Sip proxy on WebRTC

[Harifidy]

Hi all. 
I use this [url=http://doubango.org/sipml5]doubango.org/sipml5[/url] ( the project is on [url=http://github.com/DoubangoTelecom/sipml5]github.com/DoubangoTelecom/sipml5[/url] )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.
[i]
" The websocket server url is required only if you 're a developer and using your sip proxy gateway not publicly reachable"
[/i] on [url=http://doubango.org/sipml5/expert.html]doubango.org/sipml5/expert.html[/url]
So could you please guide me (socumentation, where to begin, configuration,...) about genesys sip proxy.
Thank you.

[cavagnaro]

  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.