[Freeswitch-users] webrtc client using sip.js and conference api interaction

[Jonathan Hunter]

Hi Guys,

Does anyone know best practice here to be able to interact with the
freeswitch conference API so I can mute users and list participants from a
web browser, ideally without using dtmf?

I have previously used Verto, but due to scale I am using
kamailio/rtpengine as a websockets gateway which then load balances
requests to freeswitch so its only dealing with SIP and RTP.

All works fine, I just want to now be able to kick users/list participants
and so on via the browser so wondered the best way to do this now I am now
using verto, as I used to subscribe to live array for that but now I am
using sip.js to fire SIP requests over websockets, how do I send conference
api commands and to the correct box as well if a farm of freeswitch servers
are in use.


Many thanks

Jon
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