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353eb9e2 0 parents master ... 2.0+_r3539_8d411235 2.0+_r3538_9354d4a9 2.0+_r3533_f03198c3 2.0+_r3532_a0bf6138 2.0+_r3531_100cac70 2.0+_r3530_155f68f3 2.0+_r3529_44530717

ver:2.0+ r3529 44530717 2017-04-27 18:46:34 

update srs2.0release to commit: ff87318b95ec8282ed17e2782724c5fa51f1b61f
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正在显示 42 个修改的文件 包含 2604 行增加 和 0 行删除
# bandwidth test tool config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
}
vhost bandcheck.srs.com {
enabled on;
chunk_size 65000;
bandcheck {
enabled on;
key "35c9b402c12a7246868752e2878f7e0e";
interval 30;
limit_kbps 4000;
}
}
... ...
# no-daemon and write log to console config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_api {
enabled on;
listen 1985;
}
http_server {
enabled on;
listen 8080;
}
vhost __defaultVhost__ {
}
... ...
# the config for srs demo
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
# @see full.conf for detail config.
listen 19350;
max_connections 1000;
daemon on;
srs_log_tank file;
srs_log_file ./objs/srs.demo.19350.log;
pid ./objs/srs.demo.19350.pid;
vhost __defaultVhost__ {
enabled on;
gop_cache on;
hls {
enabled on;
hls_path ./objs/nginx/html/forward;
hls_fragment 5;
hls_window 30;
}
}
... ...
# the config for srs demo
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon on;
srs_log_tank file;
srs_log_file ./objs/srs.demo.log;
pid ./objs/srs.demo.pid;
chunk_size 60000;
max_connections 2000;
http_api {
enabled on;
listen 1985;
}
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
enabled on;
gop_cache on;
}
vhost bandcheck.srs.com {
enabled on;
chunk_size 65000;
bandcheck {
enabled on;
key "35c9b402c12a7246868752e2878f7e0e";
interval 30;
limit_kbps 4000;
}
}
vhost demo.srs.com {
chunk_size 60000;
enabled on;
gop_cache on;
queue_length 30;
forward 127.0.0.1:19350;
bandcheck {
enabled off;
}
hls {
enabled on;
hls_path ./objs/nginx/html;
hls_fragment 5;
hls_window 30;
}
http_hooks {
enabled on;
on_connect http://127.0.0.1:8085/api/v1/clients;
on_close http://127.0.0.1:8085/api/v1/clients;
on_publish http://127.0.0.1:8085/api/v1/streams;
on_unpublish http://127.0.0.1:8085/api/v1/streams;
on_play http://127.0.0.1:8085/api/v1/sessions;
on_stop http://127.0.0.1:8085/api/v1/sessions;
}
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ld {
enabled on;
vfilter {
i ./doc/srs-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 1;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine sd {
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 500;
vfps 20;
vwidth 768;
vheight 320;
vthreads 1;
vprofile main;
vpreset fast;
vparams {
}
acodec libfdk_aac;
abitrate 40;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
ingest {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
vhost players {
enabled on;
gop_cache on;
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine hls {
enabled on;
vfilter {
i ./doc/srs-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 1;
vprofile baseline;
vpreset superfast;
vparams {
g 100;
}
acodec libfdk_aac;
abitrate 30;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=players_pub/[stream];
}
}
ingest {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/demo;
}
}
}
vhost players_pub {
hls {
enabled on;
hls_path ./objs/nginx/html;
hls_fragment 5;
hls_window 30;
}
}
vhost players_chat {
gop_cache off;
hls {
enabled off;
}
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine chat_hls {
enabled on;
vfilter {
}
vcodec libx264;
vbitrate 150;
vfps 15;
vwidth 576;
vheight 240;
vthreads 1;
vprofile baseline;
vpreset ultrafast;
vparams {
g 30;
}
acodec an;
output rtmp://127.0.0.1:[port]/[app]?vhost=chat_hls/[stream];
}
}
}
vhost chat_hls {
hls {
enabled on;
hls_path ./objs/nginx/html;
hls_fragment 2;
hls_window 10;
}
}
... ...
# the config for srs to dvr in custom path.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_EN_DVR#custom-path
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
dvr {
enabled on;
dvr_path ./objs/nginx/html/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
dvr_plan segment;
dvr_duration 30;
dvr_wait_keyframe on;
}
}
... ...
# the config for srs to dvr in segment mode
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
dvr {
enabled on;
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
dvr_plan segment;
dvr_duration 30;
dvr_wait_keyframe on;
}
}
... ...
# the config for srs to dvr in session mode
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
dvr {
enabled on;
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
dvr_plan session;
}
}
... ...
# the config for srs origin-edge cluster
# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
pid objs/edge.pid;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
mode remote;
origin 127.0.0.1:19350;
}
... ...
# the config for srs for token traverse authentication
# @see https://github.com/ossrs/srs/wiki/v1_CN_DRM
# @see full.conf for detail config.
listen 1935
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
mode remote;
origin 127.0.0.1:19350;
token_traverse on;
}
... ...
# the config for srs use ffmpeg to transcode
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleFFMPEG
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vfilter {
}
vcodec libx264;
vthreads 4;
vprofile main;
vpreset medium;
vparams {
}
acodec libfdk_aac;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
... ...
