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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