Files
srs/trunk/conf/push.gb28181.conf
Winlin d9fe2c458c AI: GB28181: Remove embedded SIP server and enforce external SIP usage. v7.0.75 (#4466)
This PR removes the embedded GB28181 SIP server implementation from SRS
and enforces the use of external SIP servers for production deployments.

The embedded SIP server depended on the deprecated `http-parser`
library. With the planned migration to `llhttp` (which doesn't support
SIP parsing), maintaining the embedded SIP server would require
significant additional work. Since external SIP servers are already the
recommended approach for production, removing the embedded
implementation simplifies the codebase and eliminates this dependency.

Eliminated `srs_gb28181_test` from CI workflow.

Removed SIP configuration validation tests.

---------

Co-authored-by: OSSRS-AI <winlinam@gmail.com>
Co-authored-by: haibo.chen <495810242@qq.com>
2025-09-02 09:59:40 -04:00

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max_connections 1000;
daemon off;
srs_log_tank console;
# RTMP server configuration
rtmp {
listen 1935;
}
stream_caster {
enabled on;
caster gb28181;
output rtmp://127.0.0.1/live/[stream];
listen 9000;
# Deprecated config, moved to sip.candidate, see gb28181.conf
# @see https://ossrs.io/lts/en-us/docs/v7/doc/gb28181#config-candidate
host $CANDIDATE;
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.io/lts/en-us/docs/v7/doc/webrtc#config-candidate
candidate $CANDIDATE;
}
vhost __defaultVhost__ {
rtc {
enabled on;
# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#rtmp-to-rtc
rtmp_to_rtc on;
# @see https://ossrs.io/lts/en-us/docs/v7/doc/webrtc#rtc-to-rtmp
rtc_to_rtmp on;
}
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
}
hls {
enabled on;
}
}