Hi, I have a Freeswitch installation, fronted by a Kamailio proxy
I would like to configure Kamailio to bridge SIP between webRTC clients and Freeswitch. The first issue I hit is how to set up a test webRTC client to make calls into Kamailio/FS. Can anyone point me to a simple webRTC client I can use to make test webRTC/SIP calls into my setup? Thanks BTW: I'm not JS experienced (I work with Python, C#, C/C++). Kind regards, Andy
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