- Created a new README_cross.md file with Docker build instructions. - Updated srs.conf to include logging configuration options. - Added docker-compose.yml to define the SRS SIP service with necessary ports and volume mappings. - Introduced config.yaml for general and GB28181-specific configurations. - Added initial srs.conf with settings for RTMP, HTTP API, and WebRTC support.
60 lines
1.4 KiB
Plaintext
60 lines
1.4 KiB
Plaintext
listen 1935;
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max_connections 1000;
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# For docker, please use docker logs to manage the logs of SRS.
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# See https://docs.docker.com/config/containers/logging/
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srs_log_tank console;
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daemon off;
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disable_daemon_for_docker off;
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http_api {
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enabled on;
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listen 1985;
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raw_api {
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enabled on;
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allow_reload on;
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}
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}
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http_server {
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enabled on;
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listen 8080;
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dir ./objs/nginx/html;
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}
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stream_caster {
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enabled on;
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caster gb28181;
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output rtmp://127.0.0.1/live/[stream];
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listen 9000;
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sip {
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enabled off;
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}
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}
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rtc_server {
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enabled on;
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listen 8000; # UDP port
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
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candidate $CANDIDATE;
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# Disable for Oryx.
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use_auto_detect_network_ip off;
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api_as_candidates off;
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}
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vhost __defaultVhost__ {
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http_remux {
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enabled on;
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mount [vhost]/[app]/[stream].flv;
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}
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rtc {
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enabled on;
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nack on;
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twcc on;
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stun_timeout 30;
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dtls_role passive;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtmp-to-rtc
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rtmp_to_rtc on;
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keep_bframe off;
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# @see https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#rtc-to-rtmp
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rtc_to_rtmp on;
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pli_for_rtmp 6.0;
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}
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} |