It there a way to create a webrtc call with custom signaling? 

I want that when freeswitch receive a sip call, send it to the browser without registration or authentication

Freeswitch receive incoming call:

Create second leg for webrtc

Get sdp of the second leg

I will send it to the browser using custom signaling, 

Get answer sdp from the browser, 

Send the answer sdp to freeswitch 

Bridge both legs

I saw media_webrtc=true, but i did not understand how to receive the offer sdp and how to send the answer back to freeswitch using the api. 

Can i do it using esl? 

Any idea how to implement it? Is it possible?