diff --git b/conf/bandwidth.conf a/conf/bandwidth.conf
new file mode 100644
index 0000000..a0bf060
--- /dev/null
+++ a/conf/bandwidth.conf
@@ -0,0 +1,20 @@
+# 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;
+ }
+}
diff --git b/conf/console.conf a/conf/console.conf
new file mode 100644
index 0000000..6814346
--- /dev/null
+++ a/conf/console.conf
@@ -0,0 +1,17 @@
+# 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__ {
+}
diff --git b/conf/demo.19350.conf a/conf/demo.19350.conf
new file mode 100644
index 0000000..a6dd3ef
--- /dev/null
+++ a/conf/demo.19350.conf
@@ -0,0 +1,22 @@
+# 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;
+ }
+}
+
diff --git b/conf/demo.conf a/conf/demo.conf
new file mode 100644
index 0000000..b6fb87a
--- /dev/null
+++ a/conf/demo.conf
@@ -0,0 +1,220 @@
+# 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;
+ }
+}
diff --git b/conf/dvr.path.conf a/conf/dvr.path.conf
new file mode 100644
index 0000000..c5951f0
--- /dev/null
+++ a/conf/dvr.path.conf
@@ -0,0 +1,18 @@
+# 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;
+ }
+}
diff --git b/conf/dvr.segment.conf a/conf/dvr.segment.conf
new file mode 100644
index 0000000..2265ad0
--- /dev/null
+++ a/conf/dvr.segment.conf
@@ -0,0 +1,17 @@
+# 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;
+ }
+}
diff --git b/conf/dvr.session.conf a/conf/dvr.session.conf
new file mode 100644
index 0000000..8e8d893
--- /dev/null
+++ a/conf/dvr.session.conf
@@ -0,0 +1,15 @@
+# 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;
+ }
+}
diff --git b/conf/edge.conf a/conf/edge.conf
new file mode 100644
index 0000000..2a8662f
--- /dev/null
+++ a/conf/edge.conf
@@ -0,0 +1,13 @@
+# 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;
+}
diff --git b/conf/edge.token.traverse.conf a/conf/edge.token.traverse.conf
new file mode 100644
index 0000000..bad2412
--- /dev/null
+++ a/conf/edge.token.traverse.conf
@@ -0,0 +1,13 @@
+# 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;
+}
diff --git b/conf/ffmpeg.transcode.conf a/conf/ffmpeg.transcode.conf
new file mode 100644
index 0000000..d785a44
--- /dev/null
+++ a/conf/ffmpeg.transcode.conf
@@ -0,0 +1,29 @@
+# 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];
+ }
+ }
+}
diff --git b/conf/forward.master.conf a/conf/forward.master.conf
new file mode 100644
index 0000000..5566caa
--- /dev/null
+++ a/conf/forward.master.conf
@@ -0,0 +1,14 @@
+# 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;
+}
diff --git b/conf/forward.slave.conf a/conf/forward.slave.conf
new file mode 100644
index 0000000..98286e3
--- /dev/null
+++ a/conf/forward.slave.conf
@@ -0,0 +1,11 @@
+# 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__ {
+}
diff --git b/conf/full.conf a/conf/full.conf
new file mode 100644
index 0000000..4edac7d
--- /dev/null
+++ a/conf/full.conf
@@ -0,0 +1,1467 @@
+# 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;
diff --git b/conf/hds.conf a/conf/hds.conf
new file mode 100644
index 0000000..c2a7d43
--- /dev/null
+++ a/conf/hds.conf
@@ -0,0 +1,17 @@
+# 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;
+ }
+}
diff --git b/conf/hls.conf a/conf/hls.conf
new file mode 100644
index 0000000..d0fa423
--- /dev/null
+++ a/conf/hls.conf
@@ -0,0 +1,18 @@
+# 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;
+ }
+}
diff --git b/conf/hls_total.conf a/conf/hls_total.conf
new file mode 100644
index 0000000..4dad424
--- /dev/null
+++ a/conf/hls_total.conf
@@ -0,0 +1,20 @@
+# 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;
+ }
+}
diff --git b/conf/http.aac.live.conf a/conf/http.aac.live.conf
new file mode 100644
index 0000000..22aa1bd
--- /dev/null
+++ a/conf/http.aac.live.conf
@@ -0,0 +1,22 @@
+# 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;
+
+ }
+}
diff --git b/conf/http.flv.live.conf a/conf/http.flv.live.conf
new file mode 100644
index 0000000..944889f
--- /dev/null
+++ a/conf/http.flv.live.conf
@@ -0,0 +1,20 @@
+# 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;
+ }
+}
diff --git b/conf/http.flv.live.edge1.conf a/conf/http.flv.live.edge1.conf
new file mode 100644
index 0000000..e827c22
--- /dev/null
+++ a/conf/http.flv.live.edge1.conf
@@ -0,0 +1,23 @@
+# 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;
+ }
+}
diff --git b/conf/http.flv.live.edge2.conf a/conf/http.flv.live.edge2.conf
new file mode 100644
index 0000000..669c380
--- /dev/null
+++ a/conf/http.flv.live.edge2.conf
@@ -0,0 +1,23 @@
+# 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;
+ }
+}
diff --git b/conf/http.heartbeat.conf a/conf/http.heartbeat.conf
new file mode 100644
index 0000000..d15ff6a
--- /dev/null
+++ a/conf/http.heartbeat.conf
@@ -0,0 +1,25 @@
