mirror of
https://github.com/ossrs/srs.git
synced 2025-02-13 11:51:57 +00:00
ea8cff6163
* fix annotation spell failed * RTC to RTMP using SenderReport to sync av timestamp * update pion/webrtc versio from v3.0.4 -> v3.0.13, auto config sender/receiver report * Add rtc push flv play regression test * Add unit test of ntp and av sync time * Take flag CXX to makefile of utest * Add annotation about rtc unit test * Fix compiler error in C++98 * Add FFmpeg log callback funciton.
59 lines
1.3 KiB
Text
59 lines
1.3 KiB
Text
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listen 1935;
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max_connections 1000;
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daemon on;
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disable_daemon_for_docker off;
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srs_log_tank file;
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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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http_api {
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enabled on;
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listen 1985;
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}
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stats {
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network 0;
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}
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rtc_server {
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enabled on;
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# Listen at udp://8000
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listen 8000;
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#
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# The $CANDIDATE means fetch from env, if not configed, use * as default.
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#
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# The * means retrieving server IP automatically, from all network interfaces,
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# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
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candidate $CANDIDATE;
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}
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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rtc_to_rtmp on;
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}
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play {
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atc on;
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}
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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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ingest livestream {
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enabled on;
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input {
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type file;
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url ./doc/source.200kbps.768x320.flv;
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}
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ffmpeg ./objs/ffmpeg/bin/ffmpeg;
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engine {
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enabled off;
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output rtmp://127.0.0.1:[port]/live/livestream;
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}
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}
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}
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