# the config for srs to forward
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
pid ./objs/srs.master.pid;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
forward 127.0.0.1:19350;
#the forward destination server type,default true. if the server is not srs server,connect app maybe fail,try config as false
forward_server_srs true;
}
... ...
# the config for srs to forward
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
# @see full.conf for detail config.
listen 19350;
max_connections 1000;
pid ./objs/srs.slave.pid;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
}
... ...
# all config for srs
#############################################################################################
# RTMP sections
#############################################################################################
# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
# for example, 192.168.1.100:1935 10.10.10.100:1935
# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
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.
# /dev/null to disable the log.
# default: ./objs
ff_log_dir ./objs;
# 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
# default: 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: 1000
max_connections 1000;
# whether start as daemon
# @remark: donot support reload.
# default: on
daemon on;
# whether use utc_time to generate the time struct,
# if off, use localtime() to generate it,
# if on, use gmtime() instead, which use UTC time.
# default: off
utc_time off;
# the work dir for server, to chdir(work_dir) when not empty or "./"
# user can config this directory to change the dir.
# @reamrk do not support reload.
# default: ./
work_dir ./;
# whether quit when parent process changed,
# used for supervisor mode(not daemon), srs should always quit when
# supervisor process exited.
# @remark conflict with daemon, error when both daemon and asprocess are on.
# @reamrk do not support reload.
# default: off
asprocess off;
#############################################################################################
# heartbeat/stats sections
#############################################################################################
# heartbeat to api server
# @remark, the ip report to server, is retrieve from system stat,
# which need the config item stats.network.
heartbeat {
# whether heartbeat is enalbed.
# default: off
enabled off;
# the interval seconds for heartbeat,
# recommend 0.3,0.6,0.9,1.2,1.5,1.8,2.1,2.4,2.7,3,...,6,9,12,....
# default: 9.9
interval 9.3;
# when startup, srs will heartbeat to this api.
# @remark: must be a restful http api url, where SRS will POST with following data:
# {
# "device_id": "my-srs-device",
# "ip": "192.168.1.100"
# }
# default: http://127.0.0.1:8085/api/v1/servers
url http://127.0.0.1:8085/api/v1/servers;
# the id of devide.
device_id "my-srs-device";
# whether report with summaries
# if on, put /api/v1/summaries to the request data:
# {
# "summaries": summaries object.
# }
# @remark: optional config.
# default: off
summaries off;
}
# system statistics section.
# the main cycle will retrieve the system stat,
# for example, the cpu/mem/network/disk-io data,
# the http api, for instance, /api/v1/summaries will show these data.
# @remark the heartbeat depends on the network,
# for example, the eth0 maybe the device which index is 0.
stats {
# the index of device ip.
# we may retrieve more than one network device.
# default: 0
network 0;
# the device name to stat the disk iops.
# ignore the device of /proc/diskstats if not configed.
disk sda sdb xvda xvdb;
}
#############################################################################################
# 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 listen entry is <[ip:]port>
# for example, 192.168.1.100:1985
# where the ip is optional, default to 0.0.0.0, that is 1985 equals to 0.0.0.0:1985
# default: 1985
listen 1985;
# whether enable crossdomain request.
# default: on
crossdomain on;
}
# 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.
# @remark, the http embeded stream need to config the vhost, for instance, the __defaultVhost__
# need to open the feature http of vhost.
http_server {
# whether http streaming service is enabled.
# default: off
enabled on;
# the http streaming listen entry is <[ip:]port>
# for example, 192.168.1.100:8080
# where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
# @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;
}
#############################################################################################
# Streamer sections
#############################################################################################
# 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;
}
stream_caster {
enabled off;
caster mpegts_over_udp;
output rtmp://127.0.0.1/live/livestream;
listen 8935;
}
stream_caster {
enabled off;
caster rtsp;
output rtmp://127.0.0.1/[app]/[stream];
listen 554;
rtp_port_min 57200;
rtp_port_max 57300;
}
stream_caster {
enabled off;
caster flv;
output rtmp://127.0.0.1/[app]/[stream];
listen 8936;
}
#############################################################################################
# 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__ {
}
# the security to allow or deny clients.
vhost security.srs.com {
# security for host to allow or deny clients.
# @see https://github.com/ossrs/srs/issues/211
security {
# whether enable the security for vhost.
# default: off
enabled on;
# the security list, each item format as:
# allow|deny publish|play all|<ip>
# for example:
# allow publish all;
# deny publish all;
# allow publish 127.0.0.1;
# deny publish 127.0.0.1;
# allow play all;
# deny play all;
# allow play 127.0.0.1;
# deny play 127.0.0.1;
# SRS apply the following simple strategies one by one:
# 1. allow all if security disabled.
# 2. default to deny all when security enabled.
# 3. allow if matches allow strategy.
# 4. deny if matches deny strategy.
allow play all;
allow publish all;
