Rtmp

https://github.com/ossrs/srs https://github.com/ossrs/srs/wiki/v2_CN_Home https://github.com/ossrs/srs/wiki/v1_CN_SampleRTMP https://hub.docker.com/search/?isAutomated=0&isOfficial=0&page=1&pullCount=0&q=srs&starCount=0 http://ossrs.net/srs.release/releases/ ====低延时配置====== https://github.com/ossrs/srs/wiki/v2_CN_LowLatency 考虑GOP-Cache和累积延迟，推荐的低延时配置如下（参考min.delay.com）： # the listen ports, split by space. listen 1935; vhost __defaultVhost__ { gop_cache off; queue_length 10; min_latency on; mr { enabled off; } mw_latency 100; tcp_nodelay on; } 当然，服务器的性能也要考虑，不可以让一个SRS进程跑太高带宽，一般CPU在80%以下不会影响延迟，连接数参考性能。

https://www.hongweipeng.com/index.php/archives/835/ http://www.cnblogs.com/lingyunhu/p/rtc28.html https://jitsi.org/Projects/JitsiVideobridge https://www.youtube.com/watch?v=OT6kwzsQeVc&feature=youtu.be https://my.oschina.net/langhuihui/blog/598338 https://my.oschina.net/langhuihui/blog/602265 https://my.oschina.net/langhuihui/blog/612144

http://flashphoner.com/broadcasting-of-a-streaming-video-from-a-professional-video-capturing-device-live-encoder-via-the-rtmp-protocol/ 這套大概是目前可以把rtmp轉成websocket和webrtc https://wcs5-eu.flashphoner.com/dashboard.xhtml 這是他的deom，Streamer就是把攝像頭和聲音轉成html5 (websocket、webRTC)能接收的 有下載 web call server 5.0 但要序號，問官方，說沒試用 ============ 另一個可能 Monaserver http://www.monaserver.ovh/

https://github.com/sbidolach/video-live-streaming