Using freeswitch/wss/WebRtc/JsSip? Want to do interop testing audio/video? Demo WebRtc site avail.

[Bill Ross]

Anybody using freeswitch/wss/WebRtc/JsSip? Want to do interop testing audio/video? Two options. Use your own freeswitch / wss server, or, contact me for temporary extension credentials (my fs test system).  WebRtc client: https://www.rossco.org/JsSip_demo.htm

I am looking for collaboration in the following areas (the JsSip_demo, which works, pretty basic, good for testing). Go to www.rossco.org for contact information.

-multi SIP Lines (max 4)

-conference in ONE more caller

-hold / resume

-video / audio on/off while in call

-update from jssip-3.0.27.min.js to most appropriate release

-pretty UI with selectable dialpad...

-provisioning using freeswitch methods

-in general, turn it into a business class multi-line phone

-if sufficient interest, will start project on GitHub

Regards;

Bill