}
}
# the MR(merged-read) setting for publisher.
# the MW(merged-write) settings for player.
vhost mrw.srs.com {
# whether enable min delay mode for vhost.
# for min latence mode:
# 1. disable the mr for vhost.
# 2. use timeout for cond wait for consumer queue.
# @see https://github.com/ossrs/srs/issues/257
# default: off
min_latency off;
# about MR, read https://github.com/ossrs/srs/issues/241
mr {
# whether enable the MR(merged-read)
# default: off
enabled on;
# the latency in ms for MR(merged-read),
# the performance+ when latency+, and memory+,
# memory(buffer) = latency * kbps / 8
# for example, latency=500ms, kbps=3000kbps, each publish connection will consume
# memory = 500 * 3000 / 8 = 187500B = 183KB
# when there are 2500 publisher, the total memory of SRS atleast:
# 183KB * 2500 = 446MB
# the value recomment is [300, 2000]
# default: 350
latency 350;
}
# set the MW(merged-write) latency in ms.
# SRS always set mw on, so we just set the latency value.
# the latency of stream >= mw_latency + mr_latency
# the value recomment is [300, 1800]
# default: 350
mw_latency 350;
}
# vhost for edge, edge and origin is the same vhost
vhost same.edge.srs.com {
# the mode of vhost, local or remote.
# local: vhost is origin vhost, which provides stream source.
# remote: vhost is edge vhost, which pull/push to origin.
# default: local
mode remote;
# for edge(remote mode), user must specifies the origin server
# format as: <server_name|ip>[:port]
# @remark user can specifies multiple origin for error backup, by space,
# for example, 192.168.1.100:1935 192.168.1.101:1935 192.168.1.102:1935
origin 127.0.0.1:1935 localhost:1935;
# for edge, whether open the token traverse mode,
# if token traverse on, all connections of edge will forward to origin to check(auth),
# it's very important for the edge to do the token auth.
# the better way is use http callback to do the token auth by the edge,
# but if user prefer origin check(auth), the token_traverse if better solution.
# default: off
token_traverse off;
}
# vhost for edge, edge transform vhost to fetch from another vhost.
vhost transform.edge.srs.com {
mode remote;
origin 127.0.0.1:1935;
# the vhost to transform for edge,
# to fetch from the specified vhost at origin,
# if not specified, use the current vhost of edge in origin, the variable [vhost].
# default: [vhost]
vhost same.edge.srs.com;
}
# vhost for dvr
vhost dvr.srs.com {
# dvr RTMP stream to file,
# start to record to file when encoder publish,
# reap flv according by specified dvr_plan.
dvr {
# whether enabled dvr features
# default: off
enabled on;
# the dvr plan. canbe:
# session reap flv when session end(unpublish).
# segment reap flv when flv duration exceed the specified dvr_duration.
# append always append to flv file, never reap it.
# default: session
dvr_plan session;
# the dvr output path.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# [2006], replace this const to current year.
# [01], replace this const to current month.
# [02], replace this const to current date.
# [15], replace this const to current hour.
# [04], repleace this const to current minute.
# [05], repleace this const to current second.
# [999], repleace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# @remark we use golang time format "2006-01-02 15:04:05.999" as "[2006]-[01]-[02]_[15].[04].[05]_[999]"
# for example, for url rtmp://ossrs.net/live/livestream and time 2015-01-03 10:57:30.776
# 1. No variables, the rule of SRS1.0(auto add [stream].[timestamp].flv as filename):
# dvr_path ./objs/nginx/html;
# =>
# dvr_path ./objs/nginx/html/live/livestream.1420254068776.flv;
# 2. Use stream and date as dir name, time as filename:
# dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/livestream/2015/01/03/10.57.30.776.flv;
# 3. Use stream and year/month as dir name, date and time as filename:
# dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]-[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/livestream/2015/01/03-10.57.30.776.flv;
# 4. Use vhost/app and year/month as dir name, stream/date/time as filename:
# dvr_path /data/[vhost]/[app]/[2006]/[01]/[stream]-[02]-[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/2015/01/livestream-03-10.57.30.776.flv;
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_EN_DVR#custom-path
# segment,session apply it.
# default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
# the duration for dvr file, reap if exeed, in seconds.
# segment apply it.
# session,append ignore.
# default: 30
dvr_duration 30;
# whether wait keyframe to reap segment,
# if off, reap segment when duration exceed the dvr_duration,
# if on, reap segment when duration exceed and got keyframe.
# segment apply it.
# session,append ignore.
# default: on
dvr_wait_keyframe on;
# about the stream monotonically increasing:
# 1. video timestamp is monotonically increasing,
# 2. audio timestamp is monotonically increasing,
# 3. video and audio timestamp is interleaved monotonically increasing.
# it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
# however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
# the time jitter algorithm:
# 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
# 2. zero, only ensure sttream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# apply for all dvr plan.
# default: full
time_jitter full;
# on_dvr, never config in here, should config in http_hooks.
# for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
# @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
}
}
# vhost for ingest
vhost ingest.srs.com {
# ingest file/stream/device then push to SRS over RTMP.
# the name/id used to identify the ingest, must be unique in global.
# ingest id is used in reload or http api management.
ingest livestream {
# whether enabled ingest features
# default: off
enabled on;
# input file/stream/device
# @remark only support one input.
input {
# the type of input.
# can be file/stream/device, that is,
# file: ingest file specifies by url.
# stream: ingest stream specifeis by url.
# device: not support yet.
# default: file
type file;
# the url of file/stream.
url ./doc/source.200kbps.768x320.flv;
