rtmp-to-webrtc - rtmp to webrtc

  •        303

demo 部署在个人测试服务器上, 带宽有限, 如果挂了请通知我. 服务端部署在阿里云上, 延迟在1000毫秒内, gstreamer的转封装引入了300ms-500ms延迟(目测, 还没验证). 优化后整体延迟可以在500ms以内.

https://github.com/notedit/rtmp-to-webrtc#readme

Dependencies:

body-parser : ^1.18.2
execa : ^1.0.0
fluent-ffmpeg : ^2.1.2
get-port : ^4.0.0
medooze-media-server : ^0.46.0
node-media-server : ^1.3.0
semantic-sdp : ^3.4.0
string-format : ^2.0.0

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