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notedit
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Description

rtmp to webrtc

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# 88,077
webrtc
unified
swig
golang
35 commits
# 323,957
Python
quasar-...
vue2
quasar
1 commit

rtmp-to-webrtc

基于RTMP-CDN和WebRTC的低延迟(500ms以内)直播系统

观看效果

https://rtmp-to-webrtc.dot.cc

demo 部署在个人测试服务器上, 带宽有限, 如果挂了请通知我.

如何工作

  • RTMP推流到CDN上, 需要进行编码参数和gop的参数调优
  • 边缘节点部署webrtc服务器
  • 用户访问一路视频流的时候, 边缘节点webrtc服务器去CDN进行拉流
  • 把rtmp流转封装为rtp, 喂给webrtc服务器

RTMP推流脚本

推流部分使用ffmpeg ``` ffmpeg -f lavfi -re -i color=black:s=640x480:r=15 -filter:v "drawtext=text='%{localtime:%T}':fontcolor=white:fontsize=80:x=20:y=20" \ -vcodec libx264 -tune zerolatency -preset ultrafast -bsf:v h264mp4toannexb -g 15 -keyintmin 15 -profile:v baseline -level 3.0 \ -pix_fmt yuv420p -r 15 -f flv rtmp://39.106.248.166/live/live

RTMP转封装RTP

此部分使用了gstreamer, 只所以用gstreamer是因为发现ffmpeg的转出来的rtp包, 有一定概率webrtc会解析失败, 还未找到具体原因

gst-launch-1.0 -v rtmpsrc location=rtmp://localhost/live/{stream} ! flvdemux ! h264parse ! \ rtph264pay config-interval=-1 pt={pt} ! udpsink host=127.0.0.1 port={port}

一些数据

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.

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