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Config: Refine config file, nothing changed.

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winlin 2022-09-02 22:11:41 +08:00
parent 77e1f81f91
commit 38c3d9f1a7
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trunk/conf/full.conf
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@ -1,28 +1,19 @@
# all config for srs
# 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
#############################################################################################
# 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;
#############################################################################################
# RTMP sections
#############################################################################################
# the rtmp listen ports, split by space, each listen entry is <[ip:]port>
# for example, 192.168.1.100:1935 10.10.10.100:1935
# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
listen 1935;
# The pid file to 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+
# Default: ./objs/srs.pid
pid ./objs/srs.pid;
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# however, most clients support it and it can improve
# performance about 10%.
# default: 60000
chunk_size 60000;
# the log dir for FFMPEG.
# if enabled ffmpeg, each transcoding stream will create a log file.
# /dev/null to disable the log.
@ -76,7 +67,7 @@ pithy_print_ms 10000;
# default: ./
work_dir ./;
# whether quit when parent process changed,
# used for supervisor mode(not daemon), srs should always quit when
# 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.
@ -119,6 +110,374 @@ inotify_auto_reload off;
# 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
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%.
# 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.
# 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.
# Default: 1985
listen 8080;
# whether enable crossdomain request.
# default: on
crossdomain on;
# the HTTP RAW API is more powerful api to change srs state and reload.
raw_api {
# whether enable the HTTP RAW API.
# default: off
enabled off;
# whether enable rpc reload.
# default: off
allow_reload off;
# whether enable rpc query.
# 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;
}
# For https_api or HTTPS API.
https {
# Whether enable HTTPS API.
# 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.
# Default: 1990
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# 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"
# 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.
# default: off
enabled on;
# the http streaming listen entry is <[ip:]port>
# for example, 192.168.1.100:8080
# where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
# @remark, if use lower port, for instance 80, user must start srs by root.
# default: 8080
listen 8080;
# the default dir for http root.
# default: ./objs/nginx/html
dir ./objs/nginx/html;
# whether enable crossdomain request.
# for both http static and stream server and apply on all vhosts.
# default: on
crossdomain on;
# For https_server or HTTPS Streaming.
https {
# Whether enable HTTPS Streaming.
# default: off
enabled on;
# The listen endpoint for HTTPS Streaming.
# default: 8088
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# 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"
# 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.
# default: off
enabled on;
# The UDP listen port for SRT.
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.
# default: -1
maxbw 1000000000;
# 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.
# 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.
# default: 10000
peer_idle_timeout 8000;
# Default app for vmix, see https://github.com/ossrs/srs/pull/1615
# default: live
default_app live;
# The peerlatency is set by the sender side and will notify the receiver side.
# default: 0
peerlatency 0;
# The recvlatency means latency from sender to receiver.
# default: 120
recvlatency 0;
# This latency configuration configures both recvlatency and peerlatency to the same value.
# 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.
# 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.
# default: on
tlpktdrop off;
# The send buffer size of SRT.
# default: 8192 * (1500-28)
sendbuf 2000000;
# The recv buffer size of SRT.
# default: 8192 * (1500-28)
recvbuf 2000000;
}
vhost srt.vhost.srs.com {
srt {
enabled on;
srt_to_rtmp on;
}
}
#############################################################################################
# WebRTC server section
#############################################################################################
rtc_server {
# Whether enable WebRTC server.
# default: off
enabled on;
# The udp listen port, we will reuse it for connections.
# default: 8000
listen 8000;
# 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.
# @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
# 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.
# 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.
# 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.
# 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.
# 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
# Default: off
keep_api_domain off;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# default: on
ecdsa on;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# default: on
encrypt on;
# We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
# Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
# and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
# default: 1
reuseport 1;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# 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.
# default: off
enabled off;
# The black-hole address for session.
addr 127.0.0.1:10000;
}
}
vhost rtc.vhost.srs.com {
rtc {
# Whether enable WebRTC server.
# default: off
enabled on;
# Whether support NACK.
# default: on
nack on;
# Whether directly use the packet, avoid copy.
# default: on
nack_no_copy on;
# Whether support TWCC.
# 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.
# default: 30
stun_timeout 30;
# The strict check when process stun.
# default: off
stun_strict_check on;
# The role of dtls when peer is actpass: passive or active
# default: passive
dtls_role passive;
# The version of dtls, support dtls1.0, dtls1.2, and auto
# default: auto
dtls_version auto;
# Drop the packet with the pt(payload type), 0 never drop.
# default: 0
drop_for_pt 0;
###############################################################
# Whether enable transmuxing RTMP to RTC.
# If enabled, transcode aac to opus.
# default: off
rtmp_to_rtc off;
# Whether keep B-frame, which is normal feature in live streaming,
# but usually disabled in RTC.
# default: off
keep_bframe off;
###############################################################
# Whether enable transmuxing RTC to RTMP.
# Default: off
rtc_to_rtmp off;
# The PLI interval in seconds, for RTC to RTMP.
# Note the available range is [0.5, 30]
# Default: 6.0
pli_for_rtmp 6.0;
}
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.
