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. On Apr 1, 2014 11:52 AM, "ik" idokan@gmail.com wrote:
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.
I wish to create/use Kamailio rules that will translate UDP to websockets and vice versa.
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).
Is there a way to make Kamailio a broker that understand both transports, and translate them ?
If so, can you please point me to a documentation/example that does it that might help me better understand it ?
Please note that I do not have any experience with Kamailio, and just getting started with it.
Thank you very much,
Ido
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