title: Sending WebRTC MediaStream over Websocket (RTP over HTTP/Websocket)
body: <p>WebRTC is, among other things, meant for real-time browser to browser media communication, but in my case it will be used for browser to server audio communication.</p>
  
  <p>From the information I've gathered, the MediaStream is transferred using RTP over UDP.</p>
  
  <p>This will require at least two additional ports apart from the protocol used for signalling, something I would like to avoid.</p>
  
  <p>Within WebRTC, is there any possibility to use RTP over Websocket instead of RTP over UDP so that I only need to use port 80 or 443?</p>