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# All configurations for SRS, you can find whatever you want about configuration of SRS.
# Note that please never use this file as configuration of SRS, because it's just a full set of configurations.
#
#
#
#
#
#
#********************************************************************************************
#********************************************************************************************
# CAUTION: The features below are quite stable and actively maintained.
#********************************************************************************************
#********************************************************************************************
#############################################################################################
# Global sections
#############################################################################################
# Config file, specified by cli such as `-c conf/srs.conf`.
# Overwrite by env SRS_CONFIG_FILE
# The id of server, for stat and api identification.
# Note that SRS will generate a random id if not configured.
# Overwrite by env SRS_SERVER_ID
server_id srs-ie193id;
# The pid file to write the pid, for managing the SRS process and avoiding duplicated processes.
# If need to run multiple processes, please change this pid file to another one.
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_PID
# Default: ./objs/srs.pid
pid ./objs/srs.pid;
# the log dir for FFMPEG.
# if enabled ffmpeg, each transcoding stream will create a log file.
# /dev/null to disable the log.
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_FF_LOG_DIR
# default: ./objs
ff_log_dir ./objs;
# the log level for FFMPEG.
# info warning error fatal panic quiet
# trace debug verbose
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_FF_LOG_LEVEL
# default: info
ff_log_level info;
# 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.
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_LOG_TANK or SRS_SRS_LOG_TANK
# default: file.
srs_log_tank console;
# The log level for logging to console or file. It can be:
# verbose, info, trace, warn, error
# If configure --log-level_v2=off, use SRS 4.0 level specs which is v1, the level text is:
# Verb, Info, Trace, Warn, Error
# If configure --log-level_v2=on, use SRS 5.0 level specs which is v2, the level text is:
# TRACE, DEBUG, INFO, WARN, ERROR
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_LOG_LEVEL or SRS_SRS_LOG_LEVEL
# default: trace
srs_log_level trace;
# The log level v2, rewrite the config srs_log_level if not empty, it can be:
# trace, debug, info, warn, error
# If configure --log-level_v2=off, use SRS 4.0 level specs which is v1, the level text is:
# Verb, Info, Trace, Warn, Error
# If configure --log-level_v2=on, use SRS 5.0 level specs which is v2, the level text is:
# TRACE, DEBUG, INFO, WARN, ERROR
# Overwrite by env SRS_LOG_LEVEL_V2 or SRS_SRS_LOG_LEVEL_V2
srs_log_level_v2 info;
# when srs_log_tank is file, specifies the log file.
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_LOG_FILE or SRS_SRS_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.
# Overwrite by env SRS_MAX_CONNECTIONS
# default: 1000
max_connections 1000;
# whether start as daemon
# @remark: do not support reload.
# Overwrite by env SRS_DAEMON
# default: on
daemon off;
# 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.
# Note: Do not support reloading, for SRS5+
# Overwrite by env SRS_UTC_TIME
# default: off
utc_time off;
# config for the pithy print in ms,
# which always print constant message specified by interval,
# whatever the clients in concurrency.
# Overwrite by env SRS_PITHY_PRINT_MS
# default: 10000
pithy_print_ms 10000;
# 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.
# Overwrite by env SRS_WORK_DIR
# 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.
# Overwrite by env SRS_ASPROCESS
# default: off
asprocess off;
# Whether client empty IP is ok, for example, health checking by SLB.
# If ok(on), we will ignore this connection without warnings or errors.
# Overwrite by env SRS_EMPTY_IP_OK
# default: on
empty_ip_ok on;
# Whether in docker. When SRS starting, it will detect the docker, however
# it might detect failed, then read this config.
# Overwrite by env SRS_IN_DOCKER
# Default: off
in_docker off;
# For gracefully quit, wait for a while then close listeners,
# because K8S notify SRS with SIGQUIT and update Service simultaneously,
# maybe there is some new connections incoming before Service updated.
# @see https://github.com/ossrs/srs/issues/1595#issuecomment-587516567
# Overwrite by env SRS_GRACE_START_WAIT
# default: 2300
grace_start_wait 2300;
# For gracefully quit, final wait for cleanup in milliseconds.
# @see https://github.com/ossrs/srs/issues/1579#issuecomment-587414898
# Overwrite by env SRS_GRACE_FINAL_WAIT
# default: 3200
grace_final_wait 3200;
# Whether force gracefully quit, never fast quit.
# By default, SIGTERM which means fast quit, is sent by K8S, so we need to
# force SRS to treat SIGTERM as gracefully quit for gray release or canary.
# @see https://github.com/ossrs/srs/issues/1579#issuecomment-587475077
# Overwrite by env SRS_FORCE_GRACE_QUIT
# default: off
force_grace_quit off;
# Whether disable daemon for docker.
# If on, it will set daemon to off in docker, even daemon is on.
# Overwrite by env SRS_DISABLE_DAEMON_FOR_DOCKER
# default: on
disable_daemon_for_docker on;
# Whether auto reload by watching the config file by inotify.
# Overwrite by env SRS_INOTIFY_AUTO_RELOAD
# default: off
inotify_auto_reload off;
# Whether enable inotify_auto_reload for docker.
# If on, it will set inotify_auto_reload to on in docker, even it's off.
# Overwrite by env SRS_AUTO_RELOAD_FOR_DOCKER
# default: on
auto_reload_for_docker on;
# For tcmalloc, the release rate.
# @see https://gperftools.github.io/gperftools/tcmalloc.html
# @remark Should run configure --with-gperf
# Overwrite by env SRS_TCMALLOC_RELEASE_RATE
# default: 0.8
tcmalloc_release_rate 0.8;
#############################################################################################
# RTMP/RTMPS sections
#############################################################################################
# the rtmp server section
rtmp {
# The rtmp listen endpoints, split by space, each listen entry is <[ip:]port>. The ip can be
# ipv4, ipv6, or both. For example:
# listen 1935 [::]:1935 192.168.1.100:1935 10.10.10.100:1935;
# Overwrite by env SRS_LISTEN
listen 1935;
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# however, most clients support it and it can improve
# performance about 10%.
# Overwrite by env SRS_CHUNK_SIZE
# default: 60000
chunk_size 60000;
}
# the rtmps server section
rtmps {
# Whether rtmps is enabled.
# Overwrite by env SRS_RTMPS_ENABLED
# default: off
enabled on;
# The rtmps listen endpoints, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 1443 [::]:1443;
# Overwrite by env SRS_RTMPS_LISTEN
listen 1443;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# Overwrite by env SRS_RTMPS_KEY
# default: ./conf/server.key
key ./conf/server.key;
# The SSL public cert file, generated by:
# openssl req -new -x509 -key server.key -out server.crt -days 3650 -subj "/C=CA/ST=Toronto/L=Toronto/O=Me/OU=Me/CN=ossrs.io"
# Overwrite by env SRS_RTMPS_CERT
# default: ./conf/server.crt
cert ./conf/server.crt;
}
# vhost level config for RTMP/RTMPS.
vhost rtmp.rtmps.vhost.srs.com {
# whether enable min delay mode for vhost.
# for min latency mode:
# 1. disable the publish.mr for vhost.
# 2. use timeout for cond wait for consumer queue.
# @see https://github.com/ossrs/srs/issues/257
# default: off (for RTMP/HTTP-FLV)
# default: on (for WebRTC)
min_latency off;
# whether enable the TCP_NODELAY
# if on, set the nodelay of fd by setsockopt
# Overwrite by env SRS_VHOST_TCP_NODELAY for all vhosts.
# default: off
tcp_nodelay off;
# 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.
# Overwrite by env SRS_VHOST_CHUNK_SIZE for all vhosts.
# default: global chunk size.
chunk_size 128;
# The input ack size, 0 to not set.
# Generally, it's set by the message from peer,
# but for some peer(encoder), it never send message but use a different ack size.
# We can chnage the default ack size in server-side, to send acknowledge message,
# or the encoder maybe blocked after publishing for some time.
# Overwrite by env SRS_VHOST_IN_ACK_SIZE for all vhosts.
# Default: 0
in_ack_size 0;
# The output ack size, 0 to not set.
# This is used to notify the peer(player) to send acknowledge to server.
# Overwrite by env SRS_VHOST_OUT_ACK_SIZE for all vhosts.
# Default: 2500000
out_ack_size 2500000;
# for play client, both RTMP and other HTTP stream clients,
# for instance, the HTTP FLV stream clients.
play {
# 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.
# Overwrite by env SRS_VHOST_PLAY_GOP_CACHE for all vhosts.
# default: on
gop_cache off;
# Limit the max frames in gop cache. It might cause OOM if video stream has no IDR frame, so we limit to N
# frames by default. Note that it's the size of gop cache, including videos, audios and other messages.
# Overwrite by env SRS_VHOST_PLAY_GOP_CACHE_MAX_FRAMES for all vhosts.
# default: 2500
gop_cache_max_frames 2500;
# the max live queue length in seconds.
# if the messages in the queue exceed the max length,
# drop the old whole gop.
# Overwrite by env SRS_VHOST_PLAY_QUEUE_LENGTH for all vhosts.
# default: 30
queue_length 10;
# 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 stream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# @remark for full, correct timestamp only when |delta| > 250ms.
