Jason Walton
1c2f95f609
webrtc: allow configuring timeouts ( #3404 ) ( #3406 )
...
* webrtc: allow configuring timeouts (#3404 )
* fix from code inspect
2024-05-30 13:36:58 +02:00
Alessandro Ros
407702380a
webrtc: in answer, include codecs that are actually in use ( #3374 )
2024-05-19 19:41:42 +02:00
Alessandro Ros
e283725cde
support routing multichannel Opus between RTSP, SRT, HLS, UDP and recording in MPEG-TS and fMP4 ( #3355 ) ( #3368 )
2024-05-19 14:38:57 +02:00
Alessandro Ros
d21506182b
webrtc: fix returning code 404 when a stream does not exist ( #3369 )
2024-05-19 13:46:47 +02:00
Dan Bason
87c0535823
Add option for ICE servers to be client only ( #3164 )
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* Add option for ICE servers to be client only
* add clientOnly to configuration file and API docs
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2024-04-06 18:32:53 +02:00
Alessandro Ros
1d4ea2cd9a
hls: fix freeze in case of muxing errors ( #3135 ) ( #3150 )
2024-03-19 14:01:14 +01:00
Alessandro Ros
9c6ba7e2c7
New authentication system ( #1341 ) ( #1992 ) ( #2205 ) ( #3081 )
...
This is a new authentication system that covers all the features exposed by the server, including playback, API, metrics and PPROF, improves internal authentication by adding permissions, improves HTTP-based authentication by adding the ability to exclude certain actions from being authenticated, adds an additional method (JWT-based authentication).
2024-03-04 14:20:34 +01:00
Alessandro Ros
34dbcfb508
move WebRTC tests into internal/servers/webrtc ( #3043 )
2024-02-18 22:15:08 +01:00
Alessandro Ros
ba69241377
hls: stop spamming 'stream doesn't contain any supported codec' when hlsAlwaysRemux is true ( #3018 )
2024-02-13 23:36:40 +01:00
Alessandro Ros
1ae3240b91
hls: fix crash when muxer is being recreated, improve performance ( #3017 )
2024-02-13 23:32:15 +01:00
Alessandro Ros
dd7d7c6c5d
srt: wait some seconds before returning authentication errors ( #2918 )
...
this allows to mitigate brute force attacks and is possible thanks to
https://github.com/datarhei/gosrt/pull/43
2024-01-18 22:48:25 +01:00
Alessandro Ros
514036d41a
treat different RTSP formats as different tracks in logs and API ( #2907 )
2024-01-15 12:08:14 +01:00
Alessandro Ros
20bb9b90cd
support G711 tracks with multiple channels and custom sample rates ( #2891 )
2024-01-13 11:40:26 +01:00
Alessandro Ros
7437ee7a09
update golangci-lint ( #2868 )
2024-01-03 21:13:20 +01:00
Alessandro Ros
598fadc9fb
api: add 'query' field to RTMP, RTSP, SRT and WebRTC clients ( #2689 ) ( #2844 )
2023-12-26 13:59:53 +01:00
Alessandro Ros
11988249df
move servers into internal/servers ( #2792 )
2023-12-08 19:17:17 +01:00