}
# the ffmpeg
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
# the transcode engine, @see all.transcode.srs.com
# @remark, the output is specified following.
engine {
# @see enabled of transcode engine.
# if disabled or vcodec/acodec not specified, use copy.
# default: off.
enabled off;
# output stream. variables:
# [vhost] current vhost which start the ingest.
# [port] system RTMP stream port.
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
# vhost for http static and flv vod stream for each vhost.
vhost http.static.srs.com {
# http static vhost specified config
http_static {
# whether enabled the http static service for vhost.
# default: off
enabled on;
# the url to mount to,
# typical mount to [vhost]/
# the variables:
# [vhost] current vhost for http server.
# @remark the [vhost] is optional, used to mount at specified vhost.
# @remark the http of __defaultVhost__ will override the http_server section.
# for example:
# mount to [vhost]/
# access by http://ossrs.net:8080/xxx.html
# mount to [vhost]/hls
# access by http://ossrs.net:8080/hls/xxx.html
# mount to /
# access by http://ossrs.net:8080/xxx.html
# or by http://192.168.1.173:8080/xxx.html
# mount to /hls
# access by http://ossrs.net:8080/hls/xxx.html
# or by http://192.168.1.173:8080/hls/xxx.html
# @remark the port of http is specified by http_server section.
# default: [vhost]/
mount [vhost]/hls;
# main dir of vhost,
# to delivery HTTP stream of this vhost.
# default: ./objs/nginx/html
dir ./objs/nginx/html/hls;
}
}
# vhost for http flv/aac/mp3 live stream for each vhost.
vhost http.remux.srs.com {
# http flv/mp3/aac/ts stream vhost specified config
http_remux {
# whether enable the http live streaming service for vhost.
# default: off
enabled on;
# the fast cache for audio stream(mp3/aac),
# to cache more audio and send to client in a time to make android(weixin) happy.
# @remark the flv/ts stream ignore it
# @remark 0 to disable fast cache for http audio stream.
# default: 0
fast_cache 30;
# the stream mout for rtmp to remux to live streaming.
# typical mount to [vhost]/[app]/[stream].flv
# the variables:
# [vhost] current vhost for http live stream.
# [app] current app for http live stream.
# [stream] current stream for http live stream.
# @remark the [vhost] is optional, used to mount at specified vhost.
# the extension:
# .flv mount http live flv stream, use default gop cache.
# .ts mount http live ts stream, use default gop cache.
# .mp3 mount http live mp3 stream, ignore video and audio mp3 codec required.
# .aac mount http live aac stream, ignore video and audio aac codec required.
# for example:
# mount to [vhost]/[app]/[stream].flv
# access by http://ossrs.net:8080/live/livestream.flv
# mount to /[app]/[stream].flv
# access by http://ossrs.net:8080/live/livestream.flv
# or by http://192.168.1.173:8080/live/livestream.flv
# mount to [vhost]/[app]/[stream].mp3
# access by http://ossrs.net:8080/live/livestream.mp3
# mount to [vhost]/[app]/[stream].aac
# access by http://ossrs.net:8080/live/livestream.aac
# mount to [vhost]/[app]/[stream].ts
# access by http://ossrs.net:8080/live/livestream.ts
# @remark the port of http is specified by http_server section.
# default: [vhost]/[app]/[stream].flv
mount [vhost]/[app]/[stream].flv;
# whether http stream trigger rtmp stream source when no stream available,
# for example, when encoder has not publish stream yet,
# user can play the http flv stream and wait for stream.
# default: on
hstrs on;
}
}
# the vhost with hls specified.
vhost with-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled on;
# the hls fragment in seconds, the duration of a piece of ts.
# default: 10
hls_fragment 10;
# the hls m3u8 target duration ratio,
# EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
# EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
# @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
# default: 1.5
hls_td_ratio 1.5;
# the audio overflow ratio.
# for pure audio, the duration to reap the segment.
# for example, the hls_fragment is 10s, hsl_aof_ratio is 2.0,
# the segemnt will reap to 20s for pure audio.
# default: 2.0
hls_aof_ratio 2.0;
# the hls window in seconds, the number of ts in m3u8.
# default: 60
hls_window 60;
# the error strategy. canbe:
# ignore, disable the hls.
# disconnect, require encoder republish.
# continue, ignore failed try to continue output hls.
# @see https://github.com/ossrs/srs/issues/264
# default: continue
hls_on_error continue;
# the hls output path.
# the m3u8 file is configed by hls_path/hls_m3u8_file, the default is:
# ./objs/nginx/html/[app]/[stream].m3u8
# the ts file is configed by hls_path/hls_ts_file, the default is:
# ./objs/nginx/html/[app]/[stream]-[seq].ts
# @remark the hls_path is compatible with srs v1 config.
# default: ./objs/nginx/html
hls_path ./objs/nginx/html;
# the hls m3u8 file name.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# default: [app]/[stream].m3u8
hls_m3u8_file [app]/[stream].m3u8;
# the hls ts file name.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# [2006], replace this const to current year.
