[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[John Nash]

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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