# 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.
#############################################################################################
# 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;
#############################################################################################
# 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
# 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;
#############################################################################################
# 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 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.
# Note that you're able to use a dedicated port for HTTP API, such as 1985, to be different with HTTP server. In
# this situation, you you must also set another HTTPS API port.
# 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.
# Note that you're able to use a dedicated port for HTTPS API, such as 1990, and the HTTP API should not be
# the same of HTTP server(8080) neither.
# 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=CN/ST=Beijing/L=Beijing/O=Me/OU=Me/CN=ossrs.net"
# 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 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.
# 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.
# 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=CN/ST=Beijing/L=Beijing/O=Me/OU=Me/CN=ossrs.net"
# Overwrite by env SRS_HTTP_SERVER_HTTPS_CERT
# default: ./conf/server.crt
cert ./conf/server.crt;
}
}
#############################################################################################
# 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 port for SRT.
# Overwrite by env SRS_SRT_SERVER_LISTEN
listen 10080;
# For detail parameters, please read wiki:
# @see https://ossrs.net/lts/zh-cn/docs/v5/doc/srt-params
# @see https://ossrs.io/lts/en-us/docs/v5/doc/srt-params
# 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;
# 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: on
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 port, we will reuse it for connections.
# 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 port for WebRTC. 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.net/lts/zh-cn/docs/v4/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;
}
###############################################################
# 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;
}
}
#############################################################################################
# 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];
Support HTTP-API for fetching reload result. v5.0.176 v6.0.71 (#3779)
## Reload Error Ignore
During a Reload, several stages will be passed through:
1. Parsing new configurations: Parse.
2. Transforming configurations: Transform.
3. Applying configurations: Apply.
Previously, any error at any stage would result in a direct exit, making
the system completely dependent on configuration checks:
```bash
./objs/srs -c conf/srs.conf -t
echo $?
#0
```
Optimized to: If an error occurs before applying the configuration, it
can be ignored. If an error occurs during the application of the
configuration, some of the configuration may have already taken effect,
leading to unpredictable behavior, so SRS will exit directly.
## Reload Fetch API
Added a new HTTP API to query the result of the reload.
```nginx
http_api {
enabled on;
raw_api {
enabled on;
allow_reload on;
}
}
```
```bash
curl http://localhost:1985/api/v1/raw?rpc=reload-fetch
```
```json
{
"code": 0,
"data": {
"err": 0,
"msg": "Success",
"state": 0,
"rid": "0s6y0n9"
}
}
{
"code": 0,
"data": {
"err": 1023,
"msg": "code=1023(ConfigInvalid) : parse file : parse buffer containers/conf/srs.release-local.conf : root parse : parse dir : parse include buffer containers/data/config/srs.vhost.conf : read token, line=0, state=0 : line 3: unexpected end of file, expecting ; or \"}\"",
"state": 1,
"rid": "0g4z471"
}
}
```
This way, you can know if the last reload of the system was successful.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
1 year ago
# 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. Please note that this is only a demonstrated SIP server, please never use it in your
# online production environment. Instead please use [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.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip).
sip {
# Whether enable embedded SIP server. Please disable it if you want to use your own SIP server, see
# [GB: External SIP](https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip).
# Default: on
enabled on;
Support HTTP-API for fetching reload result. v5.0.176 v6.0.71 (#3779)
## Reload Error Ignore
During a Reload, several stages will be passed through:
1. Parsing new configurations: Parse.
2. Transforming configurations: Transform.
3. Applying configurations: Apply.
Previously, any error at any stage would result in a direct exit, making
the system completely dependent on configuration checks:
```bash
./objs/srs -c conf/srs.conf -t
echo $?
#0
```
Optimized to: If an error occurs before applying the configuration, it
can be ignored. If an error occurs during the application of the
configuration, some of the configuration may have already taken effect,
leading to unpredictable behavior, so SRS will exit directly.
## Reload Fetch API
Added a new HTTP API to query the result of the reload.