# @remark disabled when atc is on.
# Overwrite by env SRS_VHOST_PLAY_TIME_JITTER for all vhosts.
# default: full
time_jitter full;
# 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
#
# @remark when atc is on, auto off the time_jitter
# Overwrite by env SRS_VHOST_PLAY_ATC for all vhosts.
# default: off
atc off;
# 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.
# @remark to use mix_correct, atc should on(or time_jitter should off).
# Overwrite by env SRS_VHOST_PLAY_MIX_CORRECT for all vhosts.
# default: off
mix_correct off;
# 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.
# Overwrite by env SRS_VHOST_PLAY_ATC_AUTO for all vhosts.
# default: off
atc_auto off;
# 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]
# @remark For WebRTC, we enable pass-by-timestamp mode, so we ignore this config.
# default: 350 (For RTMP/HTTP-FLV)
# Overwrite by env SRS_VHOST_PLAY_MW_LATENCY for all vhosts.
# default: 0 (For WebRTC)
mw_latency 350;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, min_latency on)
# default: 1 (For WebRTC, min_latency off)
# default: 8 (For RTMP/HTTP-FLV, min_latency off).
# Overwrite by env SRS_VHOST_PLAY_MW_MSGS for all vhosts.
mw_msgs 8;
# 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.
# Overwrite by env SRS_VHOST_PLAY_SEND_MIN_INTERVAL for all vhosts.
# 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.
# Overwrite by env SRS_VHOST_PLAY_REDUCE_SEQUENCE_HEADER for all vhosts.
# default: off
reduce_sequence_header on;
}
# the config for FMLE/Flash publisher, which push RTMP to SRS.
publish {
# when enabled the mr, SRS will read as large as possible.
# Overwrite by env SRS_VHOST_PUBLISH_MR for all vhosts.
# default: off
mr off;
# 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 at least:
# 183KB * 2500 = 446MB
# the recommended value is [300, 2000]
# Overwrite by env SRS_VHOST_PUBLISH_MR_LATENCY for all vhosts.
# default: 350
mr_latency 350;
# the 1st packet timeout in ms for encoder.
# Overwrite by env SRS_VHOST_PUBLISH_FIRSTPKT_TIMEOUT for all vhosts.
# default: 20000
firstpkt_timeout 20000;
# the normal packet timeout in ms for encoder.
# Overwrite by env SRS_VHOST_PUBLISH_NORMAL_TIMEOUT for all vhosts.
# default: 5000
normal_timeout 7000;
# 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.
# @remark If disabled, HLS might never update the sps/pps, it depends on this.
# Overwrite by env SRS_VHOST_PUBLISH_PARSE_SPS for all vhosts.
# default: on
parse_sps on;
# When parsing SPS/PPS, whether try ANNEXB first. If not, try IBMF first, then ANNEXB.
# Overwrite by env SRS_VHOST_PUBLISH_TRY_ANNEXB_FIRST for all vhosts.
# default: on
try_annexb_first on;
# The timeout in seconds to disconnect publisher when idle, which means no players.
# Note that 0 means no timeout or this feature is disabled.
# Note that this feature conflicts with forward, because it disconnect the publisher stream.
# Overwrite by env SRS_VHOST_PUBLISH_KICKOFF_FOR_IDLE for all vhosts.
# default: 0
kickoff_for_idle 0;
}
}
#############################################################################################
# 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.
# Overwrite by env SRS_HTTP_API_ENABLED
# default: off
enabled on;
# The http api listen endpoints, each is format as <[ip:]port>. The ip can be either ipv4
# or ipv6, or both. For example:
# listen 1985 [::]:1985 192.168.1.100:1985 10.10.10.100:1985;
# Overwrite by env SRS_HTTP_API_LISTEN
# Default: 1985
listen 8080;
# whether enable crossdomain request.
# Overwrite by env SRS_HTTP_API_CROSSDOMAIN
# default: on
crossdomain on;
# the HTTP RAW API is more powerful api to change srs state and reload.
raw_api {
# whether enable the HTTP RAW API.
# Overwrite by env SRS_HTTP_API_RAW_API_ENABLED
# default: off
enabled off;
# whether enable rpc reload.
# Overwrite by env SRS_HTTP_API_RAW_API_ALLOW_RELOAD
# default: off
allow_reload off;
# whether enable rpc query.
# Always off by https://github.com/ossrs/srs/issues/2653
#allow_query off;
# whether enable rpc update.
# Always off by https://github.com/ossrs/srs/issues/2653
#allow_update off;
}
# the auth is authentication for http api
auth {
# whether enable the HTTP AUTH.
# Overwrite by env SRS_HTTP_API_AUTH_ENABLED
# default: off
enabled on;
# The username of Basic authentication:
# Overwrite by env SRS_HTTP_API_AUTH_USERNAME
username admin;
# The password of Basic authentication:
# Overwrite by env SRS_HTTP_API_AUTH_PASSWORD
password admin;
}
# For https_api or HTTPS API.
# Note: The SRS HTTPS server is for demo only, please use Nginx/Caddy to proxy to SRS in production environment.
https {
# Whether enable HTTPS API.
# Overwrite by env SRS_HTTP_API_HTTPS_ENABLED
# default: off
enabled on;
# The listen endpoint for HTTPS API, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 1990 [::]:1990 192.168.1.100:1990 10.10.10.100:1990;
# Overwrite by env SRS_HTTP_API_HTTPS_LISTEN
# Default: 1990
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# Overwrite by env SRS_HTTP_API_HTTPS_KEY
# default: ./conf/server.key
key ./conf/server.key;
# The SSL public cert file, generated by:
# openssl req -new -x509 -key server.key -out server.crt -days 3650 -subj "/C=CA/ST=Toronto/L=Toronto/O=Me/OU=Me/CN=ossrs.io"
# Overwrite by env SRS_HTTP_API_HTTPS_CERT
# default: ./conf/server.crt
cert ./conf/server.crt;
}
}
# embedded 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 running 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 embedded 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.
# Overwrite by env SRS_HTTP_SERVER_ENABLED
# default: off
enabled on;
# The http streaming listen endpoints, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 8080 [::]:8080 192.168.1.100:8080 10.10.10.100:8080;
# Overwrite by env SRS_HTTP_SERVER_LISTEN
# default: 8080
listen 8080;
# the default dir for http root.
# Overwrite by env SRS_HTTP_SERVER_DIR
# default: ./objs/nginx/html
dir ./objs/nginx/html;
# whether enable crossdomain request.
# for both http static and stream server and apply on all vhosts.
# Overwrite by env SRS_HTTP_SERVER_CROSSDOMAIN
# default: on
crossdomain on;
# For https_server or HTTPS Streaming.
# Note: The SRS HTTPS server is for demo only, please use Nginx/Caddy to proxy to SRS in production environment.
https {
# Whether enable HTTPS Streaming.
# Overwrite by env SRS_HTTP_SERVER_HTTPS_ENABLED
# default: off
enabled on;
# The listen endpoint for HTTPS Streaming, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 8088 [::]:8088 192.168.1.100:8088 10.10.10.100:8088;
# Overwrite by env SRS_HTTP_SERVER_HTTPS_LISTEN
# default: 8088
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# Overwrite by env SRS_HTTP_SERVER_HTTPS_KEY
# default: ./conf/server.key
key ./conf/server.key;
# The SSL public cert file, generated by:
# openssl req -new -x509 -key server.key -out server.crt -days 3650 -subj "/C=CA/ST=Toronto/L=Toronto/O=Me/OU=Me/CN=ossrs.io"
# Overwrite by env SRS_HTTP_SERVER_HTTPS_CERT
# default: ./conf/server.crt
cert ./conf/server.crt;
}
}
# 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.
# Overwrite by env SRS_VHOST_HTTP_REMUX_ENABLED for all vhosts.
# 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.
# Overwrite by env SRS_VHOST_HTTP_REMUX_FAST_CACHE for all vhosts.
# default: 0
fast_cache 30;
