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胡斌
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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
  1 +# bandwidth test tool config for srs.
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +vhost __defaultVhost__ {
  9 +}
  10 +
  11 +vhost bandcheck.srs.com {
  12 + enabled on;
  13 + chunk_size 65000;
  14 + bandcheck {
  15 + enabled on;
  16 + key "35c9b402c12a7246868752e2878f7e0e";
  17 + interval 30;
  18 + limit_kbps 4000;
  19 + }
  20 +}
  1 +# no-daemon and write log to console config for srs.
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +http_api {
  9 + enabled on;
  10 + listen 1985;
  11 +}
  12 +http_server {
  13 + enabled on;
  14 + listen 8080;
  15 +}
  16 +vhost __defaultVhost__ {
  17 +}
  1 +# the config for srs demo
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 19350;
  6 +max_connections 1000;
  7 +daemon on;
  8 +srs_log_tank file;
  9 +srs_log_file ./objs/srs.demo.19350.log;
  10 +pid ./objs/srs.demo.19350.pid;
  11 +
  12 +vhost __defaultVhost__ {
  13 + enabled on;
  14 + gop_cache on;
  15 + hls {
  16 + enabled on;
  17 + hls_path ./objs/nginx/html/forward;
  18 + hls_fragment 5;
  19 + hls_window 30;
  20 + }
  21 +}
  22 +
  1 +# the config for srs demo
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleDemo
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon on;
  8 +srs_log_tank file;
  9 +srs_log_file ./objs/srs.demo.log;
  10 +pid ./objs/srs.demo.pid;
  11 +
  12 +chunk_size 60000;
  13 +max_connections 2000;
  14 +
  15 +http_api {
  16 + enabled on;
  17 + listen 1985;
  18 +}
  19 +
  20 +http_server {
  21 + enabled on;
  22 + listen 8080;
  23 + dir ./objs/nginx/html;
  24 +}
  25 +
  26 +vhost __defaultVhost__ {
  27 + enabled on;
  28 + gop_cache on;
  29 +}
  30 +
  31 +vhost bandcheck.srs.com {
  32 + enabled on;
  33 + chunk_size 65000;
  34 + bandcheck {
  35 + enabled on;
  36 + key "35c9b402c12a7246868752e2878f7e0e";
  37 + interval 30;
  38 + limit_kbps 4000;
  39 + }
  40 +}
  41 +
  42 +vhost demo.srs.com {
  43 + chunk_size 60000;
  44 + enabled on;
  45 + gop_cache on;
  46 + queue_length 30;
  47 + forward 127.0.0.1:19350;
  48 + bandcheck {
  49 + enabled off;
  50 + }
  51 + hls {
  52 + enabled on;
  53 + hls_path ./objs/nginx/html;
  54 + hls_fragment 5;
  55 + hls_window 30;
  56 + }
  57 + http_hooks {
  58 + enabled on;
  59 + on_connect http://127.0.0.1:8085/api/v1/clients;
  60 + on_close http://127.0.0.1:8085/api/v1/clients;
  61 + on_publish http://127.0.0.1:8085/api/v1/streams;
  62 + on_unpublish http://127.0.0.1:8085/api/v1/streams;
  63 + on_play http://127.0.0.1:8085/api/v1/sessions;
  64 + on_stop http://127.0.0.1:8085/api/v1/sessions;
  65 + }
  66 + transcode {
  67 + enabled on;
  68 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  69 + engine ld {
  70 + enabled on;
  71 + vfilter {
  72 + i ./doc/srs-logo.png;
  73 + filter_complex 'overlay=10:10';
  74 + }
  75 + vcodec libx264;
  76 + vbitrate 300;
  77 + vfps 20;
  78 + vwidth 768;
  79 + vheight 320;
  80 + vthreads 1;
  81 + vprofile baseline;
  82 + vpreset superfast;
  83 + vparams {
  84 + }
  85 + acodec libfdk_aac;
  86 + abitrate 45;
  87 + asample_rate 44100;
  88 + achannels 2;
  89 + aparams {
  90 + }
  91 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  92 + }
  93 + engine sd {
  94 + enabled on;
  95 + vfilter {
  96 + vf 'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
  97 + }
  98 + vcodec libx264;
  99 + vbitrate 500;
  100 + vfps 20;
  101 + vwidth 768;
  102 + vheight 320;
  103 + vthreads 1;
  104 + vprofile main;
  105 + vpreset fast;
  106 + vparams {
  107 + }
  108 + acodec libfdk_aac;
  109 + abitrate 40;
  110 + asample_rate 44100;
  111 + achannels 2;
  112 + aparams {
  113 + }
  114 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  115 + }
  116 + }
  117 + ingest {
  118 + enabled on;
  119 + input {
  120 + type file;
  121 + url ./doc/source.200kbps.768x320.flv;
  122 + }
  123 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  124 + engine {
  125 + enabled off;
  126 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
  127 + }
  128 + }
  129 +}
  130 +
  131 +vhost players {
  132 + enabled on;
  133 + gop_cache on;
  134 + transcode {
  135 + enabled on;
  136 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  137 + engine hls {
  138 + enabled on;
  139 + vfilter {
  140 + i ./doc/srs-logo.png;
  141 + filter_complex 'overlay=10:10';
  142 + }
  143 + vcodec libx264;
  144 + vbitrate 300;
  145 + vfps 20;
  146 + vwidth 768;
  147 + vheight 320;
  148 + vthreads 1;
  149 + vprofile baseline;
  150 + vpreset superfast;
  151 + vparams {
  152 + g 100;
  153 + }
  154 + acodec libfdk_aac;
  155 + abitrate 30;
  156 + asample_rate 44100;
  157 + achannels 2;
  158 + aparams {
  159 + }
  160 + output rtmp://127.0.0.1:[port]/[app]?vhost=players_pub/[stream];
  161 + }
  162 + }
  163 + ingest {
  164 + enabled on;
  165 + input {
  166 + type file;
  167 + url ./doc/source.200kbps.768x320.flv;
  168 + }
  169 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  170 + engine {
  171 + enabled off;
  172 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/demo;
  173 + }
  174 + }
  175 +}
  176 +vhost players_pub {
  177 + hls {
  178 + enabled on;
  179 + hls_path ./objs/nginx/html;
  180 + hls_fragment 5;
  181 + hls_window 30;
  182 + }
  183 +}
  184 +
  185 +vhost players_chat {
  186 + gop_cache off;
  187 + hls {
  188 + enabled off;
  189 + }
  190 + transcode {
  191 + enabled on;
  192 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  193 + engine chat_hls {
  194 + enabled on;
  195 + vfilter {
  196 + }
  197 + vcodec libx264;
  198 + vbitrate 150;
  199 + vfps 15;
  200 + vwidth 576;
  201 + vheight 240;
  202 + vthreads 1;
  203 + vprofile baseline;
  204 + vpreset ultrafast;
  205 + vparams {
  206 + g 30;
  207 + }
  208 + acodec an;
  209 + output rtmp://127.0.0.1:[port]/[app]?vhost=chat_hls/[stream];
  210 + }
  211 + }
  212 +}
  213 +vhost chat_hls {
  214 + hls {
  215 + enabled on;
  216 + hls_path ./objs/nginx/html;
  217 + hls_fragment 2;
  218 + hls_window 10;
  219 + }
  220 +}
  1 +# the config for srs to dvr in custom path.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
  3 +# @see https://github.com/ossrs/srs/wiki/v2_EN_DVR#custom-path
  4 +# @see full.conf for detail config.
  5 +
  6 +listen 1935;
  7 +max_connections 1000;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +vhost __defaultVhost__ {
  11 + dvr {
  12 + enabled on;
  13 + dvr_path ./objs/nginx/html/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
  14 + dvr_plan segment;
  15 + dvr_duration 30;
  16 + dvr_wait_keyframe on;
  17 + }
  18 +}
  1 +# the config for srs to dvr in segment mode
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + dvr {
  11 + enabled on;
  12 + dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
  13 + dvr_plan segment;
  14 + dvr_duration 30;
  15 + dvr_wait_keyframe on;
  16 + }
  17 +}
  1 +# the config for srs to dvr in session mode
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + dvr {
  11 + enabled on;
  12 + dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
  13 + dvr_plan session;
  14 + }
  15 +}
  1 +# the config for srs origin-edge cluster
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +pid objs/edge.pid;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +vhost __defaultVhost__ {
  11 + mode remote;
  12 + origin 127.0.0.1:19350;
  13 +}
  1 +# the config for srs for token traverse authentication
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_DRM
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + mode remote;
  11 + origin 127.0.0.1:19350;
  12 + token_traverse on;
  13 +}
  1 +# the config for srs use ffmpeg to transcode
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleFFMPEG
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + transcode {
  11 + enabled on;
  12 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  13 + engine ff {
  14 + enabled on;
  15 + vfilter {
  16 + }
  17 + vcodec libx264;
  18 + vthreads 4;
  19 + vprofile main;
  20 + vpreset medium;
  21 + vparams {
  22 + }
  23 + acodec libfdk_aac;
  24 + aparams {
  25 + }
  26 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  27 + }
  28 + }
  29 +}
  1 +# the config for srs to forward
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +pid ./objs/srs.master.pid;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +vhost __defaultVhost__ {
  11 + forward 127.0.0.1:19350;
  12 +#the forward destination server type,default true. if the server is not srs server,connect app maybe fail,try config as false
  13 + forward_server_srs true;
  14 +}
  1 +# the config for srs to forward
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleForward
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 19350;
  6 +max_connections 1000;
  7 +pid ./objs/srs.slave.pid;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +vhost __defaultVhost__ {
  11 +}
  1 +# all config for srs
  2 +
  3 +#############################################################################################
  4 +# RTMP sections
  5 +#############################################################################################
  6 +# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
  7 +# for example, 192.168.1.100:1935 10.10.10.100:1935
  8 +# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
  9 +listen 1935;
  10 +# the pid file
  11 +# to ensure only one process can use a pid file
  12 +# and provides the current running process id, for script,
  13 +# for example, init.d script to manage the server.
  14 +# default: ./objs/srs.pid
  15 +pid ./objs/srs.pid;
  16 +# the default chunk size is 128, max is 65536,
  17 +# some client does not support chunk size change,
  18 +# however, most clients supports it and it can improve
  19 +# performance about 10%.
  20 +# default: 60000
  21 +chunk_size 60000;
  22 +# the logs dir.
  23 +# if enabled ffmpeg, each stracoding stream will create a log file.
  24 +# /dev/null to disable the log.
  25 +# default: ./objs
  26 +ff_log_dir ./objs;
  27 +# the log tank, console or file.
  28 +# if console, print log to console.
  29 +# if file, write log to file. requires srs_log_file if log to file.