```nginx
http_api {
enabled on;
raw_api {
enabled on;
allow_reload on;
}
}
```
```bash
curl http://localhost:1985/api/v1/raw?rpc=reload-fetch
```
```json
{
"code": 0,
"data": {
"err": 0,
"msg": "Success",
"state": 0,
"rid": "0s6y0n9"
}
}
{
"code": 0,
"data": {
"err": 1023,
"msg": "code=1023(ConfigInvalid) : parse file : parse buffer containers/conf/srs.release-local.conf : root parse : parse dir : parse include buffer containers/data/config/srs.vhost.conf : read token, line=0, state=0 : line 3: unexpected end of file, expecting ; or \"}\"",
"state": 1,
"rid": "0g4z471"
}
}
```
This way, you can know if the last reload of the system was successful.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
1 year ago
# The SIP listen port, for TCP protocol.
# Default: 5060
listen 5060;
# The SIP or media transport timeout in seconds.
# Default: 60
timeout 60;
# When media disconnect, the wait time in seconds to re-invite device to publish. During this wait time, device
# might send bye or unregister message(expire is 0), so that we will cancel the re-invite.
# Default: 5
reinvite 5;
# The exposed candidate IPs, response in SDP connection 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.
# Default: *
candidate *;
}
}
#############################################################################################
# other global sections
#############################################################################################
# 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;
# 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;
# For thread pool.
threads {
# The thread pool manager cycle interval, in seconds.
# Overwrite by env SRS_THREADS_INTERVAL
# Default: 5
interval 5;
}
# 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;
}
# TencentCloud CLS(Cloud Log Service) config, logging to cloud.
# See https://cloud.tencent.com/document/product/614/11254
tencentcloud_cls {
# Whether CLS is enabled.
# Overwrite by env SRS_TENCENTCLOUD_CLS_ENABLED
# default: off
enabled off;
# The logging label to category the cluster servers.
# Overwrite by env SRS_TENCENTCLOUD_CLS_LABEL
label cn-beijing;
# The logging tag to category the cluster servers.
# Overwrite by env SRS_TENCENTCLOUD_CLS_TAG
tag cn-edge;
# The SecretId to access CLS service, see https://console.cloud.tencent.com/cam/capi
# Overwrite by env SRS_TENCENTCLOUD_CLS_SECRET_ID
secret_id AKIDxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx;
# The SecretKey to access CLS service, see https://console.cloud.tencent.com/cam/capi
# Overwrite by env SRS_TENCENTCLOUD_CLS_SECRET_KEY
secret_key xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx;
# The endpoint of CLS, format as <Region>.cls.tencentcs.com. For example:
# ap-guangzhou.cls.tencentcs.com
# Note that tencentyun.com is for internal network, while tencentcs.com is for public internet.
# See https://cloud.tencent.com/document/product/614/18940
# Overwrite by env SRS_TENCENTCLOUD_CLS_ENDPOINT
endpoint ap-guangzhou.cls.tencentcs.com;
# The topic ID of CLS, see https://cloud.tencent.com/document/product/614/41035
# Overwrite by env SRS_TENCENTCLOUD_CLS_TOPIC_ID
topic_id xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx;
# Whether enable logging for each log sending.
# Overwrite by env SRS_TENCENTCLOUD_CLS_DEBUG_LOGGING
# Default: off
debug_logging off;
# Whether enable the heartbeat stat every (5 * heartbeat_ratio)s.
# Overwrite by env SRS_TENCENTCLOUD_CLS_STAT_HEARTBEAT
# Default: on
stat_heartbeat on;
# Setup the heartbeat interval ratio, 1 means 5s, 2 means 10s, etc.
# Overwrite by env SRS_TENCENTCLOUD_CLS_HEARTBEAT_RATIO
# Default: 1
heartbeat_ratio 1;
# Whether enable the streams stat every (5 * streams_ratio)s.
# Overwrite by env SRS_TENCENTCLOUD_CLS_STAT_STREAMS
# Default: on
stat_streams on;
# Setup the streams interval ratio, 1 means 5s, 2 means 10s, etc.
# Overwrite by env SRS_TENCENTCLOUD_CLS_STREAMS_RATIO
# Default: 1
streams_ratio 1;