# Whether drop packet if not match header. For example, there is has_audio and has video flag in FLV header, if
# this is set to on and has_audio is false, then SRS will drop audio packets when got audio packets. Generally
# it should work, but sometimes you might need SRS to keep packets even when FLV header is set to false.
# See https://github.com/ossrs/srs/issues/939#issuecomment-1348740526
# TODO: Only support HTTP-FLV stream right now.
# Overwrite by env SRS_VHOST_HTTP_REMUX_DROP_IF_NOT_MATCH for all vhosts.
# Default: on
drop_if_not_match on;
# Whether stream has audio track, used as default value for stream metadata, for example, FLV header contains
# this flag. Sometimes you might want to force the metadata by disable guess_has_av.
# For HTTP-FLV, use this as default value for FLV header audio flag. See https://github.com/ossrs/srs/issues/939#issuecomment-1351385460
# For HTTP-TS, use this as default value for PMT table. See https://github.com/ossrs/srs/issues/939#issuecomment-1365086204
# Overwrite by env SRS_VHOST_HTTP_REMUX_HAS_AUDIO for all vhosts.
# Default: on
has_audio on;
# Whether stream has video track, used as default value for stream metadata, for example, FLV header contains
# this flag. Sometimes you might want to force the metadata by disable guess_has_av.
# For HTTP-FLV, use this as default value for FLV header video flag. See https://github.com/ossrs/srs/issues/939#issuecomment-1351385460
# For HTTP-TS, use this as default value for PMT table. See https://github.com/ossrs/srs/issues/939#issuecomment-1365086204
# Overwrite by env SRS_VHOST_HTTP_REMUX_HAS_VIDEO for all vhosts.
# Default: on
has_video on;
# Whether guessing stream about audio or video track, used to generate the flags in, such as FLV header. If
# guessing, depends on sequence header and frames in gop cache, so it might be incorrect especially your stream
# is not regular. If not guessing, use the configured default value has_audio and has_video.
# For HTTP-FLV, enable guessing for av header flag, because FLV can't change the header. See https://github.com/ossrs/srs/issues/939#issuecomment-1351385460
# For HTTP-TS, ignore guessing because TS refresh the PMT when codec changed. See https://github.com/ossrs/srs/issues/939#issuecomment-1365086204
# Overwrite by env SRS_VHOST_HTTP_REMUX_GUESS_HAS_AV for all vhosts.
# Default: on
guess_has_av on;
# the stream mount 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.
# Overwrite by env SRS_VHOST_HTTP_REMUX_MOUNT for all vhosts.
# default: [vhost]/[app]/[stream].flv
mount [vhost]/[app]/[stream].flv;
}
}
# 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.
# Overwrite by env SRS_VHOST_HTTP_STATIC_ENABLED for all vhosts.
# 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.
# Overwrite by env SRS_VHOST_HTTP_STATIC_MOUNT for all vhosts.
# default: [vhost]/
mount [vhost]/hls;
# main dir of vhost,
# to delivery HTTP stream of this vhost.
# default: ./objs/nginx/html
# Overwrite by env SRS_VHOST_HTTP_STATIC_DIR for all vhosts.
dir ./objs/nginx/html/hls;
}
}
#############################################################################################
# SRT server section
#############################################################################################
# @doc https://github.com/ossrs/srs/issues/1147#usage
srt_server {
# whether SRT server is enabled.
# Overwrite by env SRS_SRT_SERVER_ENABLED
# default: off
enabled on;
# The UDP listen endpoints for SRT, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 10080 [::]:10080 192.168.1.100:10080 10.10.10.100:10080;
# Overwrite by env SRS_SRT_SERVER_LISTEN
listen 10080;
# For detail parameters, please read wiki:
# @see https://ossrs.io/lts/en-us/docs/v7/doc/srt#config
# The maxbw is the max bandwidth of the sender side.
# -1: Means the biggest bandwidth is infinity.
# 0: Means the bandwidth is determined by SRTO_INPUTBW.
# >0: Means the bandwidth is the configuration value.
# Overwrite by env SRS_SRT_SERVER_MAXBW
# default: -1
maxbw 1000000000;
# Maximum Segment Size. Used for buffer allocation and rate calculation using packet counter assuming fully
# filled packets. Each party can set its own MSS value independently. During a handshake the parties exchange
# MSS values, and the lowest is used.
# Overwrite by env SRS_SRT_SERVER_MSS
# default: 1500
mss 1500;
# The timeout time of the SRT connection on the sender side in ms. When SRT connects to a peer costs time
# more than this config, it will be close.
# Overwrite by env SRS_SRT_SERVER_CONNECT_TIMEOUT
# default: 3000
connect_timeout 4000;
# The timeout time of SRT connection on the receiver side in ms. When the SRT connection is idle
# more than this config, it will be close.
# Overwrite by env SRS_SRT_SERVER_PEER_IDLE_TIMEOUT
# default: 10000
peer_idle_timeout 8000;
# Default app for vmix, see https://github.com/ossrs/srs/pull/1615
# Overwrite by env SRS_SRT_SERVER_DEFAULT_APP
# default: live
default_app live;
# Default streamid when client doesn't provide one.
# This is used when SRT client connects without setting SRTO_STREAMID socket option.
# The streamid format follows SRT standard: #!::r=app/stream,m=publish|request
# Overwrite by env SRS_SRT_SERVER_DEFAULT_STREAMID
# default: #!::r=live/livestream,m=request
default_streamid "#!::r=live/livestream,m=request";
# The peerlatency is set by the sender side and will notify the receiver side.
# Overwrite by env SRS_SRT_SERVER_PEERLATENCY
# default: 0
peerlatency 0;
# The recvlatency means latency from sender to receiver.
# Overwrite by env SRS_SRT_SERVER_RECVLATENCY
# default: 120
recvlatency 0;
# This latency configuration configures both recvlatency and peerlatency to the same value.
# Overwrite by env SRS_SRT_SERVER_LATENCY
# default: 120
latency 0;
# The tsbpd mode means timestamp based packet delivery.
# SRT sender side will pack timestamp in each packet. If this config is true,
# the receiver will read the packet according to the timestamp in the head of the packet.
# Overwrite by env SRS_SRT_SERVER_TSBPDMODE
# default: on
tsbpdmode off;
# The tlpkdrop means too-late Packet Drop
# SRT sender side will pack timestamp in each packet, When the network is congested,
# the packet will drop if latency is bigger than the configuration in both sender side and receiver side.
# And on the sender side, it also will be dropped because latency is bigger than configuration.
# Overwrite by env SRS_SRT_SERVER_TLPKTDROP
# default: on
tlpktdrop off;
# The send buffer size of SRT.
# Overwrite by env SRS_SRT_SERVER_SENDBUF
# default: 8192 * (1500-28)
sendbuf 2000000;
# The recv buffer size of SRT.
# Overwrite by env SRS_SRT_SERVER_RECVBUF
# default: 8192 * (1500-28)
recvbuf 2000000;
# The passphrase of SRT.
# If passphrase is no empty, all the srt client must be using the correct passphrase to publish or play,
# or the srt connection will reject. The length of passphrase must be in range 10~79.
# @see https://github.com/Haivision/srt/blob/master/docs/API/API-socket-options.md#srto_passphrase.
# Overwrite by env SRS_SRT_SERVER_PASSPHRASE
# default: ""
passphrase xxxxxxxxxxxx;
# The pbkeylen of SRT.
# The pbkeylen determined the AES encrypt algorithm, this option only allow 4 values which is 0, 16, 24, 32
# @see https://github.com/Haivision/srt/blob/master/docs/API/API-socket-options.md#srto_pbkeylen.
# Overwrite by env SRS_SRT_SERVER_PBKEYLEN
# default: 0
pbkeylen 16;
}
vhost srt.vhost.srs.com {
srt {
# Whether enable SRT on this vhost.
# Overwrite by env SRS_VHOST_SRT_ENABLED for all vhosts.
# Default: off
enabled on;
# Whether covert SRT to RTMP stream.
# Overwrite by env SRS_VHOST_SRT_TO_RTMP for all vhosts.
# Default: off
srt_to_rtmp on;
}
}
#############################################################################################
# WebRTC server section
#############################################################################################
rtc_server {
# Whether enable WebRTC server.
# Overwrite by env SRS_RTC_SERVER_ENABLED
# default: off
enabled on;
# The udp listen endpoints, we will reuse it for connections, each with format as <[ip:]port>.
# The ip can be either ipv4 or ipv6, or both. For example:
# listen 8000 [::]:8000 192.168.1.100:8000 10.10.10.100:8000;
# Overwrite by env SRS_RTC_SERVER_LISTEN
# default: 8000
listen 8000;