# [01], replace this const to current month.
# [02], replace this const to current date.
# [15], replace this const to current hour.
# [04], repleace this const to current minute.
# [05], repleace this const to current second.
# [999], repleace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# [seq], the sequence number of ts.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
# default: [app]/[stream]-[seq].ts
hls_ts_file [app]/[stream]-[seq].ts;
# whether use floor for the hls_ts_file path generation.
# if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
# and use enahanced algorithm to calc deviation for segment.
# @remark when floor on, recommend the hls_segment>=2*gop.
# default: off
hls_ts_floor off;
# the hls entry prefix, which is base url of ts url.
# if specified, the ts path in m3u8 will be like:
# http://your-server/live/livestream-0.ts
# http://your-server/live/livestream-1.ts
# ...
# optional, default to empty string.
hls_entry_prefix http://your-server;
# the default audio codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for mp3.
# the available audio codec:
# aac, mp3, an
# default: aac
hls_acodec aac;
# the default video codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for pure audio(without video) to vn.
# the available video codec:
# h264, vn
# default: h264
hls_vcodec h264;
# whether cleanup the old expired ts files.
# default: on
hls_cleanup on;
# the timeout in seconds to dispose the hls,
# dispose is to remove all hls files, m3u8 and ts files.
# when publisher timeout dispose hls.
# @remark 0 to disable dispose for publisher.
# @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose hls.
# default: 0
hls_dispose 0;
# the max size to notify hls,
# to read max bytes from ts of specified cdn network,
# @remark only used when on_hls_notify is config.
# default: 64
hls_nb_notify 64;
# whether wait keyframe to reap segment,
# if off, reap segment when duration exceed the fragment,
# if on, reap segment when duration exceed and got keyframe.
# default: on
hls_wait_keyframe on;
# on_hls, never config in here, should config in http_hooks.
# for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
# @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
# on_hls_notify, never config in here, should config in http_hooks.
# we support the variables to generate the notify url:
# [app], replace with the app.
# [stream], replace with the stream.
# [ts_url], replace with the ts url.
# for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
# @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
}
}
# the vhost with hls disabled.
vhost no-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled off;
}
}
# the vhost with adobe hds
vhost hds.srs.com {
hds {
# whether hds enabled
# default: off
enabled on;
# the hds fragment in seconds.
# default: 10
hds_fragment 10;
# the hds window in seconds, erase the segment when exceed the window.
# default: 60
hds_window 60;
# the path to store the hds files.
# default: ./objs/nginx/html
hds_path ./objs/nginx/html;
}
}
# the http hook callback vhost, srs will invoke the hooks for specified events.
vhost hooks.callback.srs.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",
# "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
# "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",
# "send_bytes": 10240, "recv_bytes": 10240
# }
# 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",
# "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_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;
# when srs reap a dvr file, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_dvr",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream",
# "cwd": "/usr/local/srs",
# "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
# }
# 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
on_dvr http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
# when srs reap a ts file of hls, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_hls",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream",
# "duration": 9.36, // in seconds
# "cwd": "/usr/local/srs",
# "file": "./objs/nginx/html/live/livestream/2015-04-23/01/476584165.ts",
# "url": "live/livestream/2015-04-23/01/476584165.ts",
# "m3u8": "./objs/nginx/html/live/livestream/live.m3u8",
# "m3u8_url": "live/livestream/live.m3u8",
# "seq_no": 100
# }
# 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
on_hls http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
# when srs reap a ts file of hls, call this hook,
# used to push file to cdn network, by get the ts file from cdn network.
# so we use HTTP GET and use the variable following:
# [app], replace with the app.
# [stream], replace with the stream.
# [ts_url], replace with the ts url.
# ignore any return data of server.
# @remark random select a url to report, not report all.
on_hls_notify http://127.0.0.1:8085/api/v1/hls/[app]/[stream][ts_url];
}
}
# the vhost for srs debug info, whether send args in connect(tcUrl).
vhost debug.srs.com {
# when upnode(forward to, edge push to, edge pull from) is srs,
# it's strongly recommend to open the debug_srs_upnode,
# when connect to upnode, it will take the debug info,
# for example, the id, source id, pid.
# please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
# default: on
debug_srs_upnode on;
}
# the vhost for min delay, donot cache any stream.
vhost min.delay.com {
# @see vhost mrw.srs.com for detail.
min_latency on;
mr {
enabled off;
}
mw_latency 100;
# whether cache the last gop.
# if on, cache the last gop and dispatch to client,
# to enabled 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;
# whether enable the TCP_NODELAY
# if on, set the nodelay of fd by setsockopt
# default: off
tcp_nodelay on;
}
# whether disable the sps parse, for the resolution of video.
vhost no.parse.sps.com {
publish {
# whether parse the sps when publish stream.
# we can got the resolution of video for stat api.
# but we may failed to cause publish failed.
# default: on
parse_sps on;
}
}
# the vhost to control the stream delivery feature
vhost stream.control.com {
# @see vhost mrw.srs.com for detail.
min_latency on;
mr {
enabled off;
}
mw_latency 100;
# @see vhost min.delay.com
queue_length 10;
tcp_nodelay on;
# the minimal packets send interval in ms,
# used to control the ndiff of stream by srs_rtmp_dump,
# for example, some device can only accept some stream which
# delivery packets in constant interval(not cbr).
# @remark 0 to disable the minimal interval.
# @remark >0 to make the srs to send message one by one.
# @remark user can get the right packets interval in ms by srs_rtmp_dump.
# default: 0
send_min_interval 10.0;
# whether reduce the sequence header,
# for some client which cannot got duplicated sequence header,
# while the sequence header is not changed yet.
# default: off
reduce_sequence_header on;
# the 1st packet timeout in ms for encoder.
# default: 20000
publish_1stpkt_timeout 20000;