  30 +# default: file.
  31 +srs_log_tank file;
  32 +# the log level, for all log tanks.
  33 +# can be: verbose, info, trace, warn, error
  34 +# default: trace
  35 +srs_log_level trace;
  36 +# when srs_log_tank is file, specifies the log file.
  37 +# default: ./objs/srs.log
  38 +srs_log_file ./objs/srs.log;
  39 +# the max connections.
  40 +# if exceed the max connections, server will drop the new connection.
  41 +# default: 1000
  42 +max_connections 1000;
  43 +# whether start as daemon
  44 +# @remark: donot support reload.
  45 +# default: on
  46 +daemon on;
  47 +# whether use utc_time to generate the time struct,
  48 +# if off, use localtime() to generate it,
  49 +# if on, use gmtime() instead, which use UTC time.
  50 +# default: off
  51 +utc_time off;
  52 +
  53 +# the work dir for server, to chdir(work_dir) when not empty or "./"
  54 +# user can config this directory to change the dir.
  55 +# @reamrk do not support reload.
  56 +# default: ./
  57 +work_dir ./;
  58 +# whether quit when parent process changed,
  59 +# used for supervisor mode(not daemon), srs should always quit when
  60 +# supervisor process exited.
  61 +# @remark conflict with daemon, error when both daemon and asprocess are on.
  62 +# @reamrk do not support reload.
  63 +# default: off
  64 +asprocess off;
  65 +
  66 +#############################################################################################
  67 +# heartbeat/stats sections
  68 +#############################################################################################
  69 +# heartbeat to api server
  70 +# @remark, the ip report to server, is retrieve from system stat,
  71 +# which need the config item stats.network.
  72 +heartbeat {
  73 + # whether heartbeat is enalbed.
  74 + # default: off
  75 + enabled off;
  76 + # the interval seconds for heartbeat,
  77 + # recommend 0.3,0.6,0.9,1.2,1.5,1.8,2.1,2.4,2.7,3,...,6,9,12,....
  78 + # default: 9.9
  79 + interval 9.3;
  80 + # when startup, srs will heartbeat to this api.
  81 + # @remark: must be a restful http api url, where SRS will POST with following data:
  82 + # {
  83 + # "device_id": "my-srs-device",
  84 + # "ip": "192.168.1.100"
  85 + # }
  86 + # default: http://127.0.0.1:8085/api/v1/servers
  87 + url http://127.0.0.1:8085/api/v1/servers;
  88 + # the id of devide.
  89 + device_id "my-srs-device";
  90 + # whether report with summaries
  91 + # if on, put /api/v1/summaries to the request data:
  92 + # {
  93 + # "summaries": summaries object.
  94 + # }
  95 + # @remark: optional config.
  96 + # default: off
  97 + summaries off;
  98 +}
  99 +
  100 +# system statistics section.
  101 +# the main cycle will retrieve the system stat,
  102 +# for example, the cpu/mem/network/disk-io data,
  103 +# the http api, for instance, /api/v1/summaries will show these data.
  104 +# @remark the heartbeat depends on the network,
  105 +# for example, the eth0 maybe the device which index is 0.
  106 +stats {
  107 + # the index of device ip.
  108 + # we may retrieve more than one network device.
  109 + # default: 0
  110 + network 0;
  111 + # the device name to stat the disk iops.
  112 + # ignore the device of /proc/diskstats if not configed.
  113 + disk sda sdb xvda xvdb;
  114 +}
  115 +
  116 +#############################################################################################
  117 +# HTTP sections
  118 +#############################################################################################
  119 +# api of srs.
  120 +# the http api config, export for external program to manage srs.
  121 +# user can access http api of srs in browser directly, for instance, to access by:
  122 +# curl http://192.168.1.170:1985/api/v1/reload
  123 +# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
  124 +# where the cli can only be used in shell/terminate.
  125 +http_api {
  126 + # whether http api is enabled.
  127 + # default: off
  128 + enabled on;
  129 + # the http api listen entry is <[ip:]port>
  130 + # for example, 192.168.1.100:1985
  131 + # where the ip is optional, default to 0.0.0.0, that is 1985 equals to 0.0.0.0:1985
  132 + # default: 1985
  133 + listen 1985;
  134 + # whether enable crossdomain request.
  135 + # default: on
  136 + crossdomain on;
  137 +}
  138 +# embeded http server in srs.
  139 +# the http streaming config, for HLS/HDS/DASH/HTTPProgressive
  140 +# global config for http streaming, user must config the http section for each vhost.
  141 +# the embed http server used to substitute nginx in ./objs/nginx,
  142 +# for example, srs runing in arm, can provides RTMP and HTTP service, only with srs installed.
  143 +# user can access the http server pages, generally:
  144 +# curl http://192.168.1.170:80/srs.html
  145 +# which will show srs version and welcome to srs.
  146 +# @remark, the http embeded stream need to config the vhost, for instance, the __defaultVhost__
  147 +# need to open the feature http of vhost.
  148 +http_server {
  149 + # whether http streaming service is enabled.
  150 + # default: off
  151 + enabled on;
  152 + # the http streaming listen entry is <[ip:]port>
  153 + # for example, 192.168.1.100:8080
  154 + # where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
  155 + # @remark, if use lower port, for instance 80, user must start srs by root.
  156 + # default: 8080
  157 + listen 8080;
  158 + # the default dir for http root.
  159 + # default: ./objs/nginx/html
  160 + dir ./objs/nginx/html;
  161 +}
  162 +
  163 +#############################################################################################
  164 +# Streamer sections
  165 +#############################################################################################
  166 +# the streamer cast stream from other protocol to SRS over RTMP.
  167 +# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
  168 +stream_caster {
  169 + # whether stream caster is enabled.
  170 + # default: off
  171 + enabled off;
  172 + # the caster type of stream, the casters:
  173 + # mpegts_over_udp, MPEG-TS over UDP caster.
  174 + # rtsp, Real Time Streaming Protocol (RTSP).
  175 + # flv, FLV over HTTP POST.
  176 + caster mpegts_over_udp;
  177 + # the output rtmp url.
  178 + # for mpegts_over_udp caster, the typically output url:
  179 + # rtmp://127.0.0.1/live/livestream
  180 + # for rtsp caster, the typically output url:
  181 + # rtmp://127.0.0.1/[app]/[stream]
  182 + # for example, the rtsp url:
  183 + # rtsp://192.168.1.173:8544/live/livestream.sdp
  184 + # where the [app] is "live" and [stream] is "livestream", output is:
  185 + # rtmp://127.0.0.1/live/livestream
  186 + output rtmp://127.0.0.1/live/livestream;
  187 + # the listen port for stream caster.
  188 + # for mpegts_over_udp caster, listen at udp port. for example, 8935.
  189 + # for rtsp caster, listen at tcp port. for example, 554.
  190 + # for flv caster, listen at tcp port. for example, 8936.
  191 + # TODO: support listen at <[ip:]port>
  192 + listen 8935;
  193 + # for the rtsp caster, the rtp server local port over udp,
  194 + # which reply the rtsp setup request message, the port will be used:
  195 + # [rtp_port_min, rtp_port_max)
  196 + rtp_port_min 57200;
  197 + rtp_port_max 57300;
  198 +}
  199 +stream_caster {
  200 + enabled off;
  201 + caster mpegts_over_udp;
  202 + output rtmp://127.0.0.1/live/livestream;
  203 + listen 8935;
  204 +}
  205 +stream_caster {
  206 + enabled off;
  207 + caster rtsp;
  208 + output rtmp://127.0.0.1/[app]/[stream];
  209 + listen 554;
  210 + rtp_port_min 57200;
  211 + rtp_port_max 57300;
  212 +}
  213 +stream_caster {
  214 + enabled off;
  215 + caster flv;
  216 + output rtmp://127.0.0.1/[app]/[stream];
  217 + listen 8936;
  218 +}
  219 +
  220 +#############################################################################################
  221 +# RTMP/HTTP VHOST sections
  222 +#############################################################################################
  223 +# vhost list, the __defaultVhost__ is the default vhost
  224 +# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
  225 +# for which cannot identify the required vhost.
  226 +vhost __defaultVhost__ {
  227 +}
  228 +
  229 +# the security to allow or deny clients.
  230 +vhost security.srs.com {
  231 + # security for host to allow or deny clients.
  232 + # @see https://github.com/ossrs/srs/issues/211
  233 + security {
  234 + # whether enable the security for vhost.
  235 + # default: off
  236 + enabled on;
  237 + # the security list, each item format as:
  238 + # allow|deny publish|play all|<ip>
  239 + # for example:
  240 + # allow publish all;
  241 + # deny publish all;
  242 + # allow publish 127.0.0.1;
  243 + # deny publish 127.0.0.1;
  244 + # allow play all;
  245 + # deny play all;
  246 + # allow play 127.0.0.1;
  247 + # deny play 127.0.0.1;
  248 + # SRS apply the following simple strategies one by one:
  249 + # 1. allow all if security disabled.
  250 + # 2. default to deny all when security enabled.
  251 + # 3. allow if matches allow strategy.
  252 + # 4. deny if matches deny strategy.
  253 + allow play all;
  254 + allow publish all;
  255 + }
  256 +}
  257 +
  258 +# the MR(merged-read) setting for publisher.
  259 +# the MW(merged-write) settings for player.
  260 +vhost mrw.srs.com {
  261 + # whether enable min delay mode for vhost.
  262 + # for min latence mode:
  263 + # 1. disable the mr for vhost.
  264 + # 2. use timeout for cond wait for consumer queue.
  265 + # @see https://github.com/ossrs/srs/issues/257
  266 + # default: off
  267 + min_latency off;
  268 + # about MR, read https://github.com/ossrs/srs/issues/241
  269 + mr {
  270 + # whether enable the MR(merged-read)
  271 + # default: off
  272 + enabled on;
  273 + # the latency in ms for MR(merged-read),
  274 + # the performance+ when latency+, and memory+,
  275 + # memory(buffer) = latency * kbps / 8
  276 + # for example, latency=500ms, kbps=3000kbps, each publish connection will consume
  277 + # memory = 500 * 3000 / 8 = 187500B = 183KB
  278 + # when there are 2500 publisher, the total memory of SRS atleast:
  279 + # 183KB * 2500 = 446MB
  280 + # the value recomment is [300, 2000]
  281 + # default: 350
  282 + latency 350;
  283 + }
  284 + # set the MW(merged-write) latency in ms.
  285 + # SRS always set mw on, so we just set the latency value.
  286 + # the latency of stream >= mw_latency + mr_latency
  287 + # the value recomment is [300, 1800]
  288 + # default: 350
  289 + mw_latency 350;
  290 +}
  291 +
  292 +# vhost for edge, edge and origin is the same vhost
  293 +vhost same.edge.srs.com {
  294 + # the mode of vhost, local or remote.