# For WebRTC over TCP directly, not TURN, see https://github.com/ossrs/srs/issues/2852
# Some network does not support UDP, or not very well, so we use TCP like HTTP/80 port for firewall traversing.
tcp {
# Whether enable WebRTC over TCP.
# Overwrite by env SRS_RTC_SERVER_TCP_ENABLED
# Default: off
enabled off;
# The TCP listen endpoints for WebRTC, each with format as <[ip:]port>. The ip can be either ipv4 or ipv6,
# or both. For example:
# listen 8000 [::]:8000 192.168.1.100:8000 10.10.10.100:8000;
# Highly recommend is some normally used ports, such as TCP/80, TCP/443, TCP/8000, TCP/8080 etc.
# However SRS default to TCP/8000 corresponding to UDP/8000.
# Overwrite by env SRS_RTC_SERVER_TCP_LISTEN
# Default: 8000
listen 8000;
}
# The protocol for candidate to use, it can be:
# udp Generate UDP candidates. Note that UDP server is always enabled for WebRTC.
# tcp Generate TCP candidates. Fail if rtc_server.tcp(WebRTC over TCP) is disabled.
# all Generate UDP+TCP candidates. Ignore if rtc_server.tcp(WebRTC over TCP) is disabled.
# Note that if both are connected, we will use the first connected(DTLS done) one.
# Overwrite by env SRS_RTC_SERVER_PROTOCOL
# Default: udp
protocol udp;
# The exposed candidate IPs, response in SDP candidate line. It can be:
# * Retrieve server IP automatically, from all network interfaces.
# $CANDIDATE Read the IP from ENV variable, use * if not set.
# x.x.x.x A specified IP address or DNS name, use * if 0.0.0.0.
# You can also set the candidate by the query string eip, note that you can also set the UDP port,
# for example:
# http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11
# http://locahost:1985/rtc/v1/whip/?app=live&stream=livestream&eip=192.168.3.11:18000
# @remark For Firefox, the candidate MUST be IP, MUST NOT be DNS name, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#config-candidate
# Overwrite by env SRS_RTC_SERVER_CANDIDATE
# default: *
candidate *;
# If candidate is * or 0.0.0.0, means SRS could detect IP automatically, filtered by ip_family.
# You can config this to off to disable the detecting, then SRS will try to parse the API hostname.
# Note that browser might fail if no CANDIDATE specified.
# Overwrite by env SRS_RTC_SERVER_USE_AUTO_DETECT_NETWORK_IP
# Default: on
use_auto_detect_network_ip on;
# The IP family filter for auto discover candidate, it can be:
# ipv4 Filter IP v4 candidates.
# ipv6 Filter IP v6 candidates.
# all Filter all IP v4 or v6 candidates.
# For example, if set to ipv4, we only use the IPv4 address as candidate.
# Overwrite by env SRS_RTC_SERVER_IP_FAMILY
# default: ipv4
ip_family ipv4;
# If api_as_candidates is on, SRS would try to use the IP of api server, specified by srs.sdk.js request:
# api:string "http://r.ossrs.net:1985/rtc/v1/play/"
# in this case, the r.ossrs.net and 39.107.238.185 will be added as candidates.
# Overwrite by env SRS_RTC_SERVER_API_AS_CANDIDATES
# Default: on
api_as_candidates on;
# If use api as CANDIDATE, whether resolve the api hostname.
# Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
# Note that if hostname is IPv4 address, always directly use it.
# Overwrite by env SRS_RTC_SERVER_RESOLVE_API_DOMAIN
# Default: on
resolve_api_domain on;
# If use api as CANDIDATE, whether keep original api domain name as CANDIDATE.
# Note that use original domain name as CANDIDATE, which might make Firefox failed, see https://bugzilla.mozilla.org/show_bug.cgi?id=1239006
# Overwrite by env SRS_RTC_SERVER_KEEP_API_DOMAIN
# Default: off
keep_api_domain off;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# Overwrite by env SRS_RTC_SERVER_ECDSA
# default: on
ecdsa on;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# Overwrite by env SRS_RTC_SERVER_ENCRYPT
# default: on
encrypt on;
# We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
# Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
# and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
# Overwrite by env SRS_RTC_SERVER_REUSEPORT
# default: 1
reuseport 1;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# Overwrite by env SRS_RTC_SERVER_MERGE_NALUS
# default: off
merge_nalus off;
# The black-hole to copy packet to, for debugging.
# For example, when debugging Chrome publish stream, the received packets are encrypted cipher,
# we can set the publisher black-hole, SRS will copy the plaintext packets to black-hole, and
# we are able to capture the plaintext packets by wireshark.
black_hole {
# Whether enable the black-hole.
# Overwrite by env SRS_RTC_SERVER_BLACK_HOLE_ENABLED
# default: off
enabled off;
# The black-hole address for session.
# Overwrite by env SRS_RTC_SERVER_BLACK_HOLE_ADDR
addr 127.0.0.1:10000;
}
}
vhost rtc.vhost.srs.com {
rtc {
# Whether enable WebRTC server.
# Overwrite by env SRS_VHOST_RTC_ENABLED for all vhosts.
# default: off
enabled on;
# Whether support NACK.
# Overwrite by env SRS_VHOST_RTC_NACK for all vhosts.
# default: on
nack on;
# Whether directly use the packet, avoid copy.
# Overwrite by env SRS_VHOST_RTC_NACK_NO_COPY for all vhosts.
# default: on
nack_no_copy on;
# Whether support TWCC.
# Overwrite by env SRS_VHOST_RTC_TWCC for all vhosts.
# default: on
twcc on;
# The timeout in seconds for session timeout.
# Client will send ping(STUN binding request) to server, we use it as heartbeat.
# Overwrite by env SRS_VHOST_RTC_STUN_TIMEOUT for all vhosts.
# default: 30
stun_timeout 30;
# The strict check when process stun.
# Overwrite by env SRS_VHOST_RTC_STUN_STRICT_CHECK for all vhosts.
# default: off
stun_strict_check on;
# The role of dtls when peer is actpass: passive or active
# Overwrite by env SRS_VHOST_RTC_DTLS_ROLE for all vhosts.
# default: passive
dtls_role passive;
# The version of dtls, support dtls1.0, dtls1.2, and auto
# Overwrite by env SRS_VHOST_RTC_DTLS_VERSION for all vhosts.
# default: auto
dtls_version auto;
# Drop the packet with the pt(payload type), 0 never drop.
# Overwrite by env SRS_VHOST_RTC_DROP_FOR_PT for all vhosts.
# default: 0
drop_for_pt 0;
###############################################################
# Whether enable transmuxing RTMP to RTC.
# If enabled, transcode aac to opus.
# Overwrite by env SRS_VHOST_RTC_RTMP_TO_RTC for all vhosts.
# default: off
rtmp_to_rtc off;
# Whether keep B-frame, which is normal feature in live streaming,
# but usually disabled in RTC.
# Overwrite by env SRS_VHOST_RTC_KEEP_BFRAME for all vhosts.
# default: off
keep_bframe off;
# Whether to keep the h.264 SEI type NALU packet.
# DJI drone M30T will send many SEI type NALU packet, while iOS hardware decoder (Video Toolbox)
# dislike to feed it so many SEI NALU between NonIDR and IDR NALU packets.
# @see https://github.com/ossrs/srs/issues/4052
# Overwrite by env SRS_VHOST_RTC_KEEP_AVC_NALU_SEI for all vhosts.
# Default: on
keep_avc_nalu_sei on;
# The transcode audio bitrate, for RTMP to RTC.
# Overwrite by env SRS_VHOST_RTC_OPUS_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
opus_bitrate 48000;
###############################################################
# Whether enable transmuxing RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_RTC_TO_RTMP for all vhosts.
# Default: off
rtc_to_rtmp off;
# The PLI interval in seconds, for RTC to RTMP.
# Note the available range is [0.5, 30]
# Overwrite by env SRS_VHOST_RTC_PLI_FOR_RTMP for all vhosts.
# Default: 6.0
pli_for_rtmp 6.0;
# The transcode audio bitrate, for RTC to RTMP.
# Overwrite by env SRS_VHOST_RTC_AAC_BITRATE for all vhosts.
# [8000, 320000]
# default: 48000
aac_bitrate 48000;
###############################################################
# Whether initialize RTP rate from SDP sample rate for immediate A/V sync.
# When enabled, the RTP rate (units per millisecond) is initialized from the SDP
# sample rate (e.g., 90 for video 90kHz, 48 for audio 48kHz) before receiving
# 2 RTCP SR packets. This allows immediate audio/video synchronization.
# The rate will be updated to a more precise value after receiving the 2nd SR.
# Overwrite by env SRS_VHOST_RTC_INIT_RATE_FROM_SDP for all vhosts.
# Default: off
init_rate_from_sdp off;
# Whether keep original SSRC and timestamp when forwarding RTC packets.
# When enabled, SRS will preserve the original SSRC and RTP timestamps from
# publisher to player, which is helpful for end-to-end debugging and troubleshooting
# (e.g., tracking packet loss or delays using Wireshark).
# When disabled (default), SRS generates new SSRC for each player and rebuilds
# timestamps to ensure continuous playback during stream restarts and support
# play-before-publish scenarios.
# @see https://github.com/ossrs/srs/issues/3850
# Overwrite by env SRS_VHOST_RTC_KEEP_ORIGINAL_SSRC for all vhosts.
# Default: off
keep_original_ssrc off;
}
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.
