# all config for srs
#############################################################################################
# RTMP sections
#############################################################################################
# the rtmp listen ports, split by space.
listen 1935;
# the pid file
# to ensure only one process can use a pid file
# and provides the current running process id, for script,
# for example, init.d script to manage the server.
# default: ./objs/srs.pid
pid ./objs/srs.pid;
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# however, most clients supports it and it can improve
# performance about 10%.
# default: 60000
chunk_size 60000;
# the logs dir.
# if enabled ffmpeg, each stracoding stream will create a log file.
# default: ./objs/logs
ff_log_dir ./objs/logs;
# the log tank, console or file.
# if console, print log to console.
# if file, write log to file. requires srs_log_file if log to file.
# default: file.
srs_log_tank file;
# the log level, for all log tanks.
# can be: verbose, info, trace, warn, error
# defualt: trace
srs_log_level trace;
# when srs_log_tank is file, specifies the log file.
# default: ./objs/srs.log
srs_log_file ./objs/srs.log;
# the max connections.
# if exceed the max connections, server will drop the new connection.
# default: 2000
max_connections 1000;
# whether start as deamon
# default: on
daemon on;
#############################################################################################
# HTTP sections
#############################################################################################
# api of srs.
# the http api config, export for external program to manage srs.
# user can access http api of srs in browser directly, for instance, to access by:
# curl http://192.168.1.170:1985/api/v1/reload
# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
# where the cli can only be used in shell/terminate.
http_api {
# whether http api is enabled.
# default: off
enabled on;
# the http api port
# default: 1985
listen 1985;
}
# embeded http server in srs.
# the http streaming config, for HLS/HDS/DASH/HTTPProgressive
# global config for http streaming, user must config the http section for each vhost.
# the embed http server used to substitute nginx in ./objs/nginx,
# for example, srs runing in arm, can provides RTMP and HTTP service, only with srs installed.
# user can access the http server pages, generally:
# curl http://192.168.1.170:80/srs.html
# which will show srs version and welcome to srs.
# @remeark, the http embeded stream need to config the vhost, for instance, the __defaultVhost__
# need to open the feature http of vhost.
http_stream {
# whether http streaming service is enabled.
# default: off
enabled on;
# the http streaming port
# @remark, if use lower port, for instance 80, user must start srs by root.
# default: 8080
listen 8080;
# the default dir for http root.
# default: ./objs/nginx/html
dir ./objs/nginx/html;
}
#############################################################################################
# RTMP/HTTP VHOST sections
#############################################################################################
# vhost list, the __defaultVhost__ is the default vhost
# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
# for which cannot identify the required vhost.
vhost __defaultVhost__ {
}
# vhost for http
vhost http.srs.com {
# http vhost specified config
http {
# whether enable the http streaming service for vhost.
# default: off
enabled on;
# the virtual directory root for this vhost to mount at
# for example, if mount to /hls, user access by http://server/hls
# default: /
mount /hls;
# main dir of vhost,
# to delivery HTTP stream of this vhost.
# default: ./objs/nginx/html
dir ./objs/nginx/html/hls;
}
}
# vhost for atc.
vhost atc.srs.com {
# vhost for atc for hls/hds/rtmp backup.
# generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
# when atc is on, server delivery rtmp stream by absolute time.
# atc is used, for instance, encoder will copy stream to master and slave server,
# server use atc to delivery stream to edge/client, where stream time from master/slave server
# is always the same, client/tools can slice RTMP stream to HLS according to the same time,
# if the time not the same, the HLS stream cannot slice to support system backup.
#
# @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
# @see http://www.baidu.com/#wd=hds%20hls%20atc
#
# default: off
atc on;
}
# vhost for bandwidth check
# generally, the bandcheck vhost must be: bandcheck.srs.com,
# or need to modify the vhost of client.
vhost bandcheck.srs.com {
enabled on;
chunk_size 65000;
# bandwidth check config.
bandcheck {
# whether support bandwidth check,
# default: off.
enabled on;
# the key for server to valid,
# if invalid key, server disconnect and abort the bandwidth check.
key "35c9b402c12a7246868752e2878f7e0e";
# the interval in seconds for bandwidth check,
# server donot allow new test request.
# default: 30
interval 30;
# the max available check bandwidth in kbps.
# to avoid attack of bandwidth check.
# default: 1000
limit_kbps 4000;
}
}
# set the chunk size of vhost.
vhost chunksize.vhost.com {
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# vhost chunk size will override the global value.