  295 + # local: vhost is origin vhost, which provides stream source.
  296 + # remote: vhost is edge vhost, which pull/push to origin.
  297 + # default: local
  298 + mode remote;
  299 + # for edge(remote mode), user must specifies the origin server
  300 + # format as: <server_name|ip>[:port]
  301 + # @remark user can specifies multiple origin for error backup, by space,
  302 + # for example, 192.168.1.100:1935 192.168.1.101:1935 192.168.1.102:1935
  303 + origin 127.0.0.1:1935 localhost:1935;
  304 + # for edge, whether open the token traverse mode,
  305 + # if token traverse on, all connections of edge will forward to origin to check(auth),
  306 + # it's very important for the edge to do the token auth.
  307 + # the better way is use http callback to do the token auth by the edge,
  308 + # but if user prefer origin check(auth), the token_traverse if better solution.
  309 + # default: off
  310 + token_traverse off;
  311 +}
  312 +
  313 +# vhost for edge, edge transform vhost to fetch from another vhost.
  314 +vhost transform.edge.srs.com {
  315 + mode remote;
  316 + origin 127.0.0.1:1935;
  317 + # the vhost to transform for edge,
  318 + # to fetch from the specified vhost at origin,
  319 + # if not specified, use the current vhost of edge in origin, the variable [vhost].
  320 + # default: [vhost]
  321 + vhost same.edge.srs.com;
  322 +}
  323 +
  324 +# vhost for dvr
  325 +vhost dvr.srs.com {
  326 + # dvr RTMP stream to file,
  327 + # start to record to file when encoder publish,
  328 + # reap flv according by specified dvr_plan.
  329 + dvr {
  330 + # whether enabled dvr features
  331 + # default: off
  332 + enabled on;
  333 + # the dvr plan. canbe:
  334 + # session reap flv when session end(unpublish).
  335 + # segment reap flv when flv duration exceed the specified dvr_duration.
  336 + # append always append to flv file, never reap it.
  337 + # default: session
  338 + dvr_plan session;
  339 + # the dvr output path.
  340 + # we supports some variables to generate the filename.
  341 + # [vhost], the vhost of stream.
  342 + # [app], the app of stream.
  343 + # [stream], the stream name of stream.
  344 + # [2006], replace this const to current year.
  345 + # [01], replace this const to current month.
  346 + # [02], replace this const to current date.
  347 + # [15], replace this const to current hour.
  348 + # [04], repleace this const to current minute.
  349 + # [05], repleace this const to current second.
  350 + # [999], repleace this const to current millisecond.
  351 + # [timestamp],replace this const to current UNIX timestamp in ms.
  352 + # @remark we use golang time format "2006-01-02 15:04:05.999" as "[2006]-[01]-[02]_[15].[04].[05]_[999]"
  353 + # for example, for url rtmp://ossrs.net/live/livestream and time 2015-01-03 10:57:30.776
  354 + # 1. No variables, the rule of SRS1.0(auto add [stream].[timestamp].flv as filename):
  355 + # dvr_path ./objs/nginx/html;
  356 + # =>
  357 + # dvr_path ./objs/nginx/html/live/livestream.1420254068776.flv;
  358 + # 2. Use stream and date as dir name, time as filename:
  359 + # dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
  360 + # =>
  361 + # dvr_path /data/ossrs.net/live/livestream/2015/01/03/10.57.30.776.flv;
  362 + # 3. Use stream and year/month as dir name, date and time as filename:
  363 + # dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]-[15].[04].[05].[999].flv;
  364 + # =>
  365 + # dvr_path /data/ossrs.net/live/livestream/2015/01/03-10.57.30.776.flv;
  366 + # 4. Use vhost/app and year/month as dir name, stream/date/time as filename:
  367 + # dvr_path /data/[vhost]/[app]/[2006]/[01]/[stream]-[02]-[15].[04].[05].[999].flv;
  368 + # =>
  369 + # dvr_path /data/ossrs.net/live/2015/01/livestream-03-10.57.30.776.flv;
  370 + # @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
  371 + # @see https://github.com/ossrs/srs/wiki/v2_EN_DVR#custom-path
  372 + # segment,session apply it.
  373 + # default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
  374 + dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
  375 + # the duration for dvr file, reap if exeed, in seconds.
  376 + # segment apply it.
  377 + # session,append ignore.
  378 + # default: 30
  379 + dvr_duration 30;
  380 + # whether wait keyframe to reap segment,
  381 + # if off, reap segment when duration exceed the dvr_duration,
  382 + # if on, reap segment when duration exceed and got keyframe.
  383 + # segment apply it.
  384 + # session,append ignore.
  385 + # default: on
  386 + dvr_wait_keyframe on;
  387 + # about the stream monotonically increasing:
  388 + # 1. video timestamp is monotonically increasing,
  389 + # 2. audio timestamp is monotonically increasing,
  390 + # 3. video and audio timestamp is interleaved monotonically increasing.
  391 + # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
  392 + # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
  393 + # the time jitter algorithm:
  394 + # 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
  395 + # 2. zero, only ensure sttream start at zero, ignore timestamp jitter.
  396 + # 3. off, disable the time jitter algorithm, like atc.
  397 + # apply for all dvr plan.
  398 + # default: full
  399 + time_jitter full;
  400 +
  401 + # on_dvr, never config in here, should config in http_hooks.
  402 + # for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
  403 + # @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
  404 + # @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
  405 + }
  406 +}
  407 +
  408 +# vhost for ingest
  409 +vhost ingest.srs.com {
  410 + # ingest file/stream/device then push to SRS over RTMP.
  411 + # the name/id used to identify the ingest, must be unique in global.
  412 + # ingest id is used in reload or http api management.
  413 + ingest livestream {
  414 + # whether enabled ingest features
  415 + # default: off
  416 + enabled on;
  417 + # input file/stream/device
  418 + # @remark only support one input.
  419 + input {
  420 + # the type of input.
  421 + # can be file/stream/device, that is,
  422 + # file: ingest file specifies by url.
  423 + # stream: ingest stream specifeis by url.
  424 + # device: not support yet.
  425 + # default: file
  426 + type file;
  427 + # the url of file/stream.
  428 + url ./doc/source.200kbps.768x320.flv;
  429 + }
  430 + # the ffmpeg
  431 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  432 + # the transcode engine, @see all.transcode.srs.com
  433 + # @remark, the output is specified following.
  434 + engine {
  435 + # @see enabled of transcode engine.
  436 + # if disabled or vcodec/acodec not specified, use copy.
  437 + # default: off.
  438 + enabled off;
  439 + # output stream. variables:
  440 + # [vhost] current vhost which start the ingest.
  441 + # [port] system RTMP stream port.
  442 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
  443 + }
  444 + }
  445 +}
  446 +
  447 +# vhost for http static and flv vod stream for each vhost.
  448 +vhost http.static.srs.com {
  449 + # http static vhost specified config
  450 + http_static {
  451 + # whether enabled the http static service for vhost.
  452 + # default: off
  453 + enabled on;
  454 + # the url to mount to,
  455 + # typical mount to [vhost]/
  456 + # the variables:
  457 + # [vhost] current vhost for http server.
  458 + # @remark the [vhost] is optional, used to mount at specified vhost.
  459 + # @remark the http of __defaultVhost__ will override the http_server section.
  460 + # for example:
  461 + # mount to [vhost]/
  462 + # access by http://ossrs.net:8080/xxx.html
  463 + # mount to [vhost]/hls
  464 + # access by http://ossrs.net:8080/hls/xxx.html
  465 + # mount to /
  466 + # access by http://ossrs.net:8080/xxx.html
  467 + # or by http://192.168.1.173:8080/xxx.html
  468 + # mount to /hls
  469 + # access by http://ossrs.net:8080/hls/xxx.html
  470 + # or by http://192.168.1.173:8080/hls/xxx.html
  471 + # @remark the port of http is specified by http_server section.
  472 + # default: [vhost]/
  473 + mount [vhost]/hls;
  474 + # main dir of vhost,
  475 + # to delivery HTTP stream of this vhost.
  476 + # default: ./objs/nginx/html
  477 + dir ./objs/nginx/html/hls;
  478 + }
  479 +}
  480 +
  481 +# vhost for http flv/aac/mp3 live stream for each vhost.
  482 +vhost http.remux.srs.com {
  483 + # http flv/mp3/aac/ts stream vhost specified config
  484 + http_remux {
  485 + # whether enable the http live streaming service for vhost.
  486 + # default: off
  487 + enabled on;
  488 + # the fast cache for audio stream(mp3/aac),
  489 + # to cache more audio and send to client in a time to make android(weixin) happy.
  490 + # @remark the flv/ts stream ignore it
  491 + # @remark 0 to disable fast cache for http audio stream.
  492 + # default: 0
  493 + fast_cache 30;
  494 + # the stream mout for rtmp to remux to live streaming.
  495 + # typical mount to [vhost]/[app]/[stream].flv
  496 + # the variables:
  497 + # [vhost] current vhost for http live stream.
  498 + # [app] current app for http live stream.
  499 + # [stream] current stream for http live stream.
  500 + # @remark the [vhost] is optional, used to mount at specified vhost.
  501 + # the extension:
  502 + # .flv mount http live flv stream, use default gop cache.
  503 + # .ts mount http live ts stream, use default gop cache.
  504 + # .mp3 mount http live mp3 stream, ignore video and audio mp3 codec required.
  505 + # .aac mount http live aac stream, ignore video and audio aac codec required.
  506 + # for example:
  507 + # mount to [vhost]/[app]/[stream].flv
  508 + # access by http://ossrs.net:8080/live/livestream.flv
  509 + # mount to /[app]/[stream].flv
  510 + # access by http://ossrs.net:8080/live/livestream.flv
  511 + # or by http://192.168.1.173:8080/live/livestream.flv
  512 + # mount to [vhost]/[app]/[stream].mp3
  513 + # access by http://ossrs.net:8080/live/livestream.mp3
  514 + # mount to [vhost]/[app]/[stream].aac
  515 + # access by http://ossrs.net:8080/live/livestream.aac
  516 + # mount to [vhost]/[app]/[stream].ts
  517 + # access by http://ossrs.net:8080/live/livestream.ts
  518 + # @remark the port of http is specified by http_server section.
  519 + # default: [vhost]/[app]/[stream].flv
  520 + mount [vhost]/[app]/[stream].flv;
  521 + # whether http stream trigger rtmp stream source when no stream available,
  522 + # for example, when encoder has not publish stream yet,
  523 + # user can play the http flv stream and wait for stream.
  524 + # default: on
  525 + hstrs on;
  526 + }
  527 +}
  528 +
  529 +# the vhost with hls specified.