# the normal packet timeout in ms for encoder.
# default: 5000
publish_normal_timeout 7000;
}
# 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;
}
# the vhost which forward publish streams.
vhost same.vhost.forward.srs.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}
forward 127.0.0.1:1936 127.0.0.1:1937;
}
# the main comments for transcode
vhost example.transcode.srs.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 example {
# whether the engine is enabled
# default: off.
enabled on;
# input format, can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# default: flv
iformat flv;
# 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
# @remark 0 to use source video bitrate.
# default: 0
vbitrate 1500;
# video framerate.
# @remark 0 to use source video fps.
# default: 0
vfps 25;
# video width, must be even numbers.
# @remark 0 to use source video width.
# default: 0
vwidth 768;
# video height, must be even numbers.
# @remark 0 to use source video height.
# default: 0
vheight 320;
# the max threads for ffmpeg to used.
# default: 1
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:
# libfdk_aac: use aac(libfdk_aac) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
acodec libfdk_aac;
# audio bitrate, in kbps. [16, 72] for libfdk_aac.
# @remark 0 to use source audio bitrate.
# default: 0
abitrate 70;
# audio sample rate. for flv/rtmp, it must be:
# 44100,22050,11025,5512
# @remark 0 to use source audio sample rate.
# default: 0
asample_rate 44100;
# audio channel, 1 for mono, 2 for stereo.
# @remark 0 to use source audio channels.
# default: 0
achannels 2;
# other ffmpeg audio params
aparams {
# audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
# @remark SRS supported aac profile for HLS is: aac_low, aac_he, aac_he_v2
profile:a aac_low;
bsf:a aac_adtstoasc;
}
# output format, can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# default: flv
oformat flv;
# 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];
}
}
}
# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
vhost mirror.transcode.srs.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 libfdk_aac;
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
# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
#######################################################################################################
# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
vhost crop.transcode.srs.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 libfdk_aac;
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.srs.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 libfdk_aac;
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.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine acodec {
enabled on;
vcodec copy;
acodec libfdk_aac;
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.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine vn {
enabled on;
vcodec vn;
acodec libfdk_aac;
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.srs.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
# the comments, read example.transcode.srs.com
vhost all.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ffsuper {
enabled on;
iformat flv;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 1500;
vfps 25;
vwidth 768;
vheight 320;
vthreads 12;
vprofile main;
vpreset medium;
vparams {
t 100;
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
profile:a aac_low;
}
oformat flv;
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 libfdk_aac;
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 libfdk_aac;
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 libfdk_aac;
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 libfdk_aac;
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.srs.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 libfdk_aac;
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.srs.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.srs.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;
}
}
}
# 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.srs.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;
}
# vhost for time jitter
vhost jitter.srs.com {
# about the stream monotonically increasing:
# 1. video timestamp is monotonically increasing,
# 2. audio timestamp is monotonically increasing,
# 3. video and audio timestamp is interleaved/mixed monotonically increasing.
# it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
# however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
# the time jitter algorithm:
# 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
# 2. zero, only ensure sttream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# default: full
time_jitter full;
# whether use the interleaved/mixed algorithm to correct the timestamp.
# if on, always ensure the timestamp of audio+video is interleaved/mixed monotonically increase.
# if off, use time_jitter to correct the timestamp if required.
# default: off
mix_correct off;
}
# 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;
# whether enable the auto atc,
# if enabled, detect the bravo_atc="true" in onMetaData packet,
# set atc to on if matched.
# always ignore the onMetaData if atc_auto is off.
# default: on
atc_auto on;
}
# the vhost disabled.
vhost removed.srs.com {
# whether the vhost is enabled.
# if off, all request access denied.
# default: on
enabled off;
}
# config for the pithy print,
# which always print constant message specified by interval,
# whatever the clients in concurrency.
# default: 10000
pithy_print_ms 10000;
... ...
# the config for srs to delivery hds
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHDS
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
hds {
enabled on;
hds_fragment 10;
hds_window 60;
hds_path ./objs/nginx/html;
}
}
... ...
# the config for srs to delivery hls
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream].m3u8;
hls_ts_file [app]/[stream]-[seq].ts;
}
}
... ...
# the config for srs to delivery hls
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream]/index.m3u8;
hls_ts_file [app]/[stream]/[seq].ts;
#if not configured,don't save total m3u8 file,and keep the ts cleaned
hls_total_m3u8_file [app]/[stream]/total.m3u8;
#if not configured,don't save total duration file
hls_total_file [app]/[stream]/total;
}
}
... ...
# the config for srs to remux rtmp to aac live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
http_remux {
enabled on;
fast_cache 30;
mount [vhost]/[app]/[stream].aac;
hstrs on;
}
}
... ...
# the config for srs to remux rtmp to flv live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
hstrs on;
}
}
... ...
# the config for srs to remux rtmp to flv live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 19351;
max_connections 1000;
pid objs/srs.flv.19351.pid;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8081;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
mode remote;
origin 127.0.0.1;
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
hstrs on;
}
}
... ...
# the config for srs to remux rtmp to flv live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 19352;
max_connections 1000;
pid objs/srs.flv.19352.pid;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8082;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
mode remote;
origin 127.0.0.1;
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
hstrs on;
}
}
... ...
