SRS4.0之RTMP轉WebRTC01 ---- 簡介

 

1.啟動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 這里需要配置外網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.代碼框架

 

 這里主要分為幾個部分：

1. rtmp推流到SRS
2. RTMP流轉為RTC流
3. RTC客戶端和SRS通過HTTP交互SDP信息
4. RTC客戶端通過RTP拉流