  530 +vhost with-hls.srs.com {
  531 + hls {
  532 + # whether the hls is enabled.
  533 + # if off, donot write hls(ts and m3u8) when publish.
  534 + # default: off
  535 + enabled on;
  536 + # the hls fragment in seconds, the duration of a piece of ts.
  537 + # default: 10
  538 + hls_fragment 10;
  539 + # the hls m3u8 target duration ratio,
  540 + # EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
  541 + # EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
  542 + # @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
  543 + # default: 1.5
  544 + hls_td_ratio 1.5;
  545 + # the audio overflow ratio.
  546 + # for pure audio, the duration to reap the segment.
  547 + # for example, the hls_fragment is 10s, hsl_aof_ratio is 2.0,
  548 + # the segemnt will reap to 20s for pure audio.
  549 + # default: 2.0
  550 + hls_aof_ratio 2.0;
  551 + # the hls window in seconds, the number of ts in m3u8.
  552 + # default: 60
  553 + hls_window 60;
  554 + # the error strategy. canbe:
  555 + # ignore, disable the hls.
  556 + # disconnect, require encoder republish.
  557 + # continue, ignore failed try to continue output hls.
  558 + # @see https://github.com/ossrs/srs/issues/264
  559 + # default: continue
  560 + hls_on_error continue;
  561 + # the hls output path.
  562 + # the m3u8 file is configed by hls_path/hls_m3u8_file, the default is:
  563 + # ./objs/nginx/html/[app]/[stream].m3u8
  564 + # the ts file is configed by hls_path/hls_ts_file, the default is:
  565 + # ./objs/nginx/html/[app]/[stream]-[seq].ts
  566 + # @remark the hls_path is compatible with srs v1 config.
  567 + # default: ./objs/nginx/html
  568 + hls_path ./objs/nginx/html;
  569 + # the hls m3u8 file name.
  570 + # we supports some variables to generate the filename.
  571 + # [vhost], the vhost of stream.
  572 + # [app], the app of stream.
  573 + # [stream], the stream name of stream.
  574 + # default: [app]/[stream].m3u8
  575 + hls_m3u8_file [app]/[stream].m3u8;
  576 + # the hls ts file name.
  577 + # we supports some variables to generate the filename.
  578 + # [vhost], the vhost of stream.
  579 + # [app], the app of stream.
  580 + # [stream], the stream name of stream.
  581 + # [2006], replace this const to current year.
  582 + # [01], replace this const to current month.
  583 + # [02], replace this const to current date.
  584 + # [15], replace this const to current hour.
  585 + # [04], repleace this const to current minute.
  586 + # [05], repleace this const to current second.
  587 + # [999], repleace this const to current millisecond.
  588 + # [timestamp],replace this const to current UNIX timestamp in ms.
  589 + # [seq], the sequence number of ts.
  590 + # @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
  591 + # @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
  592 + # default: [app]/[stream]-[seq].ts
  593 + hls_ts_file [app]/[stream]-[seq].ts;
  594 + # whether use floor for the hls_ts_file path generation.
  595 + # if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
  596 + # and use enahanced algorithm to calc deviation for segment.
  597 + # @remark when floor on, recommend the hls_segment>=2*gop.
  598 + # default: off
  599 + hls_ts_floor off;
  600 + # the hls entry prefix, which is base url of ts url.
  601 + # if specified, the ts path in m3u8 will be like:
  602 + # http://your-server/live/livestream-0.ts
  603 + # http://your-server/live/livestream-1.ts
  604 + # ...
  605 + # optional, default to empty string.
  606 + hls_entry_prefix http://your-server;
  607 + # the default audio codec of hls.
  608 + # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
  609 + # so user can set the default codec for mp3.
  610 + # the available audio codec:
  611 + # aac, mp3, an
  612 + # default: aac
  613 + hls_acodec aac;
  614 + # the default video codec of hls.
  615 + # when codec changed, write the PAT/PMT table, but maybe ok util next ts.
  616 + # so user can set the default codec for pure audio(without video) to vn.
  617 + # the available video codec:
  618 + # h264, vn
  619 + # default: h264
  620 + hls_vcodec h264;
  621 + # whether cleanup the old expired ts files.
  622 + # default: on
  623 + hls_cleanup on;
  624 + # the timeout in seconds to dispose the hls,
  625 + # dispose is to remove all hls files, m3u8 and ts files.
  626 + # when publisher timeout dispose hls.
  627 + # @remark 0 to disable dispose for publisher.
  628 + # @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose hls.
  629 + # default: 0
  630 + hls_dispose 0;
  631 + # the max size to notify hls,
  632 + # to read max bytes from ts of specified cdn network,
  633 + # @remark only used when on_hls_notify is config.
  634 + # default: 64
  635 + hls_nb_notify 64;
  636 + # whether wait keyframe to reap segment,
  637 + # if off, reap segment when duration exceed the fragment,
  638 + # if on, reap segment when duration exceed and got keyframe.
  639 + # default: on
  640 + hls_wait_keyframe on;
  641 +
  642 + # on_hls, never config in here, should config in http_hooks.
  643 + # for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
  644 + # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
  645 + # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
  646 +
  647 + # on_hls_notify, never config in here, should config in http_hooks.
  648 + # we support the variables to generate the notify url:
  649 + # [app], replace with the app.
  650 + # [stream], replace with the stream.
  651 + # [ts_url], replace with the ts url.
  652 + # for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
  653 + # @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
  654 + # @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
  655 + }
  656 +}
  657 +# the vhost with hls disabled.
  658 +vhost no-hls.srs.com {
  659 + hls {
  660 + # whether the hls is enabled.
  661 + # if off, donot write hls(ts and m3u8) when publish.
  662 + # default: off
  663 + enabled off;
  664 + }
  665 +}
  666 +
  667 +# the vhost with adobe hds
  668 +vhost hds.srs.com {
  669 + hds {
  670 + # whether hds enabled
  671 + # default: off
  672 + enabled on;
  673 + # the hds fragment in seconds.
  674 + # default: 10
  675 + hds_fragment 10;
  676 + # the hds window in seconds, erase the segment when exceed the window.
  677 + # default: 60
  678 + hds_window 60;
  679 + # the path to store the hds files.
  680 + # default: ./objs/nginx/html
  681 + hds_path ./objs/nginx/html;
  682 + }
  683 +}
  684 +
  685 +# the http hook callback vhost, srs will invoke the hooks for specified events.
  686 +vhost hooks.callback.srs.com {
  687 + http_hooks {
  688 + # whether the http hooks enalbe.
  689 + # default off.
  690 + enabled on;
  691 + # when client connect to vhost/app, call the hook,
  692 + # the request in the POST data string is a object encode by json:
  693 + # {
  694 + # "action": "on_connect",
  695 + # "client_id": 1985,
  696 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  697 + # "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
  698 + # "pageUrl": "http://www.test.com/live.html"
  699 + # }
  700 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  701 + # an int value specifies the error code(0 corresponding to success):
  702 + # 0
  703 + # support multiple api hooks, format:
  704 + # on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
  705 + on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
  706 + # when client close/disconnect to vhost/app/stream, call the hook,
  707 + # the request in the POST data string is a object encode by json:
  708 + # {
  709 + # "action": "on_close",
  710 + # "client_id": 1985,
  711 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  712 + # "send_bytes": 10240, "recv_bytes": 10240
  713 + # }
  714 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  715 + # an int value specifies the error code(0 corresponding to success):
  716 + # 0
  717 + # support multiple api hooks, format:
  718 + # on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
  719 + on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
  720 + # when client(encoder) publish to vhost/app/stream, call the hook,
  721 + # the request in the POST data string is a object encode by json:
  722 + # {
  723 + # "action": "on_publish",
  724 + # "client_id": 1985,
  725 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  726 + # "stream": "livestream"
  727 + # }
  728 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  729 + # an int value specifies the error code(0 corresponding to success):
  730 + # 0
  731 + # support multiple api hooks, format:
  732 + # on_publish http://xxx/api0 http://xxx/api1 http://xxx/apiN
  733 + on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
  734 + # when client(encoder) stop publish to vhost/app/stream, call the hook,
  735 + # the request in the POST data string is a object encode by json:
  736 + # {
  737 + # "action": "on_unpublish",
  738 + # "client_id": 1985,
  739 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  740 + # "stream": "livestream"
  741 + # }
  742 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  743 + # an int value specifies the error code(0 corresponding to success):
  744 + # 0
  745 + # support multiple api hooks, format:
  746 + # on_unpublish http://xxx/api0 http://xxx/api1 http://xxx/apiN
  747 + on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
  748 + # when client start to play vhost/app/stream, call the hook,
  749 + # the request in the POST data string is a object encode by json:
  750 + # {
  751 + # "action": "on_play",
  752 + # "client_id": 1985,
  753 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  754 + # "stream": "livestream",
  755 + # "pageUrl": "http://www.test.com/live.html"
  756 + # }
  757 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  758 + # an int value specifies the error code(0 corresponding to success):
  759 + # 0
  760 + # support multiple api hooks, format:
  761 + # on_play http://xxx/api0 http://xxx/api1 http://xxx/apiN
  762 + on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
  763 + # when client stop to play vhost/app/stream, call the hook,
  764 + # the request in the POST data string is a object encode by json:
  765 + # {
  766 + # "action": "on_stop",
  767 + # "client_id": 1985,
  768 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  769 + # "stream": "livestream"
  770 + # }
  771 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  772 + # an int value specifies the error code(0 corresponding to success):
  773 + # 0
  774 + # support multiple api hooks, format:
  775 + # on_stop http://xxx/api0 http://xxx/api1 http://xxx/apiN
  776 + on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
  777 + # when srs reap a dvr file, call the hook,
  778 + # the request in the POST data string is a object encode by json:
  779 + # {
  780 + # "action": "on_dvr",
  781 + # "client_id": 1985,
  782 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  783 + # "stream": "livestream",
  784 + # "cwd": "/usr/local/srs",
  785 + # "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
  786 + # }
  787 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  788 + # an int value specifies the error code(0 corresponding to success):
  789 + # 0
  790 + on_dvr http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
  791 + # when srs reap a ts file of hls, call the hook,
  792 + # the request in the POST data string is a object encode by json:
  793 + # {
  794 + # "action": "on_hls",
  795 + # "client_id": 1985,
  796 + # "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
  797 + # "stream": "livestream",
  798 + # "duration": 9.36, // in seconds
  799 + # "cwd": "/usr/local/srs",
  800 + # "file": "./objs/nginx/html/live/livestream/2015-04-23/01/476584165.ts",
  801 + # "url": "live/livestream/2015-04-23/01/476584165.ts",
  802 + # "m3u8": "./objs/nginx/html/live/livestream/live.m3u8",
  803 + # "m3u8_url": "live/livestream/live.m3u8",
  804 + # "seq_no": 100
  805 + # }
  806 + # if valid, the hook must return HTTP code 200(Stauts OK) and response
  807 + # an int value specifies the error code(0 corresponding to success):
  808 + # 0
  809 + on_hls http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
  810 + # when srs reap a ts file of hls, call this hook,
  811 + # used to push file to cdn network, by get the ts file from cdn network.