# Whether enable min delay mode for vhost.
# default: on, for RTC.
min_latency on;
play {
# set the MW(merged-write) latency in ms.
# @remark For WebRTC, we enable pass-by-timestamp mode, so we ignore this config.
# default: 0 (For WebRTC)
mw_latency 0;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, that is min_latency on)
# default: 1 (For WebRTC, that is min_latency off)
mw_msgs 0;
}
}
#############################################################################################
# Streamer sections
#############################################################################################
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
# MPEGTS over UDP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# mpegts_over_udp, MPEG-TS over UDP caster.
caster mpegts_over_udp;
# the output rtmp url.
# for mpegts_over_udp caster, the typically output url:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
listen 8935;
}
# FLV
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# flv, FLV over HTTP by POST.
caster flv;
# the output rtmp url.
# for flv caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, POST to url:
# http://127.0.0.1:8936/live/livestream.flv
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/[app]/[stream];
# the listen port for stream caster.
# for flv caster, listen at tcp port. for example, 8936.
listen 8936;
}
#############################################################################################
# other global sections
#############################################################################################
# For tcmalloc, the release rate.
# @see https://gperftools.github.io/gperftools/tcmalloc.html
# @remark Should run configure --with-gperf
@ -278,356 +637,6 @@ stats {
disk sda sdb xvda xvdb;
}
#############################################################################################
# HTTP sections
#############################################################################################
# api of srs.
# the http api config, export for external program to manage srs.
# user can access http api of srs in browser directly, for instance, to access by:
# curl http://192.168.1.170:1985/api/v1/reload
# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
# where the cli can only be used in shell/terminate.
http_api {
# whether http api is enabled.
# default: off
enabled on;
# The http api listen entry is <[ip:]port>, For example, 192.168.1.100:8080, 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.
# Default: 1985
listen 8080;
# whether enable crossdomain request.
# default: on
crossdomain on;
# the HTTP RAW API is more powerful api to change srs state and reload.
raw_api {
# whether enable the HTTP RAW API.
# default: off
enabled off;
# whether enable rpc reload.
# default: off
allow_reload off;
# whether enable rpc query.
# 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;
}
# For https_api or HTTPS API.
https {
# Whether enable HTTPS API.
# 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.
# Default: 1990
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# 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"
# 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.
# default: off
enabled on;
# the http streaming listen entry is <[ip:]port>
# for example, 192.168.1.100:8080
# where the ip is optional, default to 0.0.0.0, that is 8080 equals to 0.0.0.0:8080
# @remark, if use lower port, for instance 80, user must start srs by root.
# default: 8080
listen 8080;
# the default dir for http root.
# default: ./objs/nginx/html
dir ./objs/nginx/html;
# whether enable crossdomain request.
# for both http static and stream server and apply on all vhosts.
# default: on
crossdomain on;
# For https_server or HTTPS Streaming.
https {
# Whether enable HTTPS Streaming.
# default: off
enabled on;
# The listen endpoint for HTTPS Streaming.
# default: 8088
listen 8088;
# The SSL private key file, generated by:
# openssl genrsa -out server.key 2048
# 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"
# default: ./conf/server.crt
cert ./conf/server.crt;
}
}
#############################################################################################
# Streamer sections
#############################################################################################
# the streamer cast stream from other protocol to SRS over RTMP.
# @see https://github.com/ossrs/srs/tree/develop#stream-architecture
# MPEGTS over UDP
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# mpegts_over_udp, MPEG-TS over UDP caster.
