COBA-WEBRTC

simple demo / proof of concept of working webrtc

video-sender-receiver

there is two page: sender.html and receiver.html
user who open sender.html will be broadcasting/streaming its media
user who open receiver.html will be playing the broadcasted/streamed media
no signalling server used, the two user must use other means of communication for signalling (recommended to be in same room and use third party chat software to send the signals)

open in separate tab (one tab opens sender.html, other tab opens receiver.html)
go to tab sender.html, copy the offer (it should contains: {"sdp":"...", "type":"offer"})
go to tab receiver.html, paste the offer, then click offer
go to tab receiver.html, copy the answer (it should contains: {"type":"answer", "sdp": "..."})
go to tab sender.html, paste the answer, then click answer
go to tab sender.html, copy the candidate (it should contains: {"candidate":"...","sdpMid":"0","sdpMLineIndex":0})
go to tab receiver.html, paste the candidate, then click candidate

on most online tutorials, the signalling usually use websocket (or JS websocket library like socket.io), the purpose of this demo is to show that the signalling can use any kind of method (e.g.: copy paste)
it can be done on different computer connected to same LAN (since I have not tried using STUN/TURN)
the code used is inspired/modified from https://codelabs.developers.google.com/codelabs/webrtc-web and its code sample github https://github.com/googlecodelabs/webrtc-web

learn how to use and deploy STUN/TURN server (e.g.: use coturn in linux server)
learn fields/data exchanged in SDP

RTCPeerConnection has two tasks:

get local media conditions, such as resolution and codec capabilities. This is the metadata used for the offer-and-answer mechanism.
get potential network addresses for the app's host, known as candidates.