  812 + # so we use HTTP GET and use the variable following:
  813 + # [app], replace with the app.
  814 + # [stream], replace with the stream.
  815 + # [ts_url], replace with the ts url.
  816 + # ignore any return data of server.
  817 + # @remark random select a url to report, not report all.
  818 + on_hls_notify http://127.0.0.1:8085/api/v1/hls/[app]/[stream][ts_url];
  819 + }
  820 +}
  821 +
  822 +# the vhost for srs debug info, whether send args in connect(tcUrl).
  823 +vhost debug.srs.com {
  824 + # when upnode(forward to, edge push to, edge pull from) is srs,
  825 + # it's strongly recommend to open the debug_srs_upnode,
  826 + # when connect to upnode, it will take the debug info,
  827 + # for example, the id, source id, pid.
  828 + # please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
  829 + # default: on
  830 + debug_srs_upnode on;
  831 +}
  832 +
  833 +# the vhost for min delay, donot cache any stream.
  834 +vhost min.delay.com {
  835 + # @see vhost mrw.srs.com for detail.
  836 + min_latency on;
  837 + mr {
  838 + enabled off;
  839 + }
  840 + mw_latency 100;
  841 + # whether cache the last gop.
  842 + # if on, cache the last gop and dispatch to client,
  843 + # to enabled fast startup for client, client play immediately.
  844 + # if off, send the latest media data to client,
  845 + # client need to wait for the next Iframe to decode and show the video.
  846 + # set to off if requires min delay;
  847 + # set to on if requires client fast startup.
  848 + # default: on
  849 + gop_cache off;
  850 + # the max live queue length in seconds.
  851 + # if the messages in the queue exceed the max length,
  852 + # drop the old whole gop.
  853 + # default: 30
  854 + queue_length 10;
  855 + # whether enable the TCP_NODELAY
  856 + # if on, set the nodelay of fd by setsockopt
  857 + # default: off
  858 + tcp_nodelay on;
  859 +}
  860 +
  861 +# whether disable the sps parse, for the resolution of video.
  862 +vhost no.parse.sps.com {
  863 + publish {
  864 + # whether parse the sps when publish stream.
  865 + # we can got the resolution of video for stat api.
  866 + # but we may failed to cause publish failed.
  867 + # default: on
  868 + parse_sps on;
  869 + }
  870 +}
  871 +
  872 +# the vhost to control the stream delivery feature
  873 +vhost stream.control.com {
  874 + # @see vhost mrw.srs.com for detail.
  875 + min_latency on;
  876 + mr {
  877 + enabled off;
  878 + }
  879 + mw_latency 100;
  880 + # @see vhost min.delay.com
  881 + queue_length 10;
  882 + tcp_nodelay on;
  883 + # the minimal packets send interval in ms,
  884 + # used to control the ndiff of stream by srs_rtmp_dump,
  885 + # for example, some device can only accept some stream which
  886 + # delivery packets in constant interval(not cbr).
  887 + # @remark 0 to disable the minimal interval.
  888 + # @remark >0 to make the srs to send message one by one.
  889 + # @remark user can get the right packets interval in ms by srs_rtmp_dump.
  890 + # default: 0
  891 + send_min_interval 10.0;
  892 + # whether reduce the sequence header,
  893 + # for some client which cannot got duplicated sequence header,
  894 + # while the sequence header is not changed yet.
  895 + # default: off
  896 + reduce_sequence_header on;
  897 + # the 1st packet timeout in ms for encoder.
  898 + # default: 20000
  899 + publish_1stpkt_timeout 20000;
  900 + # the normal packet timeout in ms for encoder.
  901 + # default: 5000
  902 + publish_normal_timeout 7000;
  903 +}
  904 +
  905 +# the vhost for antisuck.
  906 +vhost refer.anti_suck.com {
  907 + # the common refer for play and publish.
  908 + # if the page url of client not in the refer, access denied.
  909 + # if not specified this field, allow all.
  910 + # default: not specified.
  911 + refer github.com github.io;
  912 + # refer for publish clients specified.
  913 + # the common refer is not overrided by this.
  914 + # if not specified this field, allow all.
  915 + # default: not specified.
  916 + refer_publish github.com github.io;
  917 + # refer for play clients specified.
  918 + # the common refer is not overrided by this.
  919 + # if not specified this field, allow all.
  920 + # default: not specified.
  921 + refer_play github.com github.io;
  922 +}
  923 +
  924 +# the vhost which forward publish streams.
  925 +vhost same.vhost.forward.srs.com {
  926 + # forward all publish stream to the specified server.
  927 + # this used to split/forward the current stream for cluster active-standby,
  928 + # active-active for cdn to build high available fault tolerance system.
  929 + # format: {ip}:{port} {ip_N}:{port_N}
  930 + forward 127.0.0.1:1936 127.0.0.1:1937;
  931 +}
  932 +
  933 +# the main comments for transcode
  934 +vhost example.transcode.srs.com {
  935 + # the streaming transcode configs.
  936 + transcode {
  937 + # whether the transcode enabled.
  938 + # if off, donot transcode.
  939 + # default: off.
  940 + enabled on;
  941 + # the ffmpeg
  942 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  943 + # the transcode engine for matched stream.
  944 + # all matched stream will transcoded to the following stream.
  945 + # the transcode set name(ie. hd) is optional and not used.
  946 + engine example {
  947 + # whether the engine is enabled
  948 + # default: off.
  949 + enabled on;
  950 + # input format, can be:
  951 + # off, do not specifies the format, ffmpeg will guess it.
  952 + # flv, for flv or RTMP stream.
  953 + # other format, for example, mp4/aac whatever.
  954 + # default: flv
  955 + iformat flv;
  956 + # ffmpeg filters, follows the main input.
  957 + vfilter {
  958 + # the logo input file.
  959 + i ./doc/ffmpeg-logo.png;
  960 + # the ffmpeg complex filter.
  961 + # for filters, @see: http://ffmpeg.org/ffmpeg-filters.html
  962 + filter_complex 'overlay=10:10';
  963 + }
  964 + # video encoder name. can be:
  965 + # libx264: use h.264(libx264) video encoder.
  966 + # copy: donot encoder the video stream, copy it.
  967 + # vn: disable video output.
  968 + vcodec libx264;
  969 + # video bitrate, in kbps
  970 + # @remark 0 to use source video bitrate.
  971 + # default: 0
  972 + vbitrate 1500;
  973 + # video framerate.
  974 + # @remark 0 to use source video fps.
  975 + # default: 0
  976 + vfps 25;
  977 + # video width, must be even numbers.
  978 + # @remark 0 to use source video width.
  979 + # default: 0
  980 + vwidth 768;
  981 + # video height, must be even numbers.
  982 + # @remark 0 to use source video height.
  983 + # default: 0
  984 + vheight 320;
  985 + # the max threads for ffmpeg to used.
  986 + # default: 1
  987 + vthreads 12;
  988 + # x264 profile, @see x264 -help, can be:
  989 + # high,main,baseline
  990 + vprofile main;
  991 + # x264 preset, @see x264 -help, can be:
  992 + # ultrafast,superfast,veryfast,faster,fast
  993 + # medium,slow,slower,veryslow,placebo
  994 + vpreset medium;
  995 + # other x264 or ffmpeg video params
  996 + vparams {
  997 + # ffmpeg options, @see: http://ffmpeg.org/ffmpeg.html
  998 + t 100;
  999 + # 264 params, @see: http://ffmpeg.org/ffmpeg-codecs.html#libx264
  1000 + coder 1;
  1001 + b_strategy 2;
  1002 + bf 3;
  1003 + refs 10;
  1004 + }
  1005 + # audio encoder name. can be:
  1006 + # libfdk_aac: use aac(libfdk_aac) audio encoder.
  1007 + # copy: donot encoder the audio stream, copy it.
  1008 + # an: disable audio output.
  1009 + acodec libfdk_aac;
  1010 + # audio bitrate, in kbps. [16, 72] for libfdk_aac.
  1011 + # @remark 0 to use source audio bitrate.
  1012 + # default: 0
  1013 + abitrate 70;
  1014 + # audio sample rate. for flv/rtmp, it must be:
  1015 + # 44100,22050,11025,5512
  1016 + # @remark 0 to use source audio sample rate.
  1017 + # default: 0
  1018 + asample_rate 44100;
  1019 + # audio channel, 1 for mono, 2 for stereo.
  1020 + # @remark 0 to use source audio channels.
  1021 + # default: 0
  1022 + achannels 2;
  1023 + # other ffmpeg audio params
  1024 + aparams {
  1025 + # audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
  1026 + # @remark SRS supported aac profile for HLS is: aac_low, aac_he, aac_he_v2
  1027 + profile:a aac_low;
  1028 + bsf:a aac_adtstoasc;
  1029 + }
  1030 + # output format, can be:
  1031 + # off, do not specifies the format, ffmpeg will guess it.
  1032 + # flv, for flv or RTMP stream.
  1033 + # other format, for example, mp4/aac whatever.
  1034 + # default: flv
  1035 + oformat flv;
  1036 + # output stream. variables:
  1037 + # [vhost] the input stream vhost.
  1038 + # [port] the intput stream port.
  1039 + # [app] the input stream app.
  1040 + # [stream] the input stream name.
  1041 + # [engine] the tanscode engine name.
