WIP: Resolve "Use webrtc for appointments"

Closes #152 (closed)

todos:

• fix size/position of user avatar images during call on mobile
• add and tests appropriate timeouts during sip connection establishment
• add ui elements indicating audio stream state
  • icon: active stream / inactive stream etc.
  • icon: muted state from application state
  • border indication on avatar: voice speaking
• look into echo cancellation / prevention
• add audio signals for changes like peer muted, ringing, hangup
• check browser compatibilities of used apis and possible optimization / polyfills
  • https://www.npmjs.com/package/webrtc-adapter
  • zone.js
• check usability and provided information
• create user documentation
• error handling with user feedback
• show more information about why the call ended (hangup or error)
• ability to dismiss while outgoing ring