# the config for srs http heartbeat, report its info to api-server
# @see full.conf for detail config.
listen 1935
max_connections 1000;
daemon off;
srs_log_tank console;
heartbeat {
enabled on;
interval 9.3;
# for python api-server
url http://127.0.0.1:8085/api/v1/servers;
device_id "my-srs-device";
# for ossrs.net monitor, device_id is the key genereated by bsm.
#url http://www.ossrs.net:1977/api/v1/robots/servers;
#device_id "35c9b402c12a7246868752e2878f7e0e";
# with detail summaries
summaries on;
}
stats {
network 0;
disk sda sdb xvda xvdb;
}
vhost __defaultVhost__ {
}
... ...
# the config for srs to delivery hls
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream].m3u8;
hls_ts_file [app]/[stream]-[seq].ts;
}
}
... ...
# http-hooks or http-callbacks config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
http_hooks {
enabled on;
on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
}
}
... ...
# the config for srs to remux rtmp to mp3 live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
http_remux {
enabled on;
fast_cache 30;
mount [vhost]/[app]/[stream].mp3;
hstrs on;
}
}
... ...
# the config for srs to serve as http server
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost ossrs.net {
http {
enabled on;
mount [vhost]/;
dir ./objs/nginx/html;
}
}
... ...
# the config for srs to remux rtmp to ts live stream.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
vhost __defaultVhost__ {
http_remux {
enabled on;
mount [vhost]/[app]/[stream].ts;
hstrs on;
}
}
... ...
# use ffmpeg to ingest file/stream/device to SRS
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleIngest
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
ingest livestream {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
... ...
# no-daemon and write log to console config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_api {
enabled on;
listen 1985;
}
http_server {
enabled on;
listen 8080;
}
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
}
ingest livestream {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
#ffmpeg ./objs/ffmpeg/bin/ffmpeg;
ffmpeg ./objs/research/librtmp/srs_ingest_flv;
engine {
enabled off;
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
... ...
# the config for srs origin-edge cluster
# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
# @see full.conf for detail config.
listen 19350;
max_connections 1000;
daemon off;
srs_log_tank console;
pid ./objs/origin.pid;
vhost __defaultVhost__ {
}
... ...
# push HTTP FLV to SRS.
# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-http-flv-to-srs
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
stream_caster {
enabled on;
caster flv;
output rtmp://127.0.0.1/[app]/[stream];
listen 8936;
}
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream].m3u8;
hls_ts_file [app]/[stream]-[seq].ts;
}
}
... ...
# push MPEG-TS over UDP to SRS.
# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-mpeg-ts-over-udp
# @see https://github.com/ossrs/srs/issues/250#issuecomment-72321769
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
stream_caster {
enabled on;
caster mpegts_over_udp;
output rtmp://127.0.0.1/live/livestream;
listen 1935;
}
vhost __defaultVhost__ {
}
... ...
# push MPEG-TS over UDP to SRS.
# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-mpeg-ts-over-udp
# @see https://github.com/ossrs/srs/issues/250#issuecomment-72321769
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
stream_caster {
enabled on;
caster rtsp;
output rtmp://127.0.0.1/[app]/[stream];
listen 554;
rtp_port_min 57200;
rtp_port_max 57300;
}
vhost __defaultVhost__ {
}
... ...
# the config for srs to delivery realtime RTMP stream
# @see https://github.com/ossrs/srs/wiki/v2_CN_SampleRealtime
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
gop_cache off;
queue_length 10;
min_latency on;
mr {
enabled off;
}
mw_latency 100;
tcp_nodelay on;
}
... ...
# the config for srs to delivery RTMP
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleRTMP
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
}
... ...
# security config for srs, allow play and deny publish.
# @see https://github.com/ossrs/srs/issues/211#issuecomment-68507035
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
security {
enabled on;
deny publish all;
allow play all;
}
}
... ...
# main config for srs.
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
srs_log_tank file;
srs_log_file ./objs/srs.log;
http_api {
enabled on;
listen 1985;
}
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
stats {
network 0;
disk sda sdb xvda xvdb;
}
vhost __defaultVhost__ {
}
... ...
# the config for srs to delivery hls
# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream].m3u8;
hls_ts_file [app]/[stream]-[seq].ts;
}
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vcodec copy;
acodec libfdk_aac;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
... ...
# the config for srs origin-edge cluster
# @see https://github.com/ossrs/srs/wiki/v2_CN_Edge#transform-vhost
# @see full.conf for detail config.
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
}
vhost source.srs.com {
ingest livestream {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
vhost transform.srs.edge.com {
mode remote;
origin 127.0.0.1:1935;
vhost source.srs.com;
}
... ...
