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

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

rtmp webrtc

Last synced: 14 days ago
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rtmp to webrtc

• Host: GitHub
• URL: https://github.com/notedit/rtmp-to-webrtc
• Owner: notedit
• Created: 2018-03-31T12:51:04.000Z (almost 7 years ago)
• Default Branch: master
• Last Pushed: 2022-12-31T01:59:45.000Z (about 2 years ago)
• Last Synced: 2025-01-16T07:11:24.079Z (22 days ago)
• Topics: rtmp, webrtc
• Language: JavaScript
• Homepage:
• Size: 513 KB
• Stars: 216
• Watchers: 12
• Forks: 60
• Open Issues: 16
• Metadata Files:
  • Readme: README.md

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README

        
# 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 h264_mp4toannexb  -g 15 -keyint_min 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.