  1042 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1043 + }
  1044 + }
  1045 +}
  1046 +# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
  1047 +vhost mirror.transcode.srs.com {
  1048 + transcode {
  1049 + enabled on;
  1050 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1051 + engine mirror {
  1052 + enabled on;
  1053 + vfilter {
  1054 + vf 'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
  1055 + }
  1056 + vcodec libx264;
  1057 + vbitrate 300;
  1058 + vfps 20;
  1059 + vwidth 768;
  1060 + vheight 320;
  1061 + vthreads 2;
  1062 + vprofile baseline;
  1063 + vpreset superfast;
  1064 + vparams {
  1065 + }
  1066 + acodec libfdk_aac;
  1067 + abitrate 45;
  1068 + asample_rate 44100;
  1069 + achannels 2;
  1070 + aparams {
  1071 + }
  1072 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1073 + }
  1074 + }
  1075 +}
  1076 +# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
  1077 +# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
  1078 +#######################################################################################################
  1079 +# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
  1080 +vhost crop.transcode.srs.com {
  1081 + transcode {
  1082 + enabled on;
  1083 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1084 + engine crop {
  1085 + enabled on;
  1086 + vfilter {
  1087 + vf 'crop=in_w-20:in_h-160:10:80';
  1088 + }
  1089 + vcodec libx264;
  1090 + vbitrate 300;
  1091 + vfps 20;
  1092 + vwidth 768;
  1093 + vheight 320;
  1094 + vthreads 2;
  1095 + vprofile baseline;
  1096 + vpreset superfast;
  1097 + vparams {
  1098 + }
  1099 + acodec libfdk_aac;
  1100 + abitrate 45;
  1101 + asample_rate 44100;
  1102 + achannels 2;
  1103 + aparams {
  1104 + }
  1105 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1106 + }
  1107 + }
  1108 +}
  1109 +# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
  1110 +vhost logo.transcode.srs.com {
  1111 + transcode {
  1112 + enabled on;
  1113 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1114 + engine logo {
  1115 + enabled on;
  1116 + vfilter {
  1117 + i ./doc/ffmpeg-logo.png;
  1118 + filter_complex 'overlay=10:10';
  1119 + }
  1120 + vcodec libx264;
  1121 + vbitrate 300;
  1122 + vfps 20;
  1123 + vwidth 768;
  1124 + vheight 320;
  1125 + vthreads 2;
  1126 + vprofile baseline;
  1127 + vpreset superfast;
  1128 + vparams {
  1129 + }
  1130 + acodec libfdk_aac;
  1131 + abitrate 45;
  1132 + asample_rate 44100;
  1133 + achannels 2;
  1134 + aparams {
  1135 + }
  1136 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1137 + }
  1138 + }
  1139 +}
  1140 +# audio transcode only.
  1141 +# for example, FMLE publish audio codec in mp3, and donot support HLS output,
  1142 +# we can transcode the audio to aac and copy video to the new stream with HLS.
  1143 +vhost audio.transcode.srs.com {
  1144 + transcode {
  1145 + enabled on;
  1146 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1147 + engine acodec {
  1148 + enabled on;
  1149 + vcodec copy;
  1150 + acodec libfdk_aac;
  1151 + abitrate 45;
  1152 + asample_rate 44100;
  1153 + achannels 2;
  1154 + aparams {
  1155 + }
  1156 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1157 + }
  1158 + }
  1159 +}
  1160 +# disable video, transcode/copy audio.
  1161 +# for example, publish pure audio stream.
  1162 +vhost vn.transcode.srs.com {
  1163 + transcode {
  1164 + enabled on;
  1165 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1166 + engine vn {
  1167 + enabled on;
  1168 + vcodec vn;
  1169 + acodec libfdk_aac;
  1170 + abitrate 45;
  1171 + asample_rate 44100;
  1172 + achannels 2;
  1173 + aparams {
  1174 + }
  1175 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1176 + }
  1177 + }
  1178 +}
  1179 +# ffmpeg-copy(forward implements by ffmpeg).
  1180 +# copy the video and audio to a new stream.
  1181 +vhost copy.transcode.srs.com {
  1182 + transcode {
  1183 + enabled on;
  1184 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1185 + engine copy {
  1186 + enabled on;
  1187 + vcodec copy;
  1188 + acodec copy;
  1189 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1190 + }
  1191 + }
  1192 +}
  1193 +# transcode all app and stream of vhost
  1194 +# the comments, read example.transcode.srs.com
  1195 +vhost all.transcode.srs.com {
  1196 + transcode {
  1197 + enabled on;
  1198 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1199 + engine ffsuper {
  1200 + enabled on;
  1201 + iformat flv;
  1202 + vfilter {
  1203 + i ./doc/ffmpeg-logo.png;
  1204 + filter_complex 'overlay=10:10';
  1205 + }
  1206 + vcodec libx264;
  1207 + vbitrate 1500;
  1208 + vfps 25;
  1209 + vwidth 768;
  1210 + vheight 320;
  1211 + vthreads 12;
  1212 + vprofile main;
  1213 + vpreset medium;
  1214 + vparams {
  1215 + t 100;
  1216 + coder 1;
  1217 + b_strategy 2;
  1218 + bf 3;
  1219 + refs 10;
  1220 + }
  1221 + acodec libfdk_aac;
  1222 + abitrate 70;
  1223 + asample_rate 44100;
  1224 + achannels 2;
  1225 + aparams {
  1226 + profile:a aac_low;
  1227 + }
  1228 + oformat flv;
  1229 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1230 + }
  1231 + engine ffhd {
  1232 + enabled on;
  1233 + vcodec libx264;
  1234 + vbitrate 1200;
  1235 + vfps 25;
  1236 + vwidth 1382;
  1237 + vheight 576;
  1238 + vthreads 6;
  1239 + vprofile main;
  1240 + vpreset medium;
  1241 + vparams {
  1242 + }
  1243 + acodec libfdk_aac;
  1244 + abitrate 70;
  1245 + asample_rate 44100;
  1246 + achannels 2;
  1247 + aparams {
  1248 + }
  1249 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1250 + }
  1251 + engine ffsd {
  1252 + enabled on;
  1253 + vcodec libx264;
  1254 + vbitrate 800;
  1255 + vfps 25;
  1256 + vwidth 1152;
  1257 + vheight 480;
  1258 + vthreads 4;
  1259 + vprofile main;
  1260 + vpreset fast;
  1261 + vparams {
  1262 + }
  1263 + acodec libfdk_aac;
  1264 + abitrate 60;
  1265 + asample_rate 44100;
  1266 + achannels 2;
  1267 + aparams {
  1268 + }
  1269 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1270 + }
  1271 + engine fffast {
  1272 + enabled on;
  1273 + vcodec libx264;
  1274 + vbitrate 300;
  1275 + vfps 20;
  1276 + vwidth 768;
  1277 + vheight 320;
  1278 + vthreads 2;
  1279 + vprofile baseline;
  1280 + vpreset superfast;
  1281 + vparams {
  1282 + }
  1283 + acodec libfdk_aac;
  1284 + abitrate 45;
  1285 + asample_rate 44100;
  1286 + achannels 2;
  1287 + aparams {
  1288 + }
  1289 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1290 + }
  1291 + engine vcopy {
  1292 + enabled on;
  1293 + vcodec copy;
  1294 + acodec libfdk_aac;
  1295 + abitrate 45;
  1296 + asample_rate 44100;
  1297 + achannels 2;
  1298 + aparams {
  1299 + }
  1300 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1301 + }
  1302 + engine acopy {
  1303 + enabled on;
  1304 + vcodec libx264;
  1305 + vbitrate 300;
  1306 + vfps 20;
  1307 + vwidth 768;
  1308 + vheight 320;
  1309 + vthreads 2;
  1310 + vprofile baseline;
  1311 + vpreset superfast;
  1312 + vparams {
  1313 + }
  1314 + acodec copy;
  1315 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1316 + }
  1317 + engine copy {
  1318 + enabled on;
  1319 + vcodec copy;
  1320 + acodec copy;
  1321 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1322 + }
  1323 + }
  1324 +}
  1325 +# transcode all stream using the empty ffmpeg demo, donothing.
  1326 +vhost ffempty.transcode.srs.com {
  1327 + transcode {
  1328 + enabled on;
  1329 + ffmpeg ./objs/research/ffempty;
  1330 + engine empty {
  1331 + enabled on;
  1332 + vcodec libx264;
  1333 + vbitrate 300;
  1334 + vfps 20;
  1335 + vwidth 768;
  1336 + vheight 320;
  1337 + vthreads 2;
  1338 + vprofile baseline;
  1339 + vpreset superfast;
  1340 + vparams {
  1341 + }
  1342 + acodec libfdk_aac;
  1343 + abitrate 45;
  1344 + asample_rate 44100;
  1345 + achannels 2;
  1346 + aparams {
  1347 + }
  1348 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  1349 + }
  1350 + }
  1351 +}
  1352 +# transcode all app and stream of app
  1353 +vhost app.transcode.srs.com {
  1354 + # the streaming transcode configs.
  1355 + # if app specified, transcode all streams of app.
  1356 + transcode live {
  1357 + enabled on;
  1358 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1359 + engine {
  1360 + enabled off;
  1361 + }
  1362 + }
  1363 +}
  1364 +# transcode specified stream.
  1365 +vhost stream.transcode.srs.com {
  1366 + # the streaming transcode configs.
  1367 + # if stream specified, transcode the matched stream.
  1368 + transcode live/livestream {
  1369 + enabled on;
  1370 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  1371 + engine {
  1372 + enabled off;
  1373 + }
  1374 + }
  1375 +}
  1376 +
  1377 +# vhost for bandwidth check
  1378 +# generally, the bandcheck vhost must be: bandcheck.srs.com,
  1379 +# or need to modify the vhost of client.
  1380 +vhost bandcheck.srs.com {
  1381 + enabled on;
  1382 + chunk_size 65000;
  1383 + # bandwidth check config.
  1384 + bandcheck {
  1385 + # whether support bandwidth check,
  1386 + # default: off.
  1387 + enabled on;
  1388 + # the key for server to valid,
  1389 + # if invalid key, server disconnect and abort the bandwidth check.
  1390 + key "35c9b402c12a7246868752e2878f7e0e";
  1391 + # the interval in seconds for bandwidth check,
  1392 + # server donot allow new test request.
  1393 + # default: 30
  1394 + interval 30;
  1395 + # the max available check bandwidth in kbps.
  1396 + # to avoid attack of bandwidth check.
  1397 + # default: 1000
  1398 + limit_kbps 4000;
  1399 + }
  1400 +}
  1401 +
  1402 +# set the chunk size of vhost.
  1403 +vhost chunksize.srs.com {
  1404 + # the default chunk size is 128, max is 65536,
  1405 + # some client does not support chunk size change,
  1406 + # vhost chunk size will override the global value.
  1407 + # default: global chunk size.
  1408 + chunk_size 128;
  1409 +}
  1410 +
  1411 +# vhost for time jitter
  1412 +vhost jitter.srs.com {
  1413 + # about the stream monotonically increasing:
  1414 + # 1. video timestamp is monotonically increasing,
  1415 + # 2. audio timestamp is monotonically increasing,
  1416 + # 3. video and audio timestamp is interleaved/mixed monotonically increasing.
  1417 + # it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
  1418 + # however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
  1419 + # the time jitter algorithm:
  1420 + # 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
  1421 + # 2. zero, only ensure sttream start at zero, ignore timestamp jitter.
  1422 + # 3. off, disable the time jitter algorithm, like atc.
  1423 + # default: full
  1424 + time_jitter full;
  1425 + # whether use the interleaved/mixed algorithm to correct the timestamp.