}
# TencentCloud APM(Application Performance Management) config.
# See https://cloud.tencent.com/document/product/1463/57462
tencentcloud_apm {
# Whether APM is enabled.
# Overwrite by env SRS_TENCENTCLOUD_APM_ENABLED
# default: off
enabled on;
# The APM team or business system ID, to which spans belongs to. For example, the team is apm-FsOsPOIMl (just an
# example, not available), please get your team from https://console.cloud.tencent.com/apm/monitor/team
# Overwrite by env SRS_TENCENTCLOUD_APM_TEAM
team apm-xxxxxxxxx;
# The APM token for authentication. For example, the token is xzddEaegsxGadEpGEDFx (just an example, not available),
# please get your token from https://console.cloud.tencent.com/apm/monitor/access
# Overwrite by env SRS_TENCENTCLOUD_APM_TOKEN
token xxxxxxxx;
# The APM endpoint. See https://github.com/open-telemetry/opentelemetry-go/tree/main/exporters/otlp/otlptrace
# Please note that 4317 is for GRPC/HTTP2, while SRS only support HTTP and the port shoule be 55681.
# Overwrite by env SRS_TENCENTCLOUD_APM_ENDPOINT
endpoint ap-guangzhou.apm.tencentcs.com:55681;
# The service.name of resource.
# Overwrite by env SRS_TENCENTCLOUD_APM_SERVICE_NAME
# Default: srs-server
service_name srs-server;
# Whether enable logging for each log sending.
# Overwrite by env SRS_TENCENTCLOUD_APM_DEBUG_LOGGING
# Default: off
debug_logging off;
}
# 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;
}
#############################################################################################
# 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.
# 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;
}
#############################################################################################
# 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
# 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;
}
# 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://ossrs.net/lts/zh-cn/docs/v4/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/v4/doc/origin-cluster
# 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;
}
}
# 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;
# 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"
# ]
# }
# }
# PS: you can transform params to backend service, such as:
# { "param": "?forward=rtmp://127.0.0.1:19351/test/livestream" }
# then backend return forward's url in response.
# if backend return empty urls, destanition is still disabled.
# only support one api hook, format:
# backend http://xxx/api0
backend http://127.0.0.1:8085/api/v1/forward;
}
}
# 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.
# 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;
}
}
# 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 {
# 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;
Kickoff publisher when stream is idle, which means no players. v6.0.31, v5.0.144 (#3105)
For some use scenario, the publisher is invited when player want to view the stream:
1. Publisher connect to system, but does not publish any stream to SRS yet.
2. Player connect to system and start to request the stream.
3. System notifies publisher to publish stream to SRS.
4. Player play the stream from SRS.
Please notice that `system` means your business system, not SRS.
This is what we called `on-demand-live-streaming`, so when the last player stop to view the stream, what happends?
1. System needs to notify publisher to stop publish.
2. Or, SRS disconnect the publisher when idle(the last player stops playing).
This PR is for the solution 2, so that the cleanup is very simple, your system does not need to notify publisher to stop publish, because SRS has already disconnected the publihser.
---------
Co-authored-by: winlin <winlin@vip.126.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
2 years ago
# 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;
}
}
# 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;
}
}
# 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;
}
}
# 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;
}
}
# 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.
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"
# }
# 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.
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"
# }
# 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];
}
}
# 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.
# 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;
}
}
# 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;
# 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.
Solve the problem of inaccurate HLS TS duration. v5.0.187 v6.0.87 (#3824)
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
1 year ago
# default: 1.0
hls_td_ratio 1.0;
# the audio overflow ratio.
# for pure audio, the duration to reap the segment.
Solve the problem of inaccurate HLS TS duration. v5.0.187 v6.0.87 (#3824)
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
1 year ago
# 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.
Change the hls_aof_ratio to 2.1. v5.0.200 v6.0.101 (#3886)
In pure audio mode, there are no keyframes. Therefore, we can only rely
on the length of the slice to determine whether it should be output.
`hls_aof_ratio` is the coefficient that, once reached, will generate a
new slice.
In scenarios with video, if the `hls_aof_ratio` is too small, for
example 1.2, and the GOP (Group of Pictures) is 10 seconds, then a slice
will definitely be generated at 12 seconds. At this point, if there are
no keyframes, it will cause the next slice to start with a non-keyframe.
