SRS4.0之RTMP转WebRTC01 ---- 简介
1.启动SRS
./objs/srs -c conf/rtc.conf
配置文件:
listen 1935; max_connections 1000; daemon off; srs_log_tank console; http_server { enabled on; listen 8080; dir ./objs/nginx/html; } http_api { enabled on; listen 1985; } stats { network 0; } rtc_server { enabled on; # Listen at udp://8000 listen 8000; # # The $CANDIDATE means fetch from env, if not configed, use * as default. # # The * means retrieving server IP automatically, from all network interfaces, # @see https://github.com/ossrs/srs/wiki/v4_CN_RTCWiki#config-candidate 这里需要配置外网IP candidate 192.168.1.103; }
推流:
ffmpeg -re -i time.flv -vcodec copy -acodec copy -f flv -y rtmp://192.168.1.103/live/livestream
播放:
http:///192.168.1.103:8080/players/rtc_player.html
2.代码框架
这里主要分为几个部分:
- rtmp推流到SRS
- RTMP流转为RTC流
- RTC客户端和SRS通过HTTP交互SDP信息
- RTC客户端通过RTP拉流