[Freeswitch-users] Create webrtc call with custom signaling

Hello 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?