But be aware, this configuration is for peer2peer connections, not for dispatching!
Kamailio will send simple SIP packets to the media relay then.
Also I used different NAT-traversal mechanism for sip and ws traffic (different routes based on client's transport protocol).
Also
you'll maybe need to have different rtpengine flags for sip and ws -
remember that WebRTC MUST have SRTP, but I had some issues in
transfering the SRTP handshake in
sipphone<-->kamailio<-->freeswitch scheme, so on webrtc
connection my "incoming" rtpengine had RTP/AVP flag, and on outgoing
webrtc it MUST have RTP/SAVP
For usual SIP calls I also conveted everything to RTP/AVP.
So you'll need to know to which type of user - ws or tcp/udp you're calling to understand which type of RTP to send them.