# default: global chunk size.
chunk_size 128;
}
# the http hook callback vhost, srs will invoke the hooks for specified events.
vhost hooks.callback.vhost.com {
http_hooks {
# whether the http hooks enalbe.
# default off.
enabled on;
# when client connect to vhost/app, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_connect",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "pageUrl": "http://www.test.com/live.html"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client close/disconnect to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_close",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client(encoder) publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_publish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_publish http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client(encoder) stop publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_unpublish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_unpublish http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client start to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_play",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_play http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
# when client stop to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_stop",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_stop http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
}
}
# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
vhost mirror.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine mirror {
enabled on;
vfilter {
vf 'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
vhost drawtext.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine drawtext {
enabled on;
vfilter {
vf 'drawtext=text=SimpleRtmpServer(SRS):x=10:y=10:fontcolor=#cccccc:fontfile=./doc/FreeSerifBold.ttf';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
vhost crop.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine crop {
enabled on;
vfilter {
vf 'crop=in_w-20:in_h-160:10:80';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
vhost logo.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine logo {
enabled on;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# audio transcode only.
# for example, FMLE publish audio codec in mp3, and donot support HLS output,
# we can transcode the audio to aac and copy video to the new stream with HLS.
vhost audio.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine acodec {
enabled on;
vcodec copy;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# disable video, transcode/copy audio.
# for example, publish pure audio stream.
vhost vn.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine vn {
enabled on;
vcodec vn;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# ffmpeg-copy(forward implements by ffmpeg).
# copy the video and audio to a new stream.
vhost copy.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of vhost
vhost all.transcode.vhost.com {
# the streaming transcode configs.
transcode {
# whether the transcode enabled.
# if off, donot transcode.
# default: off.
enabled on;
# the ffmpeg
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
# the transcode engine for matched stream.
# all matched stream will transcoded to the following stream.
# the transcode set name(ie. hd) is optional and not used.
engine ffsuper {
# whether the engine is enabled
# default: off.
enabled on;
# ffmpeg filters, follows the main input.
vfilter {
# the logo input file.
i ./doc/ffmpeg-logo.png;
# the ffmpeg complex filter.
# for filters, @see: http://ffmpeg.org/ffmpeg-filters.html
filter_complex 'overlay=10:10';
}
# video encoder name. can be:
# libx264: use h.264(libx264) video encoder.
# copy: donot encoder the video stream, copy it.
# vn: disable video output.
vcodec libx264;
# video bitrate, in kbps
vbitrate 1500;
# video framerate.
vfps 25;
# video width, must be even numbers.
vwidth 768;
# video height, must be even numbers.
vheight 320;
# the max threads for ffmpeg to used.
vthreads 12;
# x264 profile, @see x264 -help, can be:
# high,main,baseline
vprofile main;
# x264 preset, @see x264 -help, can be:
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
vpreset medium;
# other x264 or ffmpeg video params
vparams {
# ffmpeg options, @see: http://ffmpeg.org/ffmpeg.html
t 100;
# 264 params, @see: http://ffmpeg.org/ffmpeg-codecs.html#libx264
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
# audio encoder name. can be:
# libaacplus: use aac(libaacplus) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
acodec libaacplus;
# audio bitrate, in kbps. [16, 72] for libaacplus.
abitrate 70;
# audio sample rate. for flv/rtmp, it must be:
# 44100,22050,11025,5512
asample_rate 44100;
# audio channel, 1 for mono, 2 for stereo.
achannels 2;
# other ffmpeg audio params
aparams {
# audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
profile:a aac_low;
}
# output stream. variables:
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffhd {
enabled on;
vcodec libx264;
vbitrate 1200;
vfps 25;
vwidth 1382;
vheight 576;
vthreads 6;
vprofile main;
vpreset medium;
vparams {
}
acodec libaacplus;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffsd {
enabled on;
vcodec libx264;
vbitrate 800;
vfps 25;
vwidth 1152;
vheight 480;
vthreads 4;
vprofile main;
vpreset fast;
vparams {
}
acodec libaacplus;
abitrate 60;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine fffast {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine vcopy {
enabled on;
vcodec copy;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine acopy {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all stream using the empty ffmpeg demo, donothing.