caster mpegts_over_udp;
# the output rtmp url.
# for mpegts_over_udp caster, the typically output url:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/live/livestream;
# the listen port for stream caster.
# for mpegts_over_udp caster, listen at udp port. for example, 8935.
listen 8935;
}
# FLV
stream_caster {
# whether stream caster is enabled.
# default: off
enabled on;
# the caster type of stream, the casters:
# flv, FLV over HTTP by POST.
caster flv;
# the output rtmp url.
# for flv caster, the typically output url:
# rtmp://127.0.0.1/[app]/[stream]
# for example, POST to url:
# http://127.0.0.1:8936/live/livestream.flv
# where the [app] is "live" and [stream] is "livestream", output is:
# rtmp://127.0.0.1/live/livestream
output rtmp://127.0.0.1/[app]/[stream];
# the listen port for stream caster.
# for flv caster, listen at tcp port. for example, 8936.
listen 8936;
}
#############################################################################################
# SRT server section
#############################################################################################
# @doc https://github.com/ossrs/srs/issues/1147#usage
srt_server {
# whether SRT server is enabled.
# default: off
enabled on;
# The UDP listen port for SRT.
listen 10080;
# For detail parameters, please read wiki:
# @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.
# default: -1
maxbw 1000000000;
# 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.
# 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.
# default: 10000
peer_idle_timeout 8000;
# Default app for vmix, see https://github.com/ossrs/srs/pull/1615
# default: live
default_app live;
# The peerlatency is set by the sender side and will notify the receiver side.
# default: 0
peerlatency 0;
# The recvlatency means latency from sender to receiver.
# default: 120
recvlatency 0;
# This latency configuration configures both recvlatency and peerlatency to the same value.
# 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.
# 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.
# default: on
tlpktdrop off;
# The send buffer size of SRT.
# default: 8192 * (1500-28)
sendbuf 2000000;
# The recv buffer size of SRT.
# default: 8192 * (1500-28)
recvbuf 2000000;
}
vhost srt.vhost.srs.com {
srt {
enabled on;
srt_to_rtmp on;
}
}
#############################################################################################
# WebRTC server section
#############################################################################################
rtc_server {
# Whether enable WebRTC server.
# default: off
enabled on;
# The udp listen port, we will reuse it for connections.
# default: 8000
listen 8000;
# 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.
# @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
# default: *
candidate *;
# 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.
# 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.
# 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
# Default: off
keep_api_domain off;
# Whether use network interface IP which is detected automatically, filtered by ip_family.
# Note that browser might fail if no CANDIDATE specified.
# 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.
# default: ipv4
ip_family ipv4;
# Whether use ECDSA certificate.
# If not, use RSA certificate.
# default: on
ecdsa on;
# Whether encrypt RTP packet by SRTP.
# @remark Should always turn it on, or Chrome will fail.
# default: on
encrypt on;
# We listen multiple times at the same port, by REUSEPORT, to increase the UDP queue.
# Note that you can set to 1 and increase the system UDP buffer size by net.core.rmem_max
# and net.core.rmem_default or just increase this to get larger UDP recv and send buffer.
# default: 1
reuseport 1;
# Whether merge multiple NALUs into one.
# @see https://github.com/ossrs/srs/issues/307#issuecomment-612806318
# 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.
# default: off
enabled off;
# The black-hole address for session.
addr 127.0.0.1:10000;
}
}
vhost rtc.vhost.srs.com {
rtc {
# Whether enable WebRTC server.
# default: off
enabled on;
# Whether support NACK.
# default: on
nack on;
# Whether directly use the packet, avoid copy.
# default: on
nack_no_copy on;
# Whether support TWCC.
# 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.
# default: 30
stun_timeout 30;
# The strict check when process stun.
# default: off
stun_strict_check on;
# The role of dtls when peer is actpass: passive or active
# default: passive
dtls_role passive;
# The version of dtls, support dtls1.0, dtls1.2, and auto
# default: auto
dtls_version auto;
# Drop the packet with the pt(payload type), 0 never drop.
# default: 0
drop_for_pt 0;
###############################################################
# Whether enable transmuxing RTMP to RTC.
# If enabled, transcode aac to opus.
# default: off
rtmp_to_rtc off;
# Whether keep B-frame, which is normal feature in live streaming,
# but usually disabled in RTC.
# default: off
keep_bframe off;
###############################################################
# Whether enable transmuxing RTC to RTMP.
# Default: off
rtc_to_rtmp off;
# The PLI interval in seconds, for RTC to RTMP.
# Note the available range is [0.5, 30]
# Default: 6.0
pli_for_rtmp 6.0;
}
###############################################################
# For transmuxing RTMP to RTC, it will impact the default values if RTC is on.
# Whether enable min delay mode for vhost.
# default: on, for RTC.
min_latency on;
play {
# set the MW(merged-write) latency in ms.
# @remark For WebRTC, we enable pass-by-timestamp mode, so we ignore this config.
# default: 0 (For WebRTC)
mw_latency 0;
# Set the MW(merged-write) min messages.
# default: 0 (For Real-Time, that is min_latency on)
# default: 1 (For WebRTC, that is min_latency off)
mw_msgs 0;
}
}
#############################################################################################
# RTMP/HTTP VHOST sections
#############################################################################################