Re: [SR-Users] NO_DTLS_FINGERPRINT when calling from webrtc to webrtc client

Hello, I would add the RTP-WebRTC gateway between SIP Kamailio and SIP UAC, from resources point of view, it is the only leg that needs encryption/decryption. Otherwise, you can try to work with two rtpengine instances (sets) in WS Kamailio, one to use for ws client to proxy and the other one for the leg from proxy to ws client. It will be a communication between them with classic rtp, both having towards ws client webrtc. It has the drawback of decryption and encryption done two times for the same call. You would need to add rtpengine set id in record-route to be able to handle properly the re-INVITE, BYE, etc. Another option that I would use is to send a negative reply from SIP kamailio, catch that in failure_route in WS Kamailio and engace there the rtpengine with proper flags. E.g., you assume it is going to be webrtc-to-webrtc, so no encryption/decryption added first time invite comes from WS client. You forward to SIP kamailio, which based on location, if it discovers that the callee is classic SIP-RTP, will send a 4xx back to WS Kamailio -- you end previous rtpengine session and engage it again with new flags (use branch route for managing rtpengine -- like it is done in default kamailio.cfg for rtpproxy). Cheers, Daniel On 15/09/14 20:30, Paweł Sternal wrote: -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany