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     I am trying to integrate webrtc->kamailio->asterisk to call from web browser.

I am using kamailio configuration file from caruizdiaz and chrome browser with sipml5 and asterisk as media server. Till now I have achieved to call to pstn numbers through sip trunking from browser but there is no audio and in the log rtpengine is showing the following error message. "SRTP output wanted, but no crypto suite was negotiated"

I think it is the error where kamailio is not able to establish DTLS negotiation and audio packets are dropped.

My question is how to make DTLS negotiation successful, whether it is error from chrome side or asterisk? because I am using RTP/AVP profile to send media packets to asterisk.

I have included my log here kamailio-webrtc-log

THANKS IN ADVANCE.

dkakoti
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    Both peers should exchange SDP with Crypto Suite. Later those Crypto Suites will be used for DTLS handshake and symmetric algorithm will be used to decode RTP stream. Apparently there should not be any Audio packets until DTLS handshake passed. You can try to see what Crypto suites is passed via chrome://webrtc-internals. Maybe it can give a clue – Anton Nov 17 '14 at 16:49
  • @Anton thanks.I have checked DTLS handshake is passed do I also need to configure tls in asterisk side? – dkakoti Nov 18 '14 at 05:48
  • This may be bug from rtpengine because it crashes after accepting DTLS certificate "segfault at 7f00fc03b000 ip 00007f0109dab6d8 sp 00007f01021a8990 error 4 in libcrypto.so.1.0.0[7f0109d40000+1b1000]" – dkakoti Nov 26 '14 at 05:15

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