  1426 + # if on, always ensure the timestamp of audio+video is interleaved/mixed monotonically increase.
  1427 + # if off, use time_jitter to correct the timestamp if required.
  1428 + # default: off
  1429 + mix_correct off;
  1430 +}
  1431 +
  1432 +# vhost for atc.
  1433 +vhost atc.srs.com {
  1434 + # vhost for atc for hls/hds/rtmp backup.
  1435 + # generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
  1436 + # when atc is on, server delivery rtmp stream by absolute time.
  1437 + # atc is used, for instance, encoder will copy stream to master and slave server,
  1438 + # server use atc to delivery stream to edge/client, where stream time from master/slave server
  1439 + # is always the same, client/tools can slice RTMP stream to HLS according to the same time,
  1440 + # if the time not the same, the HLS stream cannot slice to support system backup.
  1441 + #
  1442 + # @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
  1443 + # @see http://www.baidu.com/#wd=hds%20hls%20atc
  1444 + #
  1445 + # default: off
  1446 + atc on;
  1447 + # whether enable the auto atc,
  1448 + # if enabled, detect the bravo_atc="true" in onMetaData packet,
  1449 + # set atc to on if matched.
  1450 + # always ignore the onMetaData if atc_auto is off.
  1451 + # default: on
  1452 + atc_auto on;
  1453 +}
  1454 +
  1455 +# the vhost disabled.
  1456 +vhost removed.srs.com {
  1457 + # whether the vhost is enabled.
  1458 + # if off, all request access denied.
  1459 + # default: on
  1460 + enabled off;
  1461 +}
  1462 +
  1463 +# config for the pithy print,
  1464 +# which always print constant message specified by interval,
  1465 +# whatever the clients in concurrency.
  1466 +# default: 10000
  1467 +pithy_print_ms 10000;
  1 +# the config for srs to delivery hds
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHDS
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +
  10 +vhost __defaultVhost__ {
  11 + hds {
  12 + enabled on;
  13 + hds_fragment 10;
  14 + hds_window 60;
  15 + hds_path ./objs/nginx/html;
  16 + }
  17 +}
  1 +# the config for srs to delivery hls
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + hls {
  11 + enabled on;
  12 + hls_fragment 10;
  13 + hls_window 60;
  14 + hls_path ./objs/nginx/html;
  15 + hls_m3u8_file [app]/[stream].m3u8;
  16 + hls_ts_file [app]/[stream]-[seq].ts;
  17 + }
  18 +}
  1 +# the config for srs to delivery hls
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +vhost __defaultVhost__ {
  8 + hls {
  9 + enabled on;
  10 + hls_fragment 10;
  11 + hls_window 60;
  12 + hls_path ./objs/nginx/html;
  13 + hls_m3u8_file [app]/[stream]/index.m3u8;
  14 + hls_ts_file [app]/[stream]/[seq].ts;
  15 +#if not configured,don't save total m3u8 file,and keep the ts cleaned
  16 + hls_total_m3u8_file [app]/[stream]/total.m3u8;
  17 +#if not configured,don't save total duration file
  18 + hls_total_file [app]/[stream]/total;
  19 + }
  20 +}
  1 +# the config for srs to remux rtmp to aac live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +vhost __defaultVhost__ {
  15 + http_remux {
  16 + enabled on;
  17 + fast_cache 30;
  18 + mount [vhost]/[app]/[stream].aac;
  19 + hstrs on;
  20 +
  21 + }
  22 +}
  1 +# the config for srs to remux rtmp to flv live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +vhost __defaultVhost__ {
  15 + http_remux {
  16 + enabled on;
  17 + mount [vhost]/[app]/[stream].flv;
  18 + hstrs on;
  19 + }
  20 +}
  1 +# the config for srs to remux rtmp to flv live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 19351;
  6 +max_connections 1000;
  7 +pid objs/srs.flv.19351.pid;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +http_server {
  11 + enabled on;
  12 + listen 8081;
  13 + dir ./objs/nginx/html;
  14 +}
  15 +vhost __defaultVhost__ {
  16 + mode remote;
  17 + origin 127.0.0.1;
  18 + http_remux {
  19 + enabled on;
  20 + mount [vhost]/[app]/[stream].flv;
  21 + hstrs on;
  22 + }
  23 +}
  1 +# the config for srs to remux rtmp to flv live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 19352;
  6 +max_connections 1000;
  7 +pid objs/srs.flv.19352.pid;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +http_server {
  11 + enabled on;
  12 + listen 8082;
  13 + dir ./objs/nginx/html;
  14 +}
  15 +vhost __defaultVhost__ {
  16 + mode remote;
  17 + origin 127.0.0.1;
  18 + http_remux {
  19 + enabled on;
  20 + mount [vhost]/[app]/[stream].flv;
  21 + hstrs on;
  22 + }
  23 +}
  1 +# the config for srs http heartbeat, report its info to api-server
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +heartbeat {
  9 + enabled on;
  10 + interval 9.3;
  11 + # for python api-server
  12 + url http://127.0.0.1:8085/api/v1/servers;
  13 + device_id "my-srs-device";
  14 + # for ossrs.net monitor, device_id is the key genereated by bsm.
  15 + #url http://www.ossrs.net:1977/api/v1/robots/servers;
  16 + #device_id "35c9b402c12a7246868752e2878f7e0e";
  17 + # with detail summaries
  18 + summaries on;
  19 +}
  20 +stats {
  21 + network 0;
  22 + disk sda sdb xvda xvdb;
  23 +}
  24 +vhost __defaultVhost__ {
  25 +}
  1 +# the config for srs to delivery hls
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +vhost __defaultVhost__ {
  15 + hls {
  16 + enabled on;
  17 + hls_fragment 10;
  18 + hls_window 60;
  19 + hls_path ./objs/nginx/html;
  20 + hls_m3u8_file [app]/[stream].m3u8;
  21 + hls_ts_file [app]/[stream]-[seq].ts;
  22 + }
  23 +}
  1 +# http-hooks or http-callbacks config for srs.
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +vhost __defaultVhost__ {
  9 + http_hooks {
  10 + enabled on;
  11 + on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
  12 + on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
  13 + on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
  14 + on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
  15 + on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
  16 + on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
  17 + }
  18 +}
  1 +# the config for srs to remux rtmp to mp3 live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +vhost __defaultVhost__ {
  15 + http_remux {
  16 + enabled on;
  17 + fast_cache 30;
  18 + mount [vhost]/[app]/[stream].mp3;
  19 + hstrs on;
  20 +
  21 + }
  22 +}
  1 +# the config for srs to serve as http server
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +http_server {
  9 + enabled on;
  10 + listen 8080;
  11 + dir ./objs/nginx/html;
  12 +}
  13 +vhost ossrs.net {
  14 + http {
  15 + enabled on;
  16 + mount [vhost]/;
  17 + dir ./objs/nginx/html;
  18 + }
  19 +}
  1 +# the config for srs to remux rtmp to ts live stream.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +vhost __defaultVhost__ {
  15 + http_remux {
  16 + enabled on;
  17 + mount [vhost]/[app]/[stream].ts;
  18 + hstrs on;
  19 + }
  20 +}
  1 +# use ffmpeg to ingest file/stream/device to SRS
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleIngest
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + ingest livestream {
  11 + enabled on;
  12 + input {
  13 + type file;
  14 + url ./doc/source.200kbps.768x320.flv;
  15 + }
  16 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  17 + engine {
  18 + enabled off;
  19 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
  20 + }
  21 + }
  22 +}
  1 +# no-daemon and write log to console config for srs.
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +daemon off;
  7 +srs_log_tank console;
  8 +http_api {
  9 + enabled on;
  10 + listen 1985;
  11 +}
  12 +http_server {
  13 + enabled on;
  14 + listen 8080;
  15 +}
  16 +vhost __defaultVhost__ {
  17 + hls {
  18 + enabled on;
  19 + hls_fragment 10;
  20 + hls_window 60;
  21 + hls_path ./objs/nginx/html;
  22 + }
  23 + ingest livestream {
  24 + enabled on;
  25 + input {
  26 + type file;
  27 + url ./doc/source.200kbps.768x320.flv;
  28 + }
  29 + #ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  30 + ffmpeg ./objs/research/librtmp/srs_ingest_flv;
  31 + engine {
  32 + enabled off;
  33 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
  34 + }
  35 + }
  36 +}
  1 +# the config for srs origin-edge cluster
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 19350;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +pid ./objs/origin.pid;
  10 +vhost __defaultVhost__ {
  11 +}
  1 +# push HTTP FLV to SRS.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-http-flv-to-srs
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +http_server {
  10 + enabled on;
  11 + listen 8080;
  12 + dir ./objs/nginx/html;
  13 +}
  14 +stream_caster {
  15 + enabled on;
  16 + caster flv;
  17 + output rtmp://127.0.0.1/[app]/[stream];
  18 + listen 8936;
  19 +}
  20 +vhost __defaultVhost__ {
  21 + hls {
  22 + enabled on;
  23 + hls_fragment 10;
  24 + hls_window 60;
  25 + hls_path ./objs/nginx/html;
  26 + hls_m3u8_file [app]/[stream].m3u8;
  27 + hls_ts_file [app]/[stream]-[seq].ts;
  28 + }
  29 +}
  1 +# push MPEG-TS over UDP to SRS.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-mpeg-ts-over-udp
  3 +# @see https://github.com/ossrs/srs/issues/250#issuecomment-72321769
  4 +# @see full.conf for detail config.
  5 +
  6 +listen 1935;
  7 +max_connections 1000;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +stream_caster {
  11 + enabled on;
  12 + caster mpegts_over_udp;
  13 + output rtmp://127.0.0.1/live/livestream;
  14 + listen 1935;
  15 +}
  16 +vhost __defaultVhost__ {
  17 +}
  1 +# push MPEG-TS over UDP to SRS.
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_Streamer#push-mpeg-ts-over-udp
  3 +# @see https://github.com/ossrs/srs/issues/250#issuecomment-72321769
  4 +# @see full.conf for detail config.
  5 +
  6 +listen 1935;
  7 +max_connections 1000;
  8 +daemon off;
  9 +srs_log_tank console;
  10 +stream_caster {
  11 + enabled on;
  12 + caster rtsp;
  13 + output rtmp://127.0.0.1/[app]/[stream];
  14 + listen 554;
  15 + rtp_port_min 57200;
  16 + rtp_port_max 57300;
  17 +}
  18 +vhost __defaultVhost__ {
  19 +}
  1 +# the config for srs to delivery realtime RTMP stream
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_SampleRealtime
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + gop_cache off;
  11 + queue_length 10;
  12 + min_latency on;
  13 + mr {
  14 + enabled off;
  15 + }
  16 + mw_latency 100;
  17 + tcp_nodelay on;
  18 +}
  1 +# the config for srs to delivery RTMP
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleRTMP
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 +}
  1 +# security config for srs, allow play and deny publish.
