Sip proxy on WebRTC

Hi all. :slight_smile:
I use this doubango.org/sipml5 ( the project is on github.com/DoubangoTelecom/sipml5 )to have a webRtc application.
I want to use my own sip proxy (genesys) instead of its. As the technical guide says, I have to provide a url to the “websocket server url” to manage that.

" The websocket server url is required only if you 're a developer and using your sip proxy gateway not publicly reachable"
on doubango.org/sipml5/expert.html
So could you please guide me (socumentation, where to begin, configuration,…) about genesys sip proxy.
Thank you.

??? ??? SIP Server is a Proxy as well.
But you need a WebRTC gateway first, TURN/STUN server…You need to read more about WebRTC architecture and components needed.