I am following http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and trying to test a call
sipml5 ----------->Opensips + rtpengine --------> SIP end point (Freeswitch) But I do not have any audio on both sides. I see this error at rtpengine log "SRTP output wanted, but no crypto suite was negotiated" Anyone tested this scenario positive?
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