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v3_CN_Streamer
HOME > CN > Stream Caster
Note: 如果觉得Github的Wiki访问太慢,可以访问 Gitee 镜像。
Streamer是SRS作为服务器侦听并接收其他协议的流(譬如RTSP,MPEG-TS over UDP等等),将这些协议的流转换成RTMP推送给自己,以使用RTMP/HLS/HTTP分发流。
常见的应用场景包括:
- Push MPEG-TS over UDP to SRS:通过UDP协议,将MPEG-TS推送到SRS,分发为RTMP/HLS/HTTP流。
- Push RTSP to SRS:通过RTSP协议,将流推送到SRS,分发为RTMP/HLS/HTTP流。
- POST FLV over HTTP to SRS: 通过HTTP协议,将FLV流POST到SRS,分发为RTMP/HLS/HTTP流。
备注:Streamer将其他支持的协议推送RTMP给SRS后,所有SRS的功能都能支持。譬如,推RTSP流给Streamer,Streamer转成RTMP推送给SRS,若vhost是edge,SRS将RTMP流转发给源站。或者将RTMP流转码,或者直接转发。另外,所有分发方法都是可用的,譬如推RTSP流给Streamer,Streamer转成RTMP推给SRS,以RTMP/HLS/HTTP分发。
编译SRS时打开StreamCaster支持,参考Build:
./configure --with-stream-caster
目前Streamer支持的协议包括:
- MPEG-TS over UDP:已支持,可使用FFMPEG或其他编码器
push MPEG-TS over UDP
到SRS上。 - Push RTSP to SRS:已支持,可以使用FFMPEG或其他编码器
push rtsp to SRS
。 - POST FLV over HTTP to SRS: 已支持。
The config for stream casters:
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
stream_caster {
# whether stream caster is enabled.
# default: off
enabled off;
# the caster type of stream, the casters:
# mpegts_over_udp, MPEG-TS over UDP caster.
# rtsp, Real Time Streaming Protocol (RTSP).
# flv, FLV over HTTP POST.
caster mpegts_over_udp;
# the output rtmp url.
# for mpegts_over_udp caster, the typically output url:
# rtmp://127.0.0.1/live/livestream
# for rtsp caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, the rtsp url:
# rtsp://192.168.1.173:8544/live/livestream.sdp
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
# for rtsp caster, listen at tcp port. for example, 554.
# for flv caster, listen at tcp port. for example, 8936.
# TODO: support listen at <[ip:]port>
listen 8935;
# for the rtsp caster, the rtp server local port over udp,
# which reply the rtsp setup request message, the port will be used:
# [rtp_port_min, rtp_port_max)
rtp_port_min 57200;
rtp_port_max 57300;
}
SRS可以侦听一个udp端口,编码器将流推送到这个udp端口(SPTS)后,SRS会转成一路RTMP流。后面RTMP流能支持的功能都支持。
配置如下,参考conf/push.mpegts.over.udp.conf
:
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
stream_caster {
enabled on;
caster mpegts_over_udp;
output rtmp://127.0.0.1/live/livestream;
listen 1935;
}
参考:https://github.com/ossrs/srs/issues/250#issuecomment-72321769
这个功能被标记为了过时(Deprecated),并且会在未来删除,详细原因参考#2304。
SRS可以侦听一个HTTP端口,编码器将流推送到这个http端口后,SRS会转成一路RTMP流。所有RTMP流的功能都能支持。
配置如下,参考conf/push.flv.conf
:
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
stream_caster {
enabled on;
caster flv;
output rtmp://127.0.0.1/[app]/[stream];
listen 8936;
}
这个配置时,客户端推流的地址,例如:http://127.0.0.1:8936/live/sea.flv
播放RTMP流地址是:rtmp://127.0.0.1/live/sea
播放HLS流地址是:http://127.0.0.1:8080/live/sea.m3u8
注意:需要配置HTTP服务器和HLS,参考conf/push.flv.conf
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