# Whether enable min delay mode for vhost.
# Overwrite by env SRS_VHOST_MIN_LATENCY for all vhosts.
# default: on, for RTC.
min_latency on;
play {
# set the MW(merged-write) latency in ms.
# @remark For WebRTC, we enable pass-by-timestamp mode, so we ignore this config.
# Overwrite by env SRS_VHOST_PLAY_MW_LATENCY for all vhosts.
# default: 0 (For WebRTC)
mw_latency 0;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, that is min_latency on)
# default: 1 (For WebRTC, that is min_latency off)
# Overwrite by env SRS_VHOST_PLAY_MW_MSGS for all vhosts.
mw_msgs 0;
}
}
#############################################################################################
# Circuit breaker sections
#############################################################################################
# For system circuit breaker.
circuit_breaker {
# Whether enable the circuit breaker.
# Overwrite by env SRS_CIRCUIT_BREAKER_ENABLED
# Default: on
enabled on;
# The CPU percent(0, 100) ever 1s, as system high water-level, which enable the circuit-break
# mechanism, for example, NACK will be disabled if high water-level.
# Overwrite by env SRS_CIRCUIT_BREAKER_HIGH_THRESHOLD
# Default: 90
high_threshold 90;
# Reset the high water-level, if number of pulse under high_threshold.
# @remark 0 to disable the high water-level.
# Overwrite by env SRS_CIRCUIT_BREAKER_HIGH_PULSE
# Default: 2
high_pulse 2;
# The CPU percent(0, 100) ever 1s, as system critical water-level, which enable the circuit-break
# mechanism, for example, TWCC will be disabled if high water-level.
# @note All circuit-break mechanism of high-water-level scope are enabled in critical.
# Overwrite by env SRS_CIRCUIT_BREAKER_CRITICAL_THRESHOLD
# Default: 95
critical_threshold 95;
# Reset the critical water-level, if number of pulse under critical_threshold.
# @remark 0 to disable the critical water-level.
# Overwrite by env SRS_CIRCUIT_BREAKER_CRITICAL_PULSE
# Default: 1
critical_pulse 1;
# If dying, also drop packets for players.
# Overwrite by env SRS_CIRCUIT_BREAKER_DYING_THRESHOLD
# Default: 99
dying_threshold 99;
# If CPU exceed the dying_pulse times, enter dying.
# @remark 0 to disable the dying water-level.
# Overwrite by env SRS_CIRCUIT_BREAKER_DYING_PULSE
# Default: 5
dying_pulse 5;
}
#############################################################################################
# Proetheus exporter sections
#############################################################################################
# Prometheus exporter config.
# See https://prometheus.io/docs/instrumenting/exporters
exporter {
# Whether exporter is enabled.
# Overwrite by env SRS_EXPORTER_ENABLED
# Default: off
enabled off;
# The http api listen port for exporter metrics.
# Overwrite by env SRS_EXPORTER_LISTEN
# Default: 9972
# See https://github.com/prometheus/prometheus/wiki/Default-port-allocations
listen 9972;
# The logging label to category the cluster servers.
# Overwrite by env SRS_EXPORTER_LABEL
label cn-beijing;
# The logging tag to category the cluster servers.
# Overwrite by env SRS_EXPORTER_TAG
tag cn-edge;
}
#############################################################################################
# Security and referer sections
#############################################################################################
# the vhost for anti-suck.
vhost refer.anti_suck.com {
# refer hotlink-denial.
refer {
# whether enable the refer hotlink-denial.
# default: off.
enabled on;
# 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.
all github.com github.io;
# refer for publish clients specified.
# the common refer is not overridden by this.
# if not specified this field, allow all.
# default: not specified.
publish github.com github.io;
# refer for play clients specified.
# the common refer is not overridden by this.
# if not specified this field, allow all.
# default: not specified.
play github.com github.io;
}
}
# the security to allow or deny clients.
vhost security.srs.com {
# security for host to allow or deny clients.
security {
# whether enable the security for vhost.
# default: off
enabled on;
# the security list, each item format as:
# allow|deny publish|play all|<ip or cidr>
# for example:
# allow publish all;
# deny publish all;
# allow publish 127.0.0.1;
# deny publish 127.0.0.1;
# allow publish 10.0.0.0/8;
# deny publish 10.0.0.0/8;
# allow play all;
# deny play all;
# allow play 127.0.0.1;
# deny play 127.0.0.1;
# allow play 10.0.0.0/8;
# deny play 10.0.0.0/8;
# 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;
}
}
#############################################################################################
# HTTP callback sections
#############################################################################################
# the http hook callback vhost, srs will invoke the hooks for specified events.
vhost hooks.callback.srs.com {
http_hooks {
# whether the http hooks enable.
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ENABLED for all vhosts.
# default off.
enabled on;
# 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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy", "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3"
# }
# if valid, the hook must return HTTP code 200(Status 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
# @remark For SRS4, the HTTPS url is supported, for example:
# on_publish https://xxx/api0 https://xxx/api1 https://xxx/apiN
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_PUBLISH for all vhosts.
# @remark When using environment variables, use space-separated URLs with proper quoting:
# SRS_VHOST_HTTP_HOOKS_ON_PUBLISH="https://xxx/api0 https://xxx/api1"
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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy", "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3"
# }
# if valid, the hook must return HTTP code 200(Status 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
# @remark For SRS4, the HTTPS url is supported, for example:
# on_unpublish https://xxx/api0 https://xxx/api1 https://xxx/apiN
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_UNPUBLISH for all vhosts.
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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy",
# "pageUrl": "http://www.test.com/live.html", "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3",
# "clients": 3
# }
# Note: The "clients" field indicates the current number of clients playing this stream
# (including the client that just started playing). This is useful for on-demand streaming
# scenarios where you want to start a publisher when the first client connects (clients=1).
# if valid, the hook must return HTTP code 200(Status 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
# @remark For SRS4, the HTTPS url is supported, for example:
# on_play https://xxx/api0 https://xxx/api1 https://xxx/apiN
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_PLAY for all vhosts.
# @remark When using environment variables, use space-separated URLs with proper quoting:
# SRS_VHOST_HTTP_HOOKS_ON_PLAY="https://xxx/api0 https://xxx/api1"
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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy", "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3",
# "clients": 2
# }
# Note: The "clients" field indicates the current number of clients still playing this stream
# (after the client has stopped). This is useful for on-demand streaming scenarios where you
# want to stop a publisher when the last client disconnects (clients=0).
# if valid, the hook must return HTTP code 200(Status 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
# @remark For SRS4, the HTTPS url is supported, for example:
# on_stop https://xxx/api0 https://xxx/api1 https://xxx/apiN
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_STOP for all vhosts.
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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy",
# "cwd": "/usr/local/srs",
# "file": "./objs/nginx/html/live/livestream.1420254068776.flv", "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3"
# }
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_DVR for all vhosts.
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": "9308h583",
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy",
# "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, "server_id": "vid-werty",
# "stream_url": "video.test.com/live/livestream", "stream_id": "vid-124q9y3"
# }
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_HLS for all vhosts.
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:
# [server_id], replace with the server_id
# [app], replace with the app.
# [stream], replace with the stream.
# [param], replace with the param.
# [ts_url], replace with the ts url.
# ignore any return data of server.
# @remark random select a url to report, not report all.
# Overwrite by env SRS_VHOST_HTTP_HOOKS_ON_HLS_NOTIFY for all vhosts.
on_hls_notify http://127.0.0.1:8085/api/v1/hls/[server_id]/[app]/[stream]/[ts_url][param];
}
}
#############################################################################################
# DASH streaming sections
#############################################################################################
# The vhost for MPEG-DASH.
vhost dash.srs.com {
dash {
# Whether DASH is enabled.
# Transmux RTMP to DASH if on.
# Overwrite by env SRS_VHOST_DASH_ENABLED for all vhosts.
# Default: off
enabled on;
# The duration of segment in seconds.
# Overwrite by env SRS_VHOST_DASH_DASH_FRAGMENT for all vhosts.
# Default: 30
dash_fragment 30;
# The period to update the MPD in seconds.
# Overwrite by env SRS_VHOST_DASH_DASH_UPDATE_PERIOD for all vhosts.
# Default: 150
dash_update_period 150;
# The depth of timeshift buffer in seconds.
# Overwrite by env SRS_VHOST_DASH_DASH_TIMESHIFT for all vhosts.
# Default: 300
dash_timeshift 300;
