33b0a0fe7d
In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
72 lines
1.4 KiB
Plaintext
72 lines
1.4 KiB
Plaintext
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listen 1935;
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max_connections 1000;
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# Force to daemon and write logs to file.
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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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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 on;
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listen 5060;
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timeout 2.1;
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reinvite 1.2;
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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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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 8000;
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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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rtmp_to_rtc on;
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keep_bframe off;
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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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# Disabling 'guess_has_av' as it is not required for the current test setup.
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guess_has_av off;
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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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