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GB28181: Support external SIP server. v6.0.144 (#4101)

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For #3369 to support an external powerful SIP server, do not use the
embedded SIP server of SRS.
For more information, detailed steps, system architecture, and
background explanation, please see
https://ossrs.net/lts/zh-cn/docs/v6/doc/gb28181#external-sip

---------

Co-authored-by: Jacob Su <suzp1984@gmail.com>
Co-authored-by: winlin <winlinvip@gmail.com>
This commit is contained in:
Haibo Chen 2024-07-27 09:06:12 +08:00 • committed by GitHub
parent f76be5fe9b
commit 65ad907fe4
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6 changed files with 238 additions and 9 deletions
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6
trunk/conf/full.conf
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@ -676,9 +676,11 @@ stream_caster {
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.
# [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.
# 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;
# The SIP listen port, for TCP protocol.

53
trunk/conf/gb28181-without-sip.conf Normal file
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@ -0,0 +1,53 @@
listen 1935;
max_connections 1000;
daemon off;
srs_log_tank console;
stream_caster {
enabled on;
caster gb28181;
output rtmp://127.0.0.1/live/[stream];
listen 9000;
sip {
enabled off;
}
}
http_server {
enabled on;
listen 8080;
dir ./objs/nginx/html;
}
http_api {
enabled on;
listen 1985;
}
stats {
network 0;
}
rtc_server {
enabled on;
listen 8000; # UDP port
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://ossrs.net/lts/zh-cn/docs/v4/doc/webrtc#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
hls {
enabled on;
}
}