# The base/home dir/path for dash.
# All init and segment files will write under this dir.
# Overwrite by env SRS_VHOST_DASH_DASH_PATH for all vhosts.
dash_path ./objs/nginx/html;
# The DASH MPD file 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.
# Overwrite by env SRS_VHOST_DASH_DASH_MPD_FILE for all vhosts.
# Default: [app]/[stream].mpd
dash_mpd_file [app]/[stream].mpd;
# The dash windows size in seconds.
# Overwrite by env SRS_VHOST_DASH_DASH_WINDOW_SIZE for all vhosts.
# Default: 5
dash_window_size 5;
# whether cleanup the old expired dash files.
# Overwrite by env SRS_VHOST_DASH_DASH_CLEANUP for all vhosts.
# default: on
dash_cleanup on;
# If there is no incoming packets, dispose DASH in this timeout in seconds,
# which removes all DASH files including m3u8 and ts files.
# @remark 0 to disable dispose for publisher.
# @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose DASH.
# Overwrite by env SRS_VHOST_DASH_DASH_DISPOSE for all vhosts.
# default: 120
dash_dispose 120;
}
}
#############################################################################################
# HLS streaming sections
#############################################################################################
# the vhost with hls specified.
vhost hls.srs.com {
hls {
# whether the hls is enabled.
# if off, do not write hls(ts and m3u8) when publish.
# Overwrite by env SRS_VHOST_HLS_ENABLED for all vhosts.
# default: off
enabled on;
# whether to use fmp4 as container
# The default value is off, then HLS use ts as container format,
# if on, HLS use fmp4 as container format.
# Overwrite by env SRS_VHOST_HLS_HLS_USE_FMP4 for all vhosts.
# default: off
hls_use_fmp4 on;
# the hls fragment in seconds, the duration of a piece of ts.
# Overwrite by env SRS_VHOST_HLS_HLS_FRAGMENT for all vhosts.
# 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
# Overwrite by env SRS_VHOST_HLS_HLS_TD_RATIO for all vhosts.
# default: 1.0
hls_td_ratio 1.0;
# the audio overflow ratio.
# for pure audio, the duration to reap the segment.
# for example, the hls_fragment is 10s, hls_aof_ratio is 1.2,
# the segment will reap to 12s for pure audio.
# Overwrite by env SRS_VHOST_HLS_HLS_AOF_RATIO for all vhosts.
# default: 2.1
hls_aof_ratio 2.1;
# the hls window in seconds, the number of ts in m3u8.
# Overwrite by env SRS_VHOST_HLS_HLS_WINDOW for all vhosts.
# default: 60
hls_window 60;
# Whether to enable HLS recover mode. When enabled, HLS sequence numbers
# continue from where they stopped last time instead of restarting from 0.
# This preserves old fragments and provides better player compatibility.
# Useful for stream interruption recovery and live event continuity.
# Overwrite by env SRS_VHOST_HLS_HLS_RECOVER for all vhosts.
# Default: on
hls_recover on;
# the error strategy. can be:
# ignore, disable the hls.
# disconnect, require encoder republish.
# continue, ignore failed try to continue output hls.
# Overwrite by env SRS_VHOST_HLS_HLS_ON_ERROR for all vhosts.
# default: continue
hls_on_error continue;
# the hls output path.
# the m3u8 file is configured by hls_path/hls_m3u8_file, the default is:
# ./objs/nginx/html/[app]/[stream].m3u8
# the ts file is configured 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.
# Overwrite by env SRS_VHOST_HLS_HLS_PATH for all vhosts.
# 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.
# Overwrite by env SRS_VHOST_HLS_HLS_M3U8_FILE for all vhosts.
# 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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# [seq], the sequence number of ts.
# [duration], replace this const to current ts duration.
# @see https://ossrs.io/lts/en-us/docs/v7/doc/dvr#custom-path
# @see https://ossrs.io/lts/en-us/docs/v7/doc/hls#config
# Overwrite by env SRS_VHOST_HLS_HLS_TS_FILE for all vhosts.
# default: [app]/[stream]-[seq].ts
hls_ts_file [app]/[stream]-[seq].ts;
# the hls fmp4 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], replace this const to current minute.
# [05], replace this const to current second.p
# [999], replace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# [seq], the sequence number of fmp4.
# [duration], replace this const to current ts duration.
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-hls#hls-config
# Overwrite by env SRS_VHOST_HLS_HLS_FMP4_FILE for all vhosts.
# default: [app]/[stream]-[seq].m4s
hls_fmp4_file [app]/[stream]-[seq].m4s;
# the hls init mp4 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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/delivery-hls#hls-config
# Overwrite by env SRS_VHOST_HLS_HLS_INIT_FILE for all vhosts.
# default: [app]/[stream]/init.mp4
hls_init_file [app]/[stream]/init.mp4;
# the hls entry prefix, which is base url of ts url.
# for example, the prefix is:
# http://your-server/
# then, the ts path in m3u8 will be like:
# http://your-server/live/livestream-0.ts
# http://your-server/live/livestream-1.ts
# ...
# Overwrite by env SRS_VHOST_HLS_HLS_ENTRY_PREFIX for all vhosts.
# optional, default to empty string.
hls_entry_prefix http://your-server;
# whether cleanup the old expired ts files.
# Overwrite by env SRS_VHOST_HLS_HLS_CLEANUP for all vhosts.
# default: on
hls_cleanup on;
# If there is no incoming packets, dispose HLS in this timeout in seconds,
# which removes all HLS files including m3u8 and ts files.
# @remark 0 to disable dispose for publisher.
# @remark apply for publisher timeout only, while "etc/init.d/srs stop" always dispose hls.
# Overwrite by env SRS_VHOST_HLS_HLS_DISPOSE for all vhosts.
# default: 120
hls_dispose 120;
# 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.
# Overwrite by env SRS_VHOST_HLS_HLS_WAIT_KEYFRAME for all vhosts.
# default: on
hls_wait_keyframe on;
# whether use floor for the hls_ts_file path generation.
# if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
# and use enhanced algorithm to calc deviation for segment.
# @remark when floor on, recommend the hls_segment>=2*gop.
# Overwrite by env SRS_VHOST_HLS_HLS_TS_FLOOR for all vhosts.
# default: off
hls_ts_floor off;
# 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.
# Overwrite by env SRS_VHOST_HLS_HLS_NB_NOTIFY for all vhosts.
# default: 64
hls_nb_notify 64;
# Whether enable hls_ctx for HLS streaming, for which we create a "fake" connection for HTTP API and callback.
# For each HLS streaming session, we use a child m3u8 with a session identified by query "hls_ctx", it simply
# work as the session id.
# Once the HLS streaming session is created, we will cleanup it when timeout in 2*hls_window seconds. So it
# takes a long time period to identify the timeout.
# Now we got a HLS stremaing session, just like RTMP/WebRTC/HTTP-FLV streaming, we're able to stat the session
# as a "fake" connection, do HTTP callback when start playing the HLS streaming. You're able to do querying and
# authentication.
# Note that it will make NGINX edge cache always missed, so never enable HLS streaming if use NGINX edges.
# Overwrite by env SRS_VHOST_HLS_HLS_CTX for all vhosts.
# Default: on
hls_ctx on;
# For HLS pseudo streaming, whether enable the session for each TS segment.
# If enabled, SRS HTTP API will show the statistics about HLS streaming bandwidth, both m3u8 and ts file. Please
# note that it also consumes resource, because each ts file should be served by SRS, all NGINX cache will be
# missed because we add session id to each ts file.
# Note that it will make NGINX edge cache always missed, so never enable HLS streaming if use NGINX edges.
# Overwrite by env SRS_VHOST_HLS_HLS_TS_CTX for all vhosts.
# Default: on
hls_ts_ctx on;
# Whether use relative path for media playlist URL in HLS master playlist.
# When on, the master playlist uses relative path like "livestream.m3u8?hls_ctx=xxx".
# When off, the master playlist uses absolute path like "/live/livestream.m3u8?hls_ctx=xxx".
# Relative path is useful for reverse proxy scenarios with path rewriting.
# For example, if external URL is "http://proxy/srs/live/stream.m3u8" and it's rewritten to
# "http://origin/live/stream.m3u8", relative path ensures the player can correctly resolve
# the media playlist URL.
# Overwrite by env SRS_VHOST_HLS_HLS_MASTER_M3U8_PATH_RELATIVE for all vhosts.
# Default: off
hls_master_m3u8_path_relative off;
# whether using AES encryption.
# Overwrite by env SRS_VHOST_HLS_HLS_KEYS for all vhosts.
# default: off
hls_keys on;
