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This commit is contained in:
winlin 2022-01-11 08:39:31 +08:00
commit cce6c7bf13
34 changed files with 427 additions and 97 deletions

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@ -77,6 +77,10 @@ asprocess off;
# 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.
# 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.
@ -392,6 +396,11 @@ rtc_server {
# @see https://github.com/ossrs/srs/wiki/v4_CN_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;
# The IP family filter for auto discover candidate, it can be:
# ipv4 Filter IP v4 candidates.
# ipv6 Filter IP v6 candidates.

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@ -0,0 +1,55 @@
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
https {
enabled on;
listen 8088;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
http_api {
enabled on;
listen 1985;
https {
enabled on;
listen 1990;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
stats {
network 0;
}
rtc_server {
enabled on;
listen 8000; # UDP port
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
hls {
enabled on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
}

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@ -31,20 +31,22 @@ stats {
}
rtc_server {
enabled on;
# Listen at udp://8000
listen 8000;
#
# The $CANDIDATE means fetch from env, if not configed, use * as default.
#
# The * means retrieving server IP automatically, from all network interfaces,
# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
listen 8000; # UDP port
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
keep_bframe off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}

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@ -0,0 +1,52 @@
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
https {
enabled on;
listen 8088;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
http_api {
enabled on;
listen 1985;
https {
enabled on;
listen 1990;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
stats {
network 0;
}
rtc_server {
enabled on;
listen 8000; # UDP port
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
}

55
trunk/conf/https.srs.conf Normal file
View file

@ -0,0 +1,55 @@
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
https {
enabled on;
listen 8088;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
http_api {
enabled on;
listen 1985;
https {
enabled on;
listen 1990;
key ./conf/server.key;
cert ./conf/server.crt;
}
}
stats {
network 0;
}
rtc_server {
enabled on;
listen 8000; # UDP port
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
hls {
enabled on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
rtc {
enabled on;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
rtmp_to_rtc off;
# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
rtc_to_rtmp off;
}
}