc libraries.



I am Attaching The logs of the calls with both the setup i.e. Sipwise and Kamailio+WebSocket.



Can anyone guide me how to configure sipwise to have successful calls between WebRTC-to-WebRTC?



Also, I would like to know how to configure sipwise for calls between WebRTC-to-Legacy sip and Legacy-Sip-To-WebRTC?



Let me know if you guys require any more information.



Thanks,

Tarak Patel

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