A safer coefficient is twice the GOP (Group of Pictures). This way, it
won't trigger incorrectly and prevent the individual transcoding of a ts
segment file.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
1 year ago
# 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;
# 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.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_TS_FILE for all vhosts.
# default: [app]/[stream]-[seq].ts
hls_ts_file [app]/[stream]-[seq].ts;
# 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;
# 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
# Overwrite by env SRS_VHOST_HLS_HLS_ACODEC for all vhosts.
# 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
# Overwrite by env SRS_VHOST_HLS_HLS_VCODEC for all vhosts.
# default: h264
hls_vcodec h264;
# 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.
Solve the problem of inaccurate HLS TS duration. v5.0.187 v6.0.87 (#3824)
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
1 year ago
# 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;
Solve the problem of inaccurate HLS TS duration. v5.0.187 v6.0.87 (#3824)
1. The comment on the ratio configuration says it can affect the slice
duration, but there is no effect after configuring it.
2. The default hls_td_ratio is 1.5, and after setting it to 1, the
duration is still slightly more than 10 seconds.
3. Even if the GOP is an integer, like 1 second, the slice is still a
non-integer, like 0.998 seconds, which seems a bit unreliable.
4. In the duration of the TS in the m3u8 file, it is one frame less than
the duration of the slice.
5. Set hls_dispose to 120s to dispose HLS files when no stream.
6. Use docker.conf for docker.
Before this patch:
```
#EXTINF:10.983, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
After this patch:
```
#EXTINF:10.000, no desc
livestream-0.ts?hls_ctx=3p095hq0
```
Note: If the fragment is set to 10 seconds, but the GOP size cannot be
divided by 10, such as not 1, 2, 5, or 10, then the duration of ts will
still be more than 10 seconds.
---------
Co-authored-by: john <hondaxiao@tencent.com>
1 year ago
# 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 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.
# Overwrite by env SRS_VHOST_HLS_HLS_KEY_URL for all vhosts.
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.
# 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.net/lts/zh-cn/docs/v4/doc/delivery-hls#http-callback
# @see https://ossrs.io/lts/en-us/docs/v4/doc/delivery-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.net/lts/zh-cn/docs/v4/doc/delivery-hls#on-hls-notify
# @see https://ossrs.io/lts/en-us/docs/v4/doc/delivery-hls#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
# Overwrite by env SRS_VHOST_HDS_ENABLED for all vhosts.
# default: off
enabled on;
# the hds fragment in seconds.
# Overwrite by env SRS_VHOST_HDS_HDS_FRAGMENT for all vhosts.
# default: 10
hds_fragment 10;
# the hds window in seconds, erase the segment when exceed the window.
# Overwrite by env SRS_VHOST_HDS_HDS_WINDOW for all vhosts.
# default: 60
hds_window 60;
# the path to store the hds files.
# Overwrite by env SRS_VHOST_HDS_HDS_PATH for all vhosts.
# 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
# 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.net/lts/zh-cn/docs/v4/doc/dvr#custom-path
# @see https://ossrs.io/lts/en-us/docs/v4/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.net/lts/zh-cn/docs/v4/doc/dvr#http-callback
# @see https://ossrs.io/lts/en-us/docs/v4/doc/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/livestream?vhost=[vhost];
}
}
}
# 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]/[stream]?vhost=[vhost];
}
}
}
# 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];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
}
}
# forward the input stream query string to output
vhost param.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]/[stream]_[engine][param];
}
}
}
# 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]/[stream]_[engine]?vhost=[vhost];
}
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]/[stream]_[engine]?vhost=[vhost];
}
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]/[stream]_[engine]?vhost=[vhost];
}
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]/[stream]_[engine]?vhost=[vhost];
}
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]/[stream]_[engine]?vhost=[vhost];
}
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]/[stream]_[engine]?vhost=[vhost];
}
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]/[stream]_[engine]?vhost=[vhost];
}
}
}
# 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;
}
}
}
#############################################################################################
# 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;
#############################################################################################
# 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;
}
}
#############################################################################################
# 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;