# the number of clear ts which one key can encrypt.
# Overwrite by env SRS_VHOST_HLS_HLS_FRAGMENTS_PER_KEY for all vhosts.
# default: 5
hls_fragments_per_key 5;
# the hls key 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.
# [seq], the sequence number of key corresponding to the ts.
# Overwrite by env SRS_VHOST_HLS_HLS_KEY_FILE for all vhosts.
hls_key_file [app]/[stream]-[seq].key;
# the key output path.
# the key file is configed by hls_path/hls_key_file, the default is:
# ./objs/nginx/html/[app]/[stream]-[seq].key
# Overwrite by env SRS_VHOST_HLS_HLS_KEY_FILE_PATH for all vhosts.
hls_key_file_path ./objs/nginx/html;
# the key root URL, use this can support https.
# @remark It's optional.
# @remark Supports query string for authentication tokens (e.g., JWT).
# Example: http://localhost:8080/?token=abc123&sig=xyz789
# Result in m3u8: http://localhost:8080/live/livestream-0.key?token=abc123&sig=xyz789
# Overwrite by env SRS_VHOST_HLS_HLS_KEY_URL for all vhosts.
hls_key_url http://localhost:8080/;
# Special control controls.
###########################################
# Whether calculate the DTS of audio frame directly.
# If on, guess the specific DTS by AAC samples, please read https://github.com/ossrs/srs/issues/547#issuecomment-294350544
# If off, directly turn the FLV timestamp to DTS, which might cause corrupt audio stream.
# @remark Recommend to set to off, unless your audio stream sample-rate and timestamp is not correct.
# Overwrite by env SRS_VHOST_HLS_HLS_DTS_DIRECTLY for all vhosts.
# Default: on
hls_dts_directly 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
# @see https://ossrs.io/lts/en-us/docs/v7/doc/hls#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.
# [param], replace with the param.
# [ts_url], replace with the ts url.
# for the hls http callback, @see http_hooks.on_hls_notify of vhost hooks.callback.srs.com
# @see https://ossrs.io/lts/en-us/docs/v7/doc/hls#on-hls-notify
}
}
#############################################################################################
# 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__ {
# whether the vhost is enabled.
# if off, all request access denied.
# default: on
enabled off;
}
#
#
#
#
#
#
#********************************************************************************************
#********************************************************************************************
# CAUTION: The features below are experimental and new, so use it cautiously.
#********************************************************************************************
#********************************************************************************************
#############################################################################################
# RTSP server sections
#############################################################################################
rtsp_server {
# Whether enable RTSP server.
# Overwrite by env SRS_RTSP_SERVER_ENABLED
# default: off
enabled on;
# The listen endpoints for RTSP server, each with format as <[ip:]port>. The ip can be
# either ipv4 or ipv6, or both. For example:
# listen 554 [::]:554 192.168.1.100:554 10.10.10.100:554;
# Overwrite by env SRS_RTSP_SERVER_LISTEN
# default: 554
listen 8554;
}
vhost rtsp.vhost.srs.com {
rtsp {
# Whether enable RTSP server.
# Overwrite by env SRS_VHOST_RTSP_ENABLED for all vhosts.
# default: off
enabled on;
# Whether transmux RTMP to RTSP.
# Overwrite by env SRS_VHOST_RTSP_RTMP_TO_RTSP for all vhosts.
# default: on
rtmp_to_rtsp on;
}
}
#############################################################################################
# In the config file, the include file can be anywhere in order to enhance the readability
# of the config file and enable the reuse of part of the config file.
# When using include files, make sure that the included files themselves have the correct SRS syntax,
# that is, config directives and blocks, and then specify the paths to these files.
#
# @see https://github.com/ossrs/srs/issues/1399
#############################################################################################
include ./conf/include.vhost.conf;
#############################################################################################
# Stream converter sections
#############################################################################################
# The stream converter coverts stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
# Push MPEGTS over UDP to SRS.
stream_caster {
# Whether stream converter is enabled.
# Default: off
enabled on;
# The type of stream converter, could be:
# mpegts_over_udp, push MPEG-TS over UDP and convert to RTMP.
caster mpegts_over_udp;
# The output rtmp url.
# For mpegts_over_udp converter, the typically output url:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# The listen port for stream converter.
# For mpegts_over_udp converter, listen at udp port. for example, 8935.
listen 8935;
}
# Push FLV by HTTP POST to SRS.
stream_caster {
# Whether stream converter is enabled.
# Default: off
enabled on;
# The type of stream converter, could be:
# flv, push FLV by HTTP POST and convert to RTMP.
caster flv;
# The output rtmp url.
# For flv converter, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# For example, POST to url:
# http://127.0.0.1:8936/live/livestream.flv
# Where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/[app]/[stream];
# The listen port for stream converter.
# For flv converter, listen at tcp port. for example, 8936.
listen 8936;
}
# For GB28181 server, see https://github.com/ossrs/srs/issues/3176
# For SIP specification, see https://www.ietf.org/rfc/rfc3261.html
# For GB28181 2016 spec, see https://openstd.samr.gov.cn/bzgk/gb/newGbInfo?hcno=469659DC56B9B8187671FF08748CEC89
stream_caster {
# Whether stream converter is enabled.
# Default: off
enabled off;
# The type of stream converter, could be:
# gb28181, Push GB28181 stream and convert to RTMP.
caster gb28181;
# The output rtmp url.
# For gb28181 converter, the typically output url:
# rtmp://127.0.0.1/live/[stream]
# The available variables:
# [stream] The video channel codec ID.
output rtmp://127.0.0.1/live/[stream];
# The listen TCP port for stream converter.
# For gb28181 converter, listen at TCP port. for example, 9000.
# @remark We always enable bundle for media streams at this port.
listen 9000;
# SIP server for GB28181. The embedded SIP server is disabled by default, please use external SIP server
# such as [jsip](https://github.com/usnistgov/jsip) and there is a demo [srs-sip](https://github.com/ossrs/srs-sip)
# also base on it, for more information please see project [GB: External SIP](https://ossrs.io/lts/zh-cn/docs/v7/doc/gb28181#external-sip).
}
#
#
#
#
#
#
#********************************************************************************************
#********************************************************************************************
# CAUTION: The features below are not recommanded, so you should use other replacement tools instead.
#********************************************************************************************
#********************************************************************************************
#############################################################################################
# Forward sections. You should use FFmpeg to forward stream.
#############################################################################################
# the vhost which forward publish streams.
vhost same.vhost.forward.srs.com {
# forward stream to other servers.
forward {
# whether enable the forward.
# default: off
enabled on;
# 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}
destination 127.0.0.1:1936 127.0.0.1:1937;
# when client(encoder) publish to vhost/app/stream, call the hook in creating backend forwarder.
# the request in the POST data string is a object encode by json:
# {
# "action": "on_forward",
# "server_id": "vid-k21d7y2",
# "client_id": "9o7g1330",
# "ip": "127.0.0.1",
# "vhost": "__defaultVhost__",
# "app": "live",
# "tcUrl": "rtmp://127.0.0.1:1935/live",
# "stream": "livestream",
# "param": ""
# }
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# {
# "code": 0,
# "data": {
# "urls":[
# "rtmp://127.0.0.1:19350/test/teststream"
# ]
# }
# }
# If backend response empty urls, the forwarding is also disabled.
# Only one backend is supported, and the format should be:
# backend http://hostport/api
backend http://127.0.0.1:8085/api/v1/forward;
}
}
#############################################################################################
# Exec command sections You should use http_hooks and FFmpeg to exec command.
#############################################################################################
# the vhost for exec, fork process when publish stream.
vhost exec.srs.com {
# the exec used to fork process when got some event.
exec {
# whether enable the exec.
# default: off.
enabled off;
# when publish stream, exec the process with variables:
# [vhost] the input stream vhost.
# [port] the input stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the transcode engine name.
# other variables for exec only:
# [url] the rtmp url which trigger the publish.
# [tcUrl] the client request tcUrl.
# [swfUrl] the client request swfUrl.
# [pageUrl] the client request pageUrl.
# we also support datetime variables.
# [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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace 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]"
# @remark empty to ignore this exec.
publish ./objs/ffmpeg/bin/ffmpeg -f flv -i [url] -c copy -y ./[stream].flv;
}
}
#############################################################################################
# DVR to VoD file sections. You should use FFmpeg to record stream.
#############################################################################################
# vhost for dvr
vhost dvr.srs.com {
# DVR RTMP stream to file,
# start to record to file when encoder publish,
# reap flv/mp4 according by specified dvr_plan.
dvr {
# whether enabled dvr features
# Overwrite by env SRS_VHOST_DVR_ENABLED for all vhosts.
# default: off
enabled on;
# the filter for dvr to apply to.
# all, dvr all streams of all apps.
# <app>/<stream>, apply to specified stream of app.
# for example, to dvr the following two streams:
# live/stream1 live/stream2
# @remark Reload is disabled, @see https://github.com/ossrs/srs/issues/2181
# default: all
dvr_apply all;
# the dvr plan. canbe:
# session reap flv/mp4 when session end(unpublish).
# segment reap flv/mp4 when flv duration exceed the specified dvr_duration.
# @remark The plan append is removed in SRS3+, for it's no use.
# Overwrite by env SRS_VHOST_DVR_DVR_PLAN for all vhosts.
# default: session
dvr_plan session;
# the dvr output path, *.flv or *.mp4.
# 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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace 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;
# 5. DVR to mp4:
# dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].mp4;
# =>
# dvr_path ./objs/nginx/html/live/livestream.1420254068776.mp4;
# @see https://ossrs.io/lts/en-us/docs/v7/doc/dvr#custom-path
# segment,session apply it.
# Overwrite by env SRS_VHOST_DVR_DVR_PATH for all vhosts.
# default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
# the duration for dvr file, reap if exceed, in seconds.
# segment apply it.
# session,append ignore.
# Overwrite by env SRS_VHOST_DVR_DVR_DURATION for all vhosts.
# 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.
# Overwrite by env SRS_VHOST_DVR_DVR_WAIT_KEYFRAME for all vhosts.
# 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 stream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# apply for all dvr plan.
# Overwrite by env SRS_VHOST_DVR_TIME_JITTER for all vhosts.
# 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
# @see https://ossrs.io/lts/en-us/docs/v7/doc/dvr#http-callback
# @see https://ossrs.io/lts/en-us/docs/v7/doc/dvr#http-callback
}
}
#############################################################################################
# Ingest stream to SRS sections. You should use FFmpeg to ingest stream to SRS.
#############################################################################################
# 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.
# @remark vhost can contains multiple ingest
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 specified by url.
# stream: ingest stream specified 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.
# we also support datetime variables.
# [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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace 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]"
output rtmp://127.0.0.1:[port]/live/livestream?vhost=[vhost];
}
}
}
#############################################################################################
# Transcode stream by FFmpeg sections. You should use FFmpeg to transcode stream.
#############################################################################################
# the main comments for transcode
vhost example.transcode.srs.com {
# the streaming transcode configs.
# @remark vhost can contains multiple transcode
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.
# we will build the parameters to fork ffmpeg:
# ffmpeg <perfile>
# -i <iformat>
# <vfilter>
# -vcodec <vcodec> -b:v <vbitrate> -r <vfps> -s <vwidth>x<vheight> -profile:v <vprofile> -preset <vpreset>
# <vparams>
# -acodec <acodec> -b:a <abitrate> -ar <asample_rate> -ac <achannels>
# <aparams>
# -f <oformat>
# -y <output>
engine example {
# whether the engine is enabled
# default: off.
enabled on;
# pre-file options, before "-i"
perfile {
re;
rtsp_transport tcp;
}
# input format "-i", 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, between "-i" and "-vcodec"
# 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, "ffmpeg -vcodec"
# can be:
# libx264: use h.264(libx264) video encoder.
# png: use png to snapshot thumbnail.
# copy: donot encoder the video stream, copy it.
# vn: disable video output.
vcodec libx264;
# video bitrate, in kbps, "ffmepg -b:v"
# @remark 0 to use source video bitrate.
# default: 0
vbitrate 1500;
# video framerate, "ffmepg -r"
# @remark 0 to use source video fps.
# default: 0
vfps 25;
# video width, must be even numbers, "ffmepg -s"
# @remark 0 to use source video width.
# default: 0
vwidth 768;
# video height, must be even numbers, "ffmepg -s"
# @remark 0 to use source video height.
# default: 0
vheight 320;
# the max threads for ffmpeg to used, "ffmepg -thread"
# default: 1
vthreads 12;
# x264 profile, "ffmepg -profile:v"
# @see x264 -help, can be:
# high,main,baseline
vprofile main;
# x264 preset, "ffmpeg -preset"
# @see x264 -help, can be:
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
vpreset medium;
# other x264 or ffmpeg video params, between "-preset" and "-acodec"
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, "ffmpeg -acodec"
# 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, "ffmpeg -b:a"
# [16, 72] for libfdk_aac.
# @remark 0 to use source audio bitrate.
# default: 0
abitrate 70;
# audio sample rate, "ffmpeg -ar"
# 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, "ffmpeg -ac"
# 1 for mono, 2 for stereo.
# @remark 0 to use source audio channels.
# default: 0
achannels 2;
# other ffmpeg audio params, between "-ac" and "-f"/"-y"
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, "ffmpeg -f" can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# image2, for vcodec png to snapshot thumbnail.
# other format, for example, mp4/aac whatever.
# default: flv
oformat flv;
# output stream, "ffmpeg -y", variables:
# [vhost] the input stream vhost.
# [port] the input stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the transcode engine name.
# [param] the input stream query string.
# we also support datetime variables.
# [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], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace 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]"
output rtmp://127.0.0.1:[port]/[app]/[stream]_[engine]?vhost=[vhost];
}
}
}
#
#
#
#
#
#
#********************************************************************************************
#********************************************************************************************
# CAUTION: The features below are deprecated, so please avoid use it.
#********************************************************************************************
#********************************************************************************************
#############################################################################################
# The origin cluster section. Replaced by proxy server.
#############################################################################################
# Note that the legacy origin cluster is deprecated, please use the new architecture of
# Origin Cluster as a Proxy Cluster instead, please read https://ossrs.io/lts/en-us/docs/v7/doc/origin-cluster
# for details.
#############################################################################################
# Edge cluster sections. Will be replaced by relay server, see https://github.com/ossrs/srs/discussions/4402
#############################################################################################
# vhost for stream cluster for RTMP/FLV
vhost cluster.srs.com {
# The config for cluster.
cluster {
# The cluster mode, local or remote.
# local: It's an origin server, serve streams itself.
# remote: It's an edge server, fetch or push stream to origin server.
# default: local
mode remote;
# For edge(mode remote), 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(mode remote), 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;
# For edge(mode remote), 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;
# For edge(mode remote), 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://ossrs.io/lts/en-us/docs/v7/doc/log
# default: on
debug_srs_upnode on;
# For origin(mode local) cluster, turn on the cluster.
# @remark Origin cluster only supports RTMP, use Edge to transmux RTMP to FLV.
# default: off
# TODO: FIXME: Support reload.
origin_cluster off;
# For origin (mode local) cluster, the co-worker's HTTP APIs.
# This origin will connect to co-workers and communicate with them.
# please see https://ossrs.io/lts/en-us/docs/v7/doc/origin-cluster#legacy
# TODO: FIXME: Support reload.
coworkers 127.0.0.1:9091 127.0.0.1:9092;
# The protocol to connect to origin.
# rtmp, Connect origin by RTMP
# flv, Connect origin by HTTP-FLV
# flvs, Connect origin by HTTPS-FLV
# Default: rtmp
protocol rtmp;
# Whether follow client protocol to connect to origin.
# @remark The FLV might use different signature(in query string) to RTMP.
# Default: off
follow_client off;
}
}
#############################################################################################
# heartbeat/stats sections
#############################################################################################
# Query the latest available version of SRS, write a log to notice user to upgrade.
# @see https://github.com/ossrs/srs/issues/2424
# @see https://github.com/ossrs/srs/issues/2508
# Overwrite by env SRS_QUERY_LATEST_VERSION
# Default: off
query_latest_version off;
# First wait when qlv(query latest version), in seconds.
# Only available when qlv is enabled.
# Overwrite by env SRS_FIRST_WAIT_FOR_QLV
# Default: 300
first_wait_for_qlv 300;
# 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 enabled.
# Overwrite by env SRS_HEARTBEAT_ENABLED
# 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,....
# Overwrite by env SRS_HEARTBEAT_INTERVAL
# 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"
# }
# Overwrite by env SRS_HEARTBEAT_URL
# default: http://127.0.0.1:8085/api/v1/servers
url http://127.0.0.1:8085/api/v1/servers;
# the id of device.
# Overwrite by env SRS_HEARTBEAT_DEVICE_ID
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.
# Overwrite by env SRS_HEARTBEAT_SUMMARIES
# default: off
summaries off;
# Whether report with listen ports.
# if on, request with the ports of SRS:
# {
# "rtmp": ["1935"], "http": ["8080"], "api": ["1985"], "srt": ["10080"], "rtc": ["8000"]
# }
# Overwrite by env SRS_HEARTBEAT_PORTS
# default: off
ports 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 {
# Whether enable the stat of system resources.
# Default: on
enabled on;
# 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 configured.
disk sda sdb xvda xvdb;
}
#############################################################################################
# To prevent user to use full.conf
#############################################################################################
# To identify the full.conf
# @remark Should never use it directly, it's only a collections of all config items.
# Default: off
is_full on;
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