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# all config for srs
#############################################################################################
# RTMP sections
#############################################################################################
# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
# for example, 192.168.1.100:1935 10.10.10.100:1935
# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
listen 1935;
# the pid file
# to ensure only one process can use a pid file
# and provides the current running process id, for script,
# for example, init.d script to manage the server.
# default: ./objs/srs.pid
pid ./objs/srs.pid;
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# however, most clients support it and it can improve
# performance about 10%.
# default: 60000
chunk_size 60000;
# the log dir for FFMPEG.
# if enabled ffmpeg, each transcoding stream will create a log file.
# /dev/null to disable the log.
# default: ./objs
ff_log_dir ./objs;
# the log level for FFMPEG.
# info warning error fatal panic quiet
# trace debug verbose
# 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.
# default: file.
srs_log_tank file;
# the log level, for all log tanks.
# can be: verbose, info, trace, warn, error
# default: trace
srs_log_level trace;
# when srs_log_tank is file, specifies the log file.
# default: ./objs/srs.log
srs_log_file ./objs/srs.log;
# the max connections.
# if exceed the max connections, server will drop the new connection.
# default: 1000
max_connections 1000;
# whether start as daemon
# @remark: do not support reload.
# default: on
daemon on;
# whether use utc_time to generate the time struct,
# if off, use localtime() to generate it,
# if on, use gmtime() instead, which use UTC time.
# default: off
utc_time off;
# config for the pithy print in ms,
# which always print constant message specified by interval,
# whatever the clients in concurrency.
# 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.
# default: ./
work_dir ./;
# whether quit when parent process changed,
# used for supervisor mode(not daemon), srs should always quit when
# supervisor process exited.
# @remark conflict with daemon, error when both daemon and asprocess are on.
# @reamrk do not support reload.
# default: off
asprocess off;
# Whether client empty IP is ok, for example, health checking by SLB.
# If ok(on), we will ignore this connection without warnings or errors.
# default: on
empty_ip_ok on;
# 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
# 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
# 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
# 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.
# default: on
disable_daemon_for_docker on;
# Whether auto reload by watching the config file by inotify.
# 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.
# 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
# default: 0.8
tcmalloc_release_rate 0.8;
#############################################################################################
# heartbeat/stats sections
#############################################################################################
# heartbeat to api server
# @remark, the ip report to server, is retrieve from system stat,
# which need the config item stats.network.
heartbeat {
# whether heartbeat is 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,....
# default: 9.9
interval 9.3;
# when startup, srs will heartbeat to this api.
# @remark: must be a restful http api url, where SRS will POST with following data:
# {
# "device_id": "my-srs-device",
# "ip": "192.168.1.100"
# }
# default: http://127.0.0.1:8085/api/v1/servers
url http://127.0.0.1:8085/api/v1/servers;
# the id of device.
device_id "my-srs-device";
# whether report with summaries
# if on, put /api/v1/summaries to the request data:
# {
# "summaries": summaries object.
# }
# @remark: optional config.
# default: off
summaries off;
}
# system statistics section.
# the main cycle will retrieve the system stat,
# for example, the cpu/mem/network/disk-io data,
# the http api, for instance, /api/v1/summaries will show these data.
# @remark the heartbeat depends on the network,
# for example, the eth0 maybe the device which index is 0.
stats {
# the index of device ip.
# we may retrieve more than one network device.
# default: 0
network 0;
# the device name to stat the disk iops.
# ignore the device of /proc/diskstats if not configured.
disk sda sdb xvda xvdb;
}
#############################################################################################
# HTTP sections
#############################################################################################
# api of srs.
# the http api config, export for external program to manage srs.
# user can access http api of srs in browser directly, for instance, to access by:
# curl http://192.168.1.170:1985/api/v1/reload
# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
# where the cli can only be used in shell/terminate.
http_api {
# whether http api is enabled.
# default: off
enabled on;
# the http api listen entry is <[ip:]port>
# for example, 192.168.1.100:1985
# where the ip is optional, default to 0.0.0.0, that is 1985 equals to 0.0.0.0:1985
# default: 1985
listen 1985;
# whether enable crossdomain request.
# default: on
crossdomain on;
# the HTTP RAW API is more powerful api to change srs state and reload.
raw_api {
# whether enable the HTTP RAW API.
# default: off
enabled off;
# whether enable rpc reload.
# default: off
allow_reload off;
# whether enable rpc query.
# default: off
allow_query off;
# whether enable rpc update.
# default: off
allow_update off;
}
}
# 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.
# default: off
enabled on;
# the http streaming listen entry is <[ip:]port>
# for example, 192.168.1.100:8080
# where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
# @remark, if use lower port, for instance 80, user must start srs by root.
# default: 8080
listen 8080;
# the default dir for http root.
# default: ./objs/nginx/html
dir ./objs/nginx/html;
# whether enable crossdomain request.
# for both http static and stream server and apply on all vhosts.
# default: on
crossdomain on;
}
#############################################################################################
# Streamer sections
#############################################################################################
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
# MPEGTS over UDP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# mpegts_over_udp, MPEG-TS over UDP caster.
caster mpegts_over_udp;
# the output rtmp url.
# for mpegts_over_udp caster, the typically output url:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
listen 8935;
}
# RTSP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# rtsp, Real Time Streaming Protocol (RTSP).
caster rtsp;
# the output rtmp url.
# for rtsp caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, the rtsp url:
# rtsp://192.168.1.173:8544/live/livestream.sdp
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/[app]/[stream];
# the listen port for stream caster.
# for rtsp caster, listen at tcp port. for example, 554.
listen 554;
# for the rtsp caster, the rtp server local port over udp,
# which reply the rtsp setup request message, the port will be used:
# [rtp_port_min, rtp_port_max)
rtp_port_min 57200;
rtp_port_max 57300;
}
# FLV
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# flv, FLV over HTTP by POST.
caster flv;
# the output rtmp url.
# for flv caster, 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 caster.
# for flv caster, listen at tcp port. for example, 8936.
listen 8936;
}
# GB28181
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# gb28181, Push GB28181 to SRS.
caster gb28181;
# the output rtmp url.
# for gb28181 caster, the typically output url:
# rtmp://127.0.0.1/live/[stream]
# where the [stream] is the VideoChannelCodecID.
output rtmp://127.0.0.1/live/[stream];
# the listen port for stream caster.
# for gb28181 caster, listen at udp port. for example, 9000.
# @remark We can bundle all gb28181 to this port, to reuse this port.
# User can choose to bundle port in API port_mode or SIP invite_port_fixed.
listen 9000;
# If not bundle ports, use specified ports for each stream.
rtp_port_min 58200;
rtp_port_max 58300;
# Whether wait for keyframe then forward to RTMP.
wait_keyframe off;
# Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
# default: 30
rtp_idle_timeout 30;
# Whether has audio.
# @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
# default: off
audio_enable off;
# The exposed IP to receive media stream.
# * Retrieve server IP automatically, from all network interfaces.
# eth0 Retrieve server IP by specified network interface name. # TODO: Implements it.
# $CANDIDATE Read the IP from ENV variable $EIP, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
# x.x.x.x A specified IP address or DNS name, which can be access by client such as Chrome.
# You can specific more than one interface name:
# eth0 eth1 Use network interface eth0 and eth1. # TODO: Implements it.
# Also by IP or DNS names:
# 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# default: *
host *;
#The media channel is automatically created according to the received RTP packet,
# and the channel ID is generated according to the RTP SSRC
# channelid format: 'chid[ssrc]' [ssrc] is rtp's ssrc
auto_create_channel off;
sip {
# Whether enable embeded SIP server.
# default: on
enabled on;
# The SIP listen port.
# default: 5060
listen 5060;
# The SIP server ID.
# default: 34020000002000000001
serial 34020000002000000001;
# The SIP server domain.
# default: 3402000000
realm 3402000000;
# The SIP ACK response timeout in seconds.
# default: 30
ack_timeout 30;
# The keepalive timeout in seconds.
# default: 120
keepalive_timeout 120;
# Whether play immediately after registered.
# default: on
auto_play on;
# Whether bundle media stream port.
# default: on
invite_port_fixed on;
# interval to query equipment list from equipment or subordinate domain, unit(s)
# default: 60
query_catalog_interval 60;
}
}
#############################################################################################
# SRT server section
#############################################################################################
# @doc https://github.com/ossrs/srs/issues/1147#issuecomment-577607026
srt_server {
# whether SRT server is enabled.
# default: off
enabled on;
# The UDP listen port for SRT.
listen 10080;
# For detail parameters, please read wiki:
# https://github.com/ossrs/srs/wiki/v4_CN_SRTParams
# https://github.com/ossrs/srs/wiki/v4_EN_SRTParams
maxbw 1000000000;
connect_timeout 4000;
peerlatency 300;
recvlatency 300;
# Default app for vmix, see https://github.com/ossrs/srs/pull/1615
# default: live
default_app live;
}
#############################################################################################
# WebRTC server section
#############################################################################################
rtc_server {
# Whether enable WebRTC server.
# default: off
enabled on;
# The udp listen port, we will reuse it for connections.
# default: 8080
listen 8000;
# The exposed candidate IPs, response in SDP candidate line. It can be:
# * Retrieve server IP automatically, from all network interfaces.
# eth0 Retrieve server IP by specified network interface name. # TODO: Implements it.
# $CANDIDATE Read the IP from ENV variable $EIP, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
# x.x.x.x A specified IP address or DNS name, which can be access by client such as Chrome.
# You can specific more than one interface name:
# eth0 eth1 Use network interface eth0 and eth1. # TODO: Implements it.
# Also by IP or DNS names:
# 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
# And by multiple ENV variables:
# $CANDIDATE $EIP # TODO: Implements it.
# default: *
candidate *;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# default: on
ecdsa on;
# The max UDP messages send by sendmmsg.
# @remark Always 1 if OS does not support sendmmsg, like OSX.
# default: 256
sendmmsg 256;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# 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.
# default: 4
reuseport 4;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# default: on
merge_nalus on;
# Whether enable GSO to send out RTP packets.
# @remark Linux 4.18+ only, for other OS always disabled.
# default: on
gso on;
# Whether pad first packet for GSO for padding bytes.
# If 0, disable padding for GSO.
# @remark The max padding size is 0x7f(127).
# default: 127
padding 127;
# Whether enable the perf stat at http://localhost:1985/api/v1/perf
# default: on
perf_stat on;
# The queue length, in number of mmsghdr, in messages.
# For example, 30 means we will cache 30K messages at most.
# If exceed, we will drop messages.
# @remark Each reuseport use a dedicated queue, if queue is 2000, reuseport is 4,
# then system queue is 2000*4 = 8k, user can incrase reuseport to incrase the queue.
# default: 2000
queue_length 2000;
}
vhost rtc.vhost.srs.com {
rtc {
# Whether enable WebRTC server.
# default: off
enabled on;
# The strategy for bframe.
# keep Keep bframe, which may make browser with playing problems.
# discard Discard bframe, maybe cause browser with little problems.
# default: keep
bframe discard;
# The strategy for aac audio.
# transcode Transcode aac to opus.
# discard Discard aac audio packet.
# default: transcode
aac transcode;
# The timeout in seconds for session timeout.
# Client will send ping(STUN binding request) to server, we use it as heartbeat.
# default: 30
stun_timeout 30;
# The strick check when process stun.
# default: off
stun_strict_check on;
}
# whether enable min delay mode for vhost.
# default: on, for RTC.
min_latency on;
play {
# set the MW(merged-write) latency in ms.
# @remark For WebRTC, we enable pass-timestamp mode, so we ignore this config.
# default: 0 (For WebRTC)
mw_latency 0;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, min_latency on)
# default: 1 (For WebRTC, min_latency off)
mw_msgs 0;
}
}
#############################################################################################
# RTMP/HTTP VHOST sections
#############################################################################################
# vhost list, the __defaultVhost__ is the default vhost
# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
# for which cannot identify the required vhost.
vhost __defaultVhost__ {
}
# the vhost scope configs.
vhost scope.vhost.srs.com {
# whether the vhost is enabled.
# if off, all request access denied.
# default: on
enabled off;
# 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
# 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.
# 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.
# 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.
# Default: 2500000
out_ack_size 2500000;
}
# set the chunk size of vhost.
vhost chunksize.srs.com {
# @see scope.vhost.srs.com
chunk_size 128;
}
# the vhost disabled.
vhost removed.srs.com {
# @see scope.vhost.srs.com
enabled off;
}
# 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://github.com/ossrs/srs/wiki/v1_CN_SrsLog
# 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 read: https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster
# TODO: FIXME: Support reload.
coworkers 127.0.0.1:9091 127.0.0.1:9092;
}
}
# vhost for edge, edge and origin is the same vhost
vhost same.edge.srs.com {
# @see cluster.srs.com
cluster {
mode remote;
origin 127.0.0.1:1935 localhost:1935;
token_traverse off;
}
}
# vhost for edge, edge transform vhost to fetch from another vhost.
vhost transform.edge.srs.com {
# @see cluster.srs.com
cluster {
mode remote;
origin 127.0.0.1:1935;
vhost same.edge.srs.com;
}
}
# the vhost for srs debug info, whether send args in connect(tcUrl).
vhost debug.srs.com {
# @see cluster.srs.com
cluster {
debug_srs_upnode on;
}
}
# 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;
}
}
# the play specified configs
vhost play.srs.com {
# for play client, both RTMP and other 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.
# default: on
gop_cache off;
# the max live queue length in seconds.
# if the messages in the queue exceed the max length,
# drop the old whole gop.
# default: 30
queue_length 10;
# 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.
# 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
# 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).
# 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.
# 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-timestamp mode, so we ignore this config.
# default: 350 (For RTMP/HTTP-FLV)
# 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).
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.
# default: 0
send_min_interval 10.0;
# whether reduce the sequence header,
# for some client which cannot got duplicated sequence header,
# while the sequence header is not changed yet.
# default: off
reduce_sequence_header on;
}
}
# vhost for time jitter
vhost jitter.srs.com {
# @see play.srs.com
# to use time_jitter full, the default config.
play {
}
# to use mix_correct.
play {
time_jitter off;
mix_correct on;
}
play {
atc on;
mix_correct on;
}
# to use atc
play {
atc on;
}
}
# vhost for atc.
vhost atc.srs.com {
# @see play.srs.com
play {
atc on;
atc_auto on;
}
}
# the MR(merged-read) setting for publisher.
# the MW(merged-write) settings for player.
vhost mrw.srs.com {
# @see scope.vhost.srs.com
min_latency off;
# @see play.srs.com
play {
mw_latency 350;
mw_msgs 8;
}
# @see publish.srs.com
publish {
mr on;
mr_latency 350;
}
}
# the vhost for min delay, do not cache any stream.
vhost min.delay.com {
# @see scope.vhost.srs.com
min_latency on;
# @see scope.vhost.srs.com
tcp_nodelay on;
# @see play.srs.com
play {
mw_latency 100;
mw_msgs 4;
gop_cache off;
queue_length 10;
}
# @see publish.srs.com
publish {
mr off;
}
}
# whether disable the sps parse, for the resolution of video.
vhost no.parse.sps.com {
# @see publish.srs.com
publish {
parse_sps on;
}
}
# the vhost to control the stream delivery feature
vhost stream.control.com {
# @see scope.vhost.srs.com
min_latency on;
# @see scope.vhost.srs.com
tcp_nodelay on;
# @see play.srs.com
play {
mw_latency 100;
mw_msgs 4;
queue_length 10;
send_min_interval 10.0;
reduce_sequence_header on;
}
# @see publish.srs.com
publish {
mr off;
firstpkt_timeout 20000;
normal_timeout 7000;
}
}
# the publish specified configs
vhost publish.srs.com {
# the config for FMLE/Flash publisher, which push RTMP to SRS.
publish {
# about MR, read https://github.com/ossrs/srs/issues/241
# when enabled the mr, SRS will read as large as possible.
# 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]
# default: 350
mr_latency 350;
# the 1st packet timeout in ms for encoder.
# default: 20000
firstpkt_timeout 20000;
# the normal packet timeout in ms for encoder.
# 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.
# default: on
parse_sps on;
}
}
# 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;
}
}
# vhost for bwt(bandwidth check)
# generally, the bandcheck vhost must be: bandcheck.srs.com,
# or need to modify the vhost of client.
vhost bandcheck.srs.com {
enabled on;
chunk_size 65000;
# bandwidth check config.
bandcheck {
# whether support bandwidth check,
# default: off.
enabled on;
# the key for server to valid,
# if invalid key, server disconnect and abort the bandwidth check.
key "35c9b402c12a7246868752e2878f7e0e";
# the interval in seconds for bandwidth check,
# server do not allow new test request.
# default: 30
interval 30;
# the max available check bandwidth in kbps.
# to avoid attack of bandwidth check.
# default: 1000
limit_kbps 4000;
}
}
# the security to allow or deny clients.
vhost security.srs.com {
# security for host to allow or deny clients.
# @see https://github.com/ossrs/srs/issues/211
security {
# whether enable the security for vhost.
# default: off
enabled on;
# the security list, each item format as:
# allow|deny publish|play all|<ip>
# for example:
# allow publish all;
# deny publish all;
# allow publish 127.0.0.1;
# deny publish 127.0.0.1;
# allow play all;
# deny play all;
# allow play 127.0.0.1;
# deny play 127.0.0.1;
# SRS apply the following simple strategies one by one:
# 1. allow all if security disabled.
# 2. default to deny all when security enabled.
# 3. allow if matches allow strategy.
# 4. deny if matches deny strategy.
allow play all;
allow publish all;
}
}
# vhost for http static and flv vod stream for each vhost.
vhost http.static.srs.com {
# http static vhost specified config
http_static {
# whether enabled the http static service for vhost.
# default: off
enabled on;
# the url to mount to,
# typical mount to [vhost]/
# the variables:
# [vhost] current vhost for http server.
# @remark the [vhost] is optional, used to mount at specified vhost.
# @remark the http of __defaultVhost__ will override the http_server section.
# for example:
# mount to [vhost]/
# access by http://ossrs.net:8080/xxx.html
# mount to [vhost]/hls
# access by http://ossrs.net:8080/hls/xxx.html
# mount to /
# access by http://ossrs.net:8080/xxx.html
# or by http://192.168.1.173:8080/xxx.html
# mount to /hls
# access by http://ossrs.net:8080/hls/xxx.html
# or by http://192.168.1.173:8080/hls/xxx.html
# @remark the port of http is specified by http_server section.
# default: [vhost]/
mount [vhost]/hls;
# main dir of vhost,
# to delivery HTTP stream of this vhost.
# default: ./objs/nginx/html
dir ./objs/nginx/html/hls;
}
}
# vhost for http flv/aac/mp3 live stream for each vhost.
vhost http.remux.srs.com {
# http flv/mp3/aac/ts stream vhost specified config
http_remux {
# whether enable the http live streaming service for vhost.
# default: off
enabled on;
# the fast cache for audio stream(mp3/aac),
# to cache more audio and send to client in a time to make android(weixin) happy.
# @remark the flv/ts stream ignore it
# @remark 0 to disable fast cache for http audio stream.
# default: 0
fast_cache 30;
# the stream 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.
# default: [vhost]/[app]/[stream].flv
mount [vhost]/[app]/[stream].flv;
}
}
# 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.
# default off.
enabled on;
# when client connect to vhost/app, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_connect",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
# "pageUrl": "http://www.test.com/live.html"
# }
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client close/disconnect to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_close",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "send_bytes": 10240, "recv_bytes": 10240
# }
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client(encoder) publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_publish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy"
# }
# 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
on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client(encoder) stop publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_unpublish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy"
# }
# 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
on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client start to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_play",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy",
# "pageUrl": "http://www.test.com/live.html"
# }
# 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
on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
# when client stop to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_stop",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy"
# }
# 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
on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
# when srs reap a dvr file, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_dvr",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "param":"?token=xxx&salt=yyy",
# "cwd": "/usr/local/srs",
# "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
# }
# 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
on_dvr http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
# when srs reap a ts file of hls, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_hls",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream", "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
# }
# 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
on_hls http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
# when srs reap a ts file of hls, call this hook,
# used to push file to cdn network, by get the ts file from cdn network.
# so we use HTTP GET and use the variable following:
# [app], replace with the app.
# [stream], replace with the stream.
# [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.
on_hls_notify http://127.0.0.1:8085/api/v1/hls/[app]/[stream]/[ts_url][param];
}
}
# 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;
}
}
# The vhost for MPEG-DASH.
vhost dash.srs.com {
dash {
# Whether DASH is enabled.
# Transmux RTMP to DASH if on.
# Default: off
enabled on;
# The duration of segment in seconds.
# Default: 30
dash_fragment 30;
# The period to update the MPD in seconds.
# Default: 150
dash_update_period 150;
# The depth of timeshift buffer in seconds.
# Default: 300
dash_timeshift 300;
# The base/home dir/path for dash.
# All init and segment files will write under this dir.
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.
# Default: [app]/[stream].mpd
dash_mpd_file [app]/[stream].mpd;
}
}
# 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.
# default: off
enabled on;
# the hls fragment in seconds, the duration of a piece of ts.
# default: 10
hls_fragment 10;
# the hls m3u8 target duration ratio,
# EXT-X-TARGETDURATION = hls_td_ratio * hls_fragment // init
# EXT-X-TARGETDURATION = max(ts_duration, EXT-X-TARGETDURATION) // for each ts
# @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
# default: 1.5
hls_td_ratio 1.5;
# the audio overflow ratio.
# for pure audio, the duration to reap the segment.
# for example, the hls_fragment is 10s, hls_aof_ratio is 2.0,
# the segment will reap to 20s for pure audio.
# default: 2.0
hls_aof_ratio 2.0;
# the hls window in seconds, the number of ts in m3u8.
# default: 60
hls_window 60;
# the error strategy. can be:
# ignore, disable the hls.
# disconnect, require encoder republish.
# continue, ignore failed try to continue output hls.
# @see https://github.com/ossrs/srs/issues/264
# default: continue
hls_on_error continue;
# the hls output path.
# the m3u8 file is 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.
# default: ./objs/nginx/html
hls_path ./objs/nginx/html;
# the hls m3u8 file name.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# default: [app]/[stream].m3u8
hls_m3u8_file [app]/[stream].m3u8;
# the hls ts file name.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# [2006], replace this const to current year.
# [01], replace this const to current month.
# [02], replace this const to current date.
# [15], replace this const to current hour.
# [04], 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://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
# default: [app]/[stream]-[seq].ts
hls_ts_file [app]/[stream]-[seq].ts;
# whether use floor for the hls_ts_file path generation.
# if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
# and use enhanced algorithm to calc deviation for segment.
# @remark when floor on, recommend the hls_segment>=2*gop.
# default: off
hls_ts_floor off;
# the hls entry prefix, which is base url of ts url.
# 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
# ...
# optional, default to empty string.
hls_entry_prefix http://your-server;
# the default audio codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for mp3.
# the available audio codec:
# aac, mp3, an
# default: aac
hls_acodec aac;
# the default video codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for pure audio(without video) to vn.
# the available video codec:
# h264, vn
# default: h264
hls_vcodec h264;
# whether cleanup the old expired ts files.
# default: on
hls_cleanup on;
# 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.
# default: 0
hls_dispose 0;
# the max size to notify hls,
# to read max bytes from ts of specified cdn network,
# @remark only used when on_hls_notify is config.
# default: 64
hls_nb_notify 64;
# whether wait keyframe to reap segment,
# if off, reap segment when duration exceed the fragment,
# if on, reap segment when duration exceed and got keyframe.
# default: on
hls_wait_keyframe on;
# whether using AES encryption.
# default: off
hls_keys on;
# the number of clear ts which one key can encrypt.
# 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.
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
hls_key_file_path ./objs/nginx/html;
# the key root URL, use this can support https.
# @remark It's optional.
hls_key_url https://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.
# 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
# @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
# @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
# on_hls_notify, never config in here, should config in http_hooks.
# we support the variables to generate the notify url:
# [app], replace with the app.
# [stream], replace with the stream.
# [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
# @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#on-hls-notify
# @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#on-hls-notify
}
}
# the vhost with hls disabled.
vhost no-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, do not write hls(ts and m3u8) when publish.
# default: off
enabled off;
}
}
# the vhost with adobe hds
vhost hds.srs.com {
hds {
# whether hds enabled
# default: off
enabled on;
# the hds fragment in seconds.
# default: 10
hds_fragment 10;
# the hds window in seconds, erase the segment when exceed the window.
# default: 60
hds_window 60;
# the path to store the hds files.
# default: ./objs/nginx/html
hds_path ./objs/nginx/html;
}
}
# 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
# 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
# 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.
# 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://github.com/ossrs/srs/wiki/v3_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v3_EN_DVR#custom-path
# segment,session apply it.
# default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
# the duration for dvr file, reap if exceed, in seconds.
# segment apply it.
# session,append ignore.
# default: 30
dvr_duration 30;
# whether wait keyframe to reap segment,
# if off, reap segment when duration exceed the dvr_duration,
# if on, reap segment when duration exceed and got keyframe.
# segment apply it.
# session,append ignore.
# default: on
dvr_wait_keyframe on;
# about the stream monotonically increasing:
# 1. video timestamp is monotonically increasing,
# 2. audio timestamp is monotonically increasing,
# 3. video and audio timestamp is interleaved monotonically increasing.
# it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
# however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
# the time jitter algorithm:
# 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
# 2. zero, only ensure stream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# apply for all dvr plan.
# default: full
time_jitter full;
# on_dvr, never config in here, should config in http_hooks.
# for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
# @read https://github.com/ossrs/srs/wiki/v2_EN_DVR#http-callback
}
}
# vhost for ingest
vhost ingest.srs.com {
# ingest file/stream/device then push to SRS over RTMP.
# the name/id used to identify the ingest, must be unique in global.
# ingest id is used in reload or http api management.
# @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?vhost=[vhost]/livestream;
}
}
}
# the vhost for ingest with transcode engine.
vhost transcode.ingest.srs.com {
ingest livestream {
enabled on;
input {
type file;
url ./doc/source.200kbps.768x320.flv;
}
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
perfile {
re;
rtsp_transport tcp;
}
iformat flv;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 1500;
vfps 25;
vwidth 768;
vheight 320;
vthreads 12;
vprofile main;
vpreset medium;
vparams {
t 100;
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
profile:a aac_low;
}
oformat flv;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream];
}
}
}
# 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.
# 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]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
vhost mirror.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine mirror {
enabled on;
vfilter {
vf 'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
#######################################################################################################
# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
vhost crop.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine crop {
enabled on;
vfilter {
vf 'crop=in_w-20:in_h-160:10:80';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
vhost logo.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine logo {
enabled on;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# audio transcode only.
# for example, FMLE publish audio codec in mp3, and do not support HLS output,
# we can transcode the audio to aac and copy video to the new stream with HLS.
vhost audio.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine acodec {
enabled on;
vcodec copy;
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# disable video, transcode/copy audio.
# for example, publish pure audio stream.
vhost vn.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine vn {
enabled on;
vcodec vn;
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# ffmpeg-copy(forward implements by ffmpeg).
# copy the video and audio to a new stream.
vhost copy.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of vhost
# the comments, read example.transcode.srs.com
vhost all.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ffsuper {
enabled on;
iformat flv;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 1500;
vfps 25;
vwidth 768;
vheight 320;
vthreads 12;
vprofile main;
vpreset medium;
vparams {
t 100;
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
profile:a aac_low;
}
oformat flv;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffhd {
enabled on;
vcodec libx264;
vbitrate 1200;
vfps 25;
vwidth 1382;
vheight 576;
vthreads 6;
vprofile main;
vpreset medium;
vparams {
}
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffsd {
enabled on;
vcodec libx264;
vbitrate 800;
vfps 25;
vwidth 1152;
vheight 480;
vthreads 4;
vprofile main;
vpreset fast;
vparams {
}
acodec libfdk_aac;
abitrate 60;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine fffast {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine vcopy {
enabled on;
vcodec copy;
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine acopy {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all stream using the empty ffmpeg demo, do nothing.
vhost ffempty.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/research/ffempty;
engine empty {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of app
vhost app.transcode.srs.com {
# the streaming transcode configs.
# if app specified, transcode all streams of app.
transcode live {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
# transcode specified stream.
vhost stream.transcode.srs.com {
# the streaming transcode configs.
# if stream specified, transcode the matched stream.
transcode live/livestream {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
#############################################################################################
# The origin cluster section
#############################################################################################
http_api {
enabled on;
listen 9090;
}
vhost a.origin.cluster.srs.com {
cluster {
mode local;
origin_cluster on;
coworkers 127.0.0.1:9091;
}
}
http_api {
enabled on;
listen 9091;
}
vhost b.origin.cluster.srs.com {
cluster {
mode local;
origin_cluster on;
coworkers 127.0.0.1:9090;
}
}
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