vhost ffempty.transcode.vhost.com {
transcode {
enabled on;
ffmpeg ./objs/research/ffempty;
engine empty {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of app
vhost app.transcode.vhost.com {
# the streaming transcode configs.
# if app specified, transcode all streams of app.
transcode live {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
# transcode specified stream.
vhost stream.transcode.vhost.com {
# the streaming transcode configs.
# if stream specified, transcode the matched stream.
transcode live/livestream {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
# the vhost which forward publish streams.
vhost same.vhost.forward.vhost.com {
# forward all publish stream to the specified server.
# this used to split/forward the current stream for cluster active-standby,
# active-active for cdn to build high available fault tolerance system.
# format: {ip}:{port} {ip_N}:{port_N}
# or specify the vhost by params, @see: change.vhost.forward.vhost.com
# if vhost not specified, use the request vhost instead.
forward 127.0.0.1:1936 127.0.0.1:1937;
}
# TODO: FIXME: change the forward schema with vhost.
# [plan] the vhost which forward publish streams to other vhosts.
vhost change.vhost.forward.vhost.com {
forward 127.0.0.1:1936 127.0.0.1:1937 {
# specify the vhost to override the vhost in client request.
vhost forward2.vhost.com;
# specify the refer(pageUrl) to override the refer in client request.
refer http://srs/index.html;
}
forward 127.0.0.1:1938 {
vhost forward3.vhost.com;
}
}
# the vhost disabled.
vhost removed.vhost.com {
# whether the vhost is enabled.
# if off, all request access denied.
# default: on
enabled off;
}
# the vhost with hls specified.
vhost with-hls.vhost.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled on;
# the hls output path.
# the app dir is auto created under the hls_path.
# for example, for rtmp stream:
# rtmp://127.0.0.1/live/livestream
# http://127.0.0.1/live/livestream.m3u8
# where hls_path is /hls, srs will create the following files:
# /hls/live the app dir for all streams.
# /hls/live/livestream.m3u8 the HLS m3u8 file.
# /hls/live/livestream-1.ts the HLS media/ts file.
# in a word, the hls_path is for vhost.
# default: ./objs/nginx/html
hls_path /data/nginx/html;
# the hls fragment in seconds, the duration of a piece of ts.
# default: 10
hls_fragment 10;
# the hls window in seconds, the number of ts in m3u8.
# default: 60
hls_window 60;
}
}
# the vhost with hls disabled.
vhost no-hls.vhost.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled off;
}
}
# the vhost for min delay, donot cache any stream.
vhost min.delay.com {
# whether cache the last gop.
# if on, cache the last gop and dispatch to client,
# to enable fast startup for client, client play immediately.
# if off, send the latest media data to client,
# client need to wait for the next Iframe to decode and show the video.
# set to off if requires min delay;
# set to on if requires client fast startup.
# default: on
gop_cache off;
# the max live queue length in seconds.
# if the messages in the queue exceed the max length,
# drop the old whole gop.
# default: 30
queue_length 10;
}
# the vhost for antisuck.
vhost refer.anti_suck.com {
# the common refer for play and publish.
# if the page url of client not in the refer, access denied.
# if not specified this field, allow all.
# default: not specified.
refer github.com github.io;
# refer for publish clients specified.
# the common refer is not overrided by this.
# if not specified this field, allow all.
# default: not specified.
refer_publish github.com github.io;
# refer for play clients specified.
# the common refer is not overrided by this.
# if not specified this field, allow all.
# default: not specified.
refer_play github.com github.io;
}
# config for the pithy print,
# which always print constant message specified by interval,
# whatever the clients in concurrency.
pithy_print {
# shared print interval for all publish clients, in milliseconds.
# if not specified, set to 1100.
publish 2000;
# shared print interval for all play clients, in milliseconds.
# if not specified, set to 1300.
play 3000;
# shared print interval for all forwarders, in milliseconds.
# if not specified, set to 2000.
forwarder 3000;
# shared print interval for all encoders, in milliseconds.
# if not specified, set to 2000.
encoder 3000;
# shared print interval for all hls, in milliseconds.
# if not specified, set to 2000.
hls 3000;
}