You might need to also add asterisk 12 b2b in order to convert to simple sip to solve issues with ice on the same box.
IdoPlease note that I do not have any experience with Kamailio, and just getting started with it.Is there a way to make Kamailio a broker that understand both transports, and translate them ?I have found few examples over the internet, but as it seems (to me), they are just doing normal SIP operations under websockets (registration, routing, voicemails etc).I wish to create/use Kamailio rules that will translate UDP to websockets and vice versa.Hello,I'm a newbie with Kamailio, and I require to connect webrtc (websockets) based phones, into a VoIP PBX that does not support websockets.
If so, can you please point me to a documentation/example that does it that might help me better understand it ?
Thank you very much,
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