+# 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__ {
+}
diff --git b/conf/http.hls.conf a/conf/http.hls.conf
new file mode 100644
index 0000000..17416a2
--- /dev/null
+++ a/conf/http.hls.conf
@@ -0,0 +1,23 @@
+# 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;
+ }
+}
diff --git b/conf/http.hooks.callback.conf a/conf/http.hooks.callback.conf
new file mode 100644
index 0000000..d5374d9
--- /dev/null
+++ a/conf/http.hooks.callback.conf
@@ -0,0 +1,18 @@
+# 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;
+ }
+}
diff --git b/conf/http.mp3.live.conf a/conf/http.mp3.live.conf
new file mode 100644
index 0000000..690a453
--- /dev/null
+++ a/conf/http.mp3.live.conf
@@ -0,0 +1,22 @@
+# 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;
+
+ }
+}
diff --git b/conf/http.server.conf a/conf/http.server.conf
new file mode 100644
index 0000000..0c84089
--- /dev/null
+++ a/conf/http.server.conf
@@ -0,0 +1,19 @@
+# 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;
+ }
+}
diff --git b/conf/http.ts.live.conf a/conf/http.ts.live.conf
new file mode 100644
index 0000000..d445672
--- /dev/null
+++ a/conf/http.ts.live.conf
@@ -0,0 +1,20 @@
+# 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;
+ }
+}
diff --git b/conf/ingest.conf a/conf/ingest.conf
new file mode 100644
index 0000000..d77a2f9
--- /dev/null
+++ a/conf/ingest.conf
@@ -0,0 +1,22 @@
+# 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;
+ }
+ }
+}
diff --git b/conf/mac.dev.conf a/conf/mac.dev.conf
new file mode 100644
index 0000000..e624e16
--- /dev/null
+++ a/conf/mac.dev.conf
@@ -0,0 +1,36 @@
+# 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;
+ }
+ }
+}
diff --git b/conf/origin.conf a/conf/origin.conf
new file mode 100644
index 0000000..f27abb0
--- /dev/null
+++ a/conf/origin.conf
@@ -0,0 +1,11 @@
+# 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__ {
+}
diff --git b/conf/push.flv.conf a/conf/push.flv.conf
new file mode 100644
index 0000000..9e871f9
--- /dev/null
+++ a/conf/push.flv.conf
@@ -0,0 +1,29 @@
+# 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;
+ }
+}
diff --git b/conf/push.mpegts.over.udp.conf a/conf/push.mpegts.over.udp.conf
new file mode 100644
index 0000000..82f2f04
--- /dev/null
+++ a/conf/push.mpegts.over.udp.conf
@@ -0,0 +1,17 @@
+# 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__ {
+}
diff --git b/conf/push.rtsp.conf a/conf/push.rtsp.conf
new file mode 100644
index 0000000..016a067
--- /dev/null
+++ a/conf/push.rtsp.conf
@@ -0,0 +1,19 @@
+# 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__ {
+}
diff --git b/conf/realtime.conf a/conf/realtime.conf
new file mode 100644
index 0000000..249c750
--- /dev/null
+++ a/conf/realtime.conf
@@ -0,0 +1,18 @@
+# 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;
+}
diff --git b/conf/rtmp.conf a/conf/rtmp.conf
new file mode 100644
index 0000000..e05f154
--- /dev/null
+++ a/conf/rtmp.conf
@@ -0,0 +1,10 @@
+# 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__ {
+}
diff --git b/conf/security.deny.publish.conf a/conf/security.deny.publish.conf
new file mode 100644
index 0000000..fc2a2e8
--- /dev/null
+++ a/conf/security.deny.publish.conf
@@ -0,0 +1,15 @@
+# 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;
+ }
+}
diff --git b/conf/srs.conf a/conf/srs.conf
new file mode 100644
index 0000000..da06284
--- /dev/null
+++ a/conf/srs.conf
@@ -0,0 +1,22 @@
+# 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__ {
+}
diff --git b/conf/transcode2hls.audio.only.conf a/conf/transcode2hls.audio.only.conf
new file mode 100644
index 0000000..f4361bf
--- /dev/null
+++ a/conf/transcode2hls.audio.only.conf
@@ -0,0 +1,30 @@
+# 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];
+ }
+ }
+}
diff --git b/conf/transform.edge.conf a/conf/transform.edge.conf
new file mode 100644
index 0000000..5b3a04f
--- /dev/null
+++ a/conf/transform.edge.conf
@@ -0,0 +1,29 @@
+# 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;
+}
diff --git b/etc/init.d/srs a/etc/init.d/srs
new file mode 100755
index 0000000..f04a432
--- /dev/null
+++ a/etc/init.d/srs
@@ -0,0 +1,197 @@
+#!/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}
diff --git b/objs/ffmpeg/bin/ffmpeg a/objs/ffmpeg/bin/ffmpeg
new file mode 100755
index 0000000..ae01263
Binary files /dev/null and a/objs/ffmpeg/bin/ffmpeg differ
diff --git b/objs/nginx/html/crossdomain.xml a/objs/nginx/html/crossdomain.xml
new file mode 100644
index 0000000..ae91084
--- /dev/null
+++ a/objs/nginx/html/crossdomain.xml
@@ -0,0 +1,3 @@
+<cross-domain-policy>
+ <allow-access-from domain="*"/>
+</cross-domain-policy>
diff --git b/objs/srs a/objs/srs
new file mode 100755
index 0000000..f2ac7ce
Binary files /dev/null and a/objs/srs differ