  2 +# @see https://github.com/ossrs/srs/issues/211#issuecomment-68507035
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + security {
  11 + enabled on;
  12 + deny publish all;
  13 + allow play all;
  14 + }
  15 +}
  1 +# main config for srs.
  2 +# @see full.conf for detail config.
  3 +
  4 +listen 1935;
  5 +max_connections 1000;
  6 +srs_log_tank file;
  7 +srs_log_file ./objs/srs.log;
  8 +http_api {
  9 + enabled on;
  10 + listen 1985;
  11 +}
  12 +http_server {
  13 + enabled on;
  14 + listen 8080;
  15 + dir ./objs/nginx/html;
  16 +}
  17 +stats {
  18 + network 0;
  19 + disk sda sdb xvda xvdb;
  20 +}
  21 +vhost __defaultVhost__ {
  22 +}
  1 +# the config for srs to delivery hls
  2 +# @see https://github.com/ossrs/srs/wiki/v1_CN_SampleHLS
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 + hls {
  11 + enabled on;
  12 + hls_fragment 10;
  13 + hls_window 60;
  14 + hls_path ./objs/nginx/html;
  15 + hls_m3u8_file [app]/[stream].m3u8;
  16 + hls_ts_file [app]/[stream]-[seq].ts;
  17 + }
  18 + transcode {
  19 + enabled on;
  20 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  21 + engine ff {
  22 + enabled on;
  23 + vcodec copy;
  24 + acodec libfdk_aac;
  25 + aparams {
  26 + }
  27 + output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
  28 + }
  29 + }
  30 +}
  1 +# the config for srs origin-edge cluster
  2 +# @see https://github.com/ossrs/srs/wiki/v2_CN_Edge#transform-vhost
  3 +# @see full.conf for detail config.
  4 +
  5 +listen 1935;
  6 +max_connections 1000;
  7 +daemon off;
  8 +srs_log_tank console;
  9 +vhost __defaultVhost__ {
  10 +}
  11 +vhost source.srs.com {
  12 + ingest livestream {
  13 + enabled on;
  14 + input {
  15 + type file;
  16 + url ./doc/source.200kbps.768x320.flv;
  17 + }
  18 + ffmpeg ./objs/ffmpeg/bin/ffmpeg;
  19 + engine {
  20 + enabled off;
  21 + output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
  22 + }
  23 + }
  24 +}
  25 +vhost transform.srs.edge.com {
  26 + mode remote;
  27 + origin 127.0.0.1:1935;
  28 + vhost source.srs.com;
  29 +}
  1 +#!/bin/bash
  2 +
  3 +### BEGIN INIT INFO
  4 +# Provides: ossrs(srs)
  5 +# RequiRED-Start: $all
  6 +# RequiRED-Stop: $all
  7 +# Default-Start: 2 3 4 5
  8 +# Default-Stop: 0 1 6
  9 +# Short-Description: ossrs(srs)
  10 +# Description: https://github.com/ossrs/srs
  11 +### END INIT INFO
  12 +
  13 +# the config of ROOT, user must modify it when start srs from other directory,
  14 +# it's ok to use the script by command ./etc/init.d/ossrs
  15 +ROOT="/usr/local/srs"
  16 +APP="./objs/srs"
  17 +CONFIG="./conf/srs.conf"
  18 +DEFAULT_PID_FILE='./objs/srs.pid'
  19 +DEFAULT_LOG_FILE='./objs/srs.log'
  20 +
  21 +########################################################################
  22 +# utility functions
  23 +########################################################################
  24 +RED="\\033[31m"
  25 +GREEN="\\033[32m"
  26 +YELLOW="\\033[33m"
  27 +BLACK="\\033[0m"
  28 +POS="\\033[60G"
  29 +
  30 +ok_msg() {
  31 + echo -e "${1}${POS}${BLACK}[${GREEN} OK ${BLACK}]"
  32 +}
  33 +
  34 +failed_msg() {
  35 + echo -e "${1}${POS}${BLACK}[${RED}FAILED${BLACK}]"
  36 +}
  37 +
  38 +# load process info of srs
  39 +# @set variable $srs_pid to the process id in srs.pid file.
  40 +# @return 0, if process exists; otherwise:
  41 +# 1, for pid file not exists.
  42 +# 2, for get proecess info by pid failed.
  43 +# @set variable $error_msg if error.
  44 +# @set variable $pid_file to pid file.
  45 +load_process_info() {
  46 + # get pid file
  47 + pid_file=`cd ${ROOT} && cat ${CONFIG} |grep ^pid|awk '{print $2}'|awk -F ';' '{print $1}'`
  48 + if [[ -z $pid_file ]]; then pid_file=${DEFAULT_PID_FILE}; fi
  49 + # get abs path
  50 + pid_dir=`dirname $pid_file`
  51 + pid_file=`(cd ${ROOT}; cd $pid_dir; pwd)`/`basename $pid_file`
  52 +
  53 + srs_pid=`cat $pid_file 2>/dev/null`
  54 + ret=$?; if [[ 0 -ne $ret ]]; then error_msg="file $pid_file does not exists"; return 1; fi
  55 +
  56 + ps -p ${srs_pid} >/dev/null 2>/dev/null
  57 + ret=$?; if [[ 0 -ne $ret ]]; then error_msg="process $srs_pid does not exists"; return 2; fi
  58 +
  59 + return 0;
  60 +}
  61 +
  62 +start() {
  63 + # if exists, exit.
  64 + load_process_info
  65 + if [[ 0 -eq $? ]]; then failed_msg "SRS started(pid ${srs_pid}), should not start it again."; return 0; fi
  66 +
  67 + # not exists, start server
  68 + ok_msg "Starting SRS..."
  69 +
  70 + # get log file
  71 + log_file=`cd ${ROOT} && cat ${CONFIG} |grep '^log_file'| awk '{print $2}'| awk -F ';' '{print $1}'`
  72 + if [[ -z $log_file ]]; then log_file=${DEFAULT_LOG_FILE}; fi
  73 + # get abs path
  74 + log_dir=`dirname $log_file`
  75 + log_file=`(cd ${ROOT} && cd $log_dir && pwd)`/`basename $log_file`
  76 +
  77 + # TODO: FIXME: set limit by, for instance, "ulimit -HSn 10000"
  78 + if [[ -z $log_file ]]; then
  79 + (ulimit -c unlimited && cd ${ROOT}; ${APP} -c ${CONFIG} >/dev/null 2>&1)
  80 + else
  81 + (ulimit -c unlimited && cd ${ROOT}; ${APP} -c ${CONFIG} >> $log_file 2>&1)
  82 + fi
  83 +
  84 + # check again after start server
  85 + for ((i = 0; i < 5; i++)); do
  86 + # sleep a little while, for srs may start then crash.
  87 + sleep 0.1
  88 + load_process_info
  89 + ret=$?; if [[ 0 -ne $ret ]]; then
  90 + failed_msg "SRS start failed";
  91 + failed_msg "see $log_file";
  92 + return $ret;
  93 + fi
  94 + done
  95 +
  96 + # check whether started.
  97 + load_process_info
  98 + ret=$?; if [[ 0 -eq $? ]]; then ok_msg "SRS started(pid ${srs_pid})"; return 0; fi
  99 +
  100 + failed_msg "SRS not started"
  101 + return $ret
  102 +}
  103 +
  104 +stop() {
  105 + # not start, exit
  106 + load_process_info
  107 + if [[ 0 -ne $? ]]; then failed_msg "SRS not start."; return 0; fi
  108 +
  109 + ok_msg "Stopping SRS(pid ${srs_pid})..."
  110 +
  111 + # process exists, try to kill to stop normally
  112 + for((i=0;i<100;i++)); do
  113 + load_process_info
  114 + if [[ 0 -eq $? ]]; then
  115 + kill -s SIGTERM ${srs_pid} 2>/dev/null
  116 + ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "send signal SIGTERM failed ret=$ret"; return $ret; fi
  117 + sleep 0.3
  118 + else
  119 + ok_msg "SRS stopped by SIGTERM"
  120 + # delete the pid file when stop success.
  121 + rm -f ${pid_file}
  122 + break;
  123 + fi
  124 + done
  125 +
  126 + # process exists, use kill -9 to force to exit
  127 + load_process_info
  128 + if [[ 0 -eq $? ]]; then
  129 + kill -s SIGKILL ${srs_pid} 2>/dev/null
  130 + ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "send signal SIGKILL failed ret=$ret"; return $ret; fi
  131 + ok_msg "SRS stopped by SIGKILL"
  132 + else
  133 + # delete the pid file when stop success.
  134 + rm -f ${pid_file}
  135 + fi
  136 +
  137 + sleep 0.1
  138 + return 0
  139 +}
  140 +
  141 +# get the status of srs process
  142 +# @return 0 if srs is running; otherwise, 1 for stopped.
  143 +status() {
  144 + load_process_info
  145 + ret=$?; if [[ 0 -eq $ret ]]; then echo "SRS(pid ${srs_pid}) is running."; return 0; fi
  146 +
  147 + echo "SRS is stopped, $error_msg"
  148 + return 1
  149 +}
  150 +
  151 +reload() {
  152 + # not start, exit
  153 + load_process_info
  154 + if [[ 0 -ne $? ]]; then failed_msg "SRS not start."; return 0; fi
  155 +
  156 + ok_msg "Reload SRS(pid ${srs_pid})..."
  157 +
  158 + # process exists, reload it
  159 + kill -s SIGHUP ${srs_pid} 2>/dev/null
  160 + ret=$?; if [[ 0 -ne $ret ]]; then failed_msg "Reload SRS failed ret=$ret"; return $ret; fi
  161 +
  162 + load_process_info
  163 + if [[ 0 -ne $? ]]; then failed_msg "SRS reload failed."; return $ret; fi
  164 +
  165 + ok_msg "SRS reloaded"
  166 + return 0
  167 +}
  168 +
  169 +menu() {
  170 + case "$1" in
  171 + start)
  172 + start
  173 + ;;
  174 + stop)
  175 + stop
  176 + ;;
  177 + restart)
  178 + stop
  179 + start
  180 + ;;
  181 + status)
  182 + status
  183 + ;;
  184 + reload)
  185 + reload
  186 + ;;
  187 + *)
  188 + echo "Usage: $0 {start|stop|status|restart|reload}"
  189 + return 1
  190 + ;;
  191 + esac
  192 +}
  193 +
  194 +menu $1
  195 +
  196 +code=$?
  197 +exit ${code}
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  1 +<cross-domain-policy>
  2 + <allow-access-from domain="*"/>
  3 +</cross-domain-policy>
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