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Diffstat (limited to 'TODO')
| -rw-r--r-- | TODO | 48 |
1 files changed, 48 insertions, 0 deletions
@@ -0,0 +1,48 @@ + * 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/> |