#!/bin/bash
### BEGIN INIT INFO
# Provides: ossrs(srs)
# RequiRED-Start: $all
# RequiRED-Stop: $all
# Default-Start: 2 3 4 5
# Default-Stop: 0 1 6
# Short-Description: ossrs(srs)
# Description: https://github.com/ossrs/srs
### END INIT INFO
# the config of ROOT, user must modify it when start srs from other directory,
# it's ok to use the script by command ./etc/init.d/ossrs
ROOT="/usr/local/srs"
APP="./objs/srs"
CONFIG="./conf/srs.conf"
DEFAULT_PID_FILE='./objs/srs.pid'
DEFAULT_LOG_FILE='./objs/srs.log'
########################################################################
# utility functions
########################################################################
RED="\\033[31m"
GREEN="\\033[32m"
YELLOW="\\033[33m"
BLACK="\\033[0m"
POS="\\033[60G"
ok_msg() {
echo -e "${1}${POS}${BLACK}[${GREEN} OK ${BLACK}]"
}
failed_msg() {
echo -e "${1}${POS}${BLACK}[${RED}FAILED${BLACK}]"
}
# load process info of srs
# @set variable $srs_pid to the process id in srs.pid file.
# @return 0, if process exists; otherwise:
# 1, for pid file not exists.
# 2, for get proecess info by pid failed.
# @set variable $error_msg if error.
# @set variable $pid_file to pid file.
load_process_info() {
# get pid file
pid_file=`cd ${ROOT} && cat ${CONFIG} |grep ^pid|awk '{print $2}'|awk -F ';' '{print $1}'`
if [[ -z $pid_file ]]; then pid_file=${DEFAULT_PID_FILE}; fi
# get abs path
pid_dir=`dirname $pid_file`
pid_file=`(cd ${ROOT}; cd $pid_dir; pwd)`/`basename $pid_file`
srs_pid=`cat $pid_file 2>/dev/null`
ret=$?; if [[ 0 -ne $ret ]]; then error_msg="file $pid_file does not exists"; return 1; fi
ps -p ${srs_pid} >/dev/null 2>/dev/null
ret=$?; if [[ 0 -ne $ret ]]; then error_msg="process $srs_pid does not exists"; return 2; fi
return 0;
}
start() {
# if exists, exit.
load_process_info
if [[ 0 -eq $? ]]; then failed_msg "SRS started(pid ${srs_pid}), should not start it again."; return 0; fi
# not exists, start server
ok_msg "Starting SRS..."
# get log file
log_file=`cd ${ROOT} && cat ${CONFIG} |grep '^log_file'| awk '{print $2}'| awk -F ';' '{print $1}'`
if [[ -z $log_file ]]; then log_file=${DEFAULT_LOG_FILE}; fi
# get abs path
log_dir=`dirname $log_file`
log_file=`(cd ${ROOT} && cd $log_dir && pwd)`/`basename $log_file`
# TODO: FIXME: set limit by, for instance, "ulimit -HSn 10000"
if [[ -z $log_file ]]; then
(ulimit -c unlimited && cd ${ROOT}; ${APP} -c ${CONFIG} >/dev/null 2>&1)
else
(ulimit -c unlimited && cd ${ROOT}; ${APP} -c ${CONFIG} >> $log_file 2>&1)
fi
# check again after start server
for ((i = 0; i < 5; i++)); do
# sleep a little while, for srs may start then crash.
sleep 0.1
load_process_info
ret=$?; if [[ 0 -ne $ret ]]; then
failed_msg "SRS start failed";
failed_msg "see $log_file";
return $ret;
fi
done
# check whether started.
load_process_info
ret=$?; if [[ 0 -eq $? ]]; then ok_msg "SRS started(pid ${srs_pid})"; return 0; fi
failed_msg "SRS not started"
return $ret
}
stop() {
# not start, exit
load_process_info
if [[ 0 -ne $? ]]; then failed_msg "SRS not start."; return 0; fi
ok_msg "Stopping SRS(pid ${srs_pid})..."
# process exists, try to kill to stop normally
for((i=0;i<100;i++)); do
load_process_info
if [[ 0 -eq $? ]]; then
kill -s SIGTERM ${srs_pid} 2>/dev/null
ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "send signal SIGTERM failed ret=$ret"; return $ret; fi
sleep 0.3
else
ok_msg "SRS stopped by SIGTERM"
# delete the pid file when stop success.
rm -f ${pid_file}
break;
fi
done
# process exists, use kill -9 to force to exit
load_process_info
if [[ 0 -eq $? ]]; then
kill -s SIGKILL ${srs_pid} 2>/dev/null
ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "send signal SIGKILL failed ret=$ret"; return $ret; fi
ok_msg "SRS stopped by SIGKILL"
else
# delete the pid file when stop success.
rm -f ${pid_file}
fi
sleep 0.1
return 0
}
# get the status of srs process
# @return 0 if srs is running; otherwise, 1 for stopped.
status() {
load_process_info
ret=$?; if [[ 0 -eq $ret ]]; then echo "SRS(pid ${srs_pid}) is running."; return 0; fi
echo "SRS is stopped, $error_msg"
return 1
}
reload() {
# not start, exit
load_process_info
if [[ 0 -ne $? ]]; then failed_msg "SRS not start."; return 0; fi
ok_msg "Reload SRS(pid ${srs_pid})..."
# process exists, reload it
kill -s SIGHUP ${srs_pid} 2>/dev/null
ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "Reload SRS failed ret=$ret"; return $ret; fi
load_process_info
if [[ 0 -ne $? ]]; then failed_msg "SRS reload failed."; return $ret; fi
ok_msg "SRS reloaded"
return 0
}
menu() {
case "$1" in
start)
start
;;
stop)
stop
;;
restart)
stop
start
;;
status)
status
;;
reload)
reload
;;
*)
echo "Usage: $0 {start|stop|status|restart|reload}"
return 1
;;
esac
}
menu $1
code=$?
exit ${code}
... ...
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<cross-domain-policy>
<allow-access-from domain="*"/>
</cross-domain-policy>
... ...
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