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
