TODO



  * Document setup of coturn
    + Enable options needed for WebRTC
      <https://github.com/coturn/coturn/wiki/turnserver#webrtc-usage>
  * Support adaptive streaming
    + Simulcast
      - HLS for MPEG4
      - DASH for WebM
    + Scalable Video Codec (spatial/temporal/quality)
      + DASH for WebM (VP9 only?)
    <https://webrtchacks.com/sfu-simulcast/>
    <http://www.rtcbits.com/2017/04/howto-implement-temporal-scalability.html>
  * Support MSE-aided sub-second streaming
    <https://stackoverflow.com/questions/5858936/html5-live-streaming/39750043#39750043>
    + Support client-side graceful fallback to HLS/DASH (or maybe RTMP)
      <https://www.quora.com/Which-protocol-is-best-for-a-video-live-streaming-from-a-server-to-an-Android-RTSP-RTMP-HTTP-or-something-else>
  * Support variety of streaming setups
    + stream live (e.g. IP cam) RTSP feeds to web clients
      <https://flashphoner.com/rtsp-demo-player>
    + stream WebRTC Screen Sharing session to webRTC clients
      <https://flashphoner.com/screen-sharing-from-a-web-browser>
    + host WebRTC 1-on-1 videochat session
      <https://flashphoner.com/webrtc-video-chat-in-a-browser>
    + grab webcam with WebRTC and stream to web clients
      <https://flashphoner.com/browser-online-broadcasting-from-a-web-cam>
    + accept RTMP Live Encoder streams and forward as RTMP feed
      <https://flashphoner.com/broadcasting-of-a-streaming-video-from-a-professional-video-capturing-device-live-encoder-via-the-rtmp-protocol>
    + proxy WebRTC to HLS broadcasting feed
      <https://flashphoner.com/live-broadcasting-of-a-webrtc-stream-to-hls/>
    + grab webcam with WebRTC and forward streams as RTMP broadcasting feed
      <https://flashphoner.com/webrtc-as-rtmp-re-publishing/>
  * Support WebRTC multi-party videochat session
      <https://flashphoner.com/webrtc-video-conferencing/>
    + Support SFU interactive limiting of concurrent streams
      <https://support.livestorm.co/article/21-invite-someone-on-stage-during-the-webinar>
    + Support SFU automated limiting of concurrent streams
      by routing only most noisy audio and corresponding video
    + Support MCU/SFU hybrid limiting of concurrent streams
      by down-mixing audio and route only video tied to most noisy audio
  * Support bridging with SIP
    + accept SIP calls and forward streams as RTMP feed
      <https://flashphoner.com/broadcasting-of-a-sip-call-to-rtmp-cdn>
    + accept SIP calls and stream to web clients
      <https://flashphoner.com/browser-based-web-telephone-with-the-sip-support>
    + initiate SIP calls from web clients
      <https://flashphoner.com/online-call-from-a-website-to-mobile-phones-and-sip-click-to-call-function>
  * Offer test tools
    + WebRTC Troubleshooter
      <https://caniuse.spreed.me/>
    + Interoperability
      <https://github.com/webrtc/KITE>
    + Simulcast
      <https://github.com/CoSMoSoftware/simulcast-testsuite>
    + <https://github.com/MyPureCloud/webrtc-troubleshooter>