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
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
1 year ago
1d4ea2cd9a
hls: fix freeze in case of muxing errors ( #3135 ) ( #3150 )
1 year ago
9c6ba7e2c7
New authentication system ( #1341 ) ( #1992 ) ( #2205 ) ( #3081 )
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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).
1 year ago
34dbcfb508
move WebRTC tests into internal/servers/webrtc ( #3043 )
1 year ago
ba69241377
hls: stop spamming 'stream doesn't contain any supported codec' when hlsAlwaysRemux is true ( #3018 )
1 year ago
1ae3240b91
hls: fix crash when muxer is being recreated, improve performance ( #3017 )
1 year ago
dd7d7c6c5d
srt: wait some seconds before returning authentication errors ( #2918 )
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this allows to mitigate brute force attacks and is possible thanks to
https://github.com/datarhei/gosrt/pull/43
1 year ago
514036d41a
treat different RTSP formats as different tracks in logs and API ( #2907 )
1 year ago
20bb9b90cd
support G711 tracks with multiple channels and custom sample rates ( #2891 )
1 year ago
7437ee7a09
update golangci-lint ( #2868 )
1 year ago
598fadc9fb
api: add 'query' field to RTMP, RTSP, SRT and WebRTC clients ( #2689 ) ( #2844 )
1 year ago
11988249df
move servers into internal/servers ( #2792 )
2 years ago
ce45498769
move hooks into dedicated package ( #2746 )
2 years ago
813611057d
add runOnUnDemand hook ( #2645 )
2 years ago
43d41c070b
move static sources into dedicated package ( #2616 )
2 years ago
64eb90738a
webrtc: return detailed errors in responses ( #2594 )
2 years ago
99bc327d67
move protocol-related code into internal/protocols ( #2572 )
2 years ago
28452acf56
move webrtc utilities into internal/webrtc ( #2559 )
2 years ago
3a5bb06e26
add environment variable MTX_QUERY to some hooks ( #2483 ) ( #2522 )
2 years ago
95ab9375c7
support recording to MPEG-TS ( #2505 )
2 years ago
c37fd38f5c
webrtc: print lost packets ( #2468 )
2 years ago
64d9060560
add additional environment variables to custom commands ( #1414 ) ( #2356 )
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new variables: MTX_CONN_TYPE, MTX_CONN_ID, MTX_SOURCE_TYPE, MTX_SOURCE_ID, MTX_READER_TYPE, MTX_READ_ID
2 years ago
ed77560811
add runOnDisconnect, runOnNotReady, runOnUnread ( #1464 ) ( #2355 )
2 years ago
f07886db5f
print the reason why a source is started or stopped ( #2322 )
2 years ago
95baade478
srt: fix memory leak during reader disconnection ( #2273 )
2 years ago
5fb7f4e846
force all readers to use an asynchronous writer ( #2265 )
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needed by #2255
2 years ago
b72f3577c8
print warning when the write queue is full ( #2251 )
2 years ago
cf86dbb303
switch to gortsplib/v4 ( #2244 )
2 years ago
bf8e69ea89
rename readBufferCount into writeQueueSize ( #2248 )
2 years ago
dd91abae9b
api: add transport to RTSP sessions ( #2151 )
2 years ago
bc3084ae7b
support proxying WebRTC streams ( #2142 )
2 years ago
72b1d233df
normalize channels and methods ( #2127 )
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needed by #2068
2 years ago
e3d4856b4f
update gortsplib ( #2126 )
2 years ago
db3862cf0d
move stream in a dedicated package ( #2121 )
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needed by #2068
2 years ago
b42154fa6a
return an error in case the random number generator fails ( #2120 )
2 years ago
9b491499bc
webrtc: speed up track detection ( #2105 )
2 years ago
1fa53b49d4
webrtc, hls: prevent brute-force attacks by waiting before sending responses ( #2100 )
2 years ago
0137734294
webrtc, hls: show IP in logs in case of failed authentication ( #2099 )
2 years ago
36298f8bc8
webrtc: send session ID to external auth server ( #1981 ) ( #2098 )
2 years ago
af23609d47
api: fix crash when calling /v1/webrtcsessions/list just after session creation ( #2097 )
2 years ago
473c075d89
webrtc: fix memory leak during shutdown or session kick ( #2079 )
2 years ago
22b120ef22
update list of supported codecs inside error messages ( #2058 ) ( #2073 )
2 years ago
5066ba403c
webrtc: fix race condition that caused random crashes during handshake ( #2072 )
2 years ago
47317ea8e5
api: add path to RTMP connections, RTSP sessions, WebRTC sessions ( #1962 ) ( #2022 )
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* api: add path to rtmp response
* add 'path' to RTSP and WebRTC sessions too
* add tests
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
2 years ago
1a748bb971
webrtc: allow using special characters in ICE server credentials ( #1953 ) ( #2000 )
2 years ago
20a3b07d0a
webrtc: move codec and bitrate settings on client side ( #1990 )
2 years ago
79ee4e06f3
webrtc: add option to disable audio effects ( #1908 ) ( #1989 )
2 years ago
4aef466103
webrtc: allow setting Opus bitrate ( #1908 ) ( #1985 )
2 years ago
6663f7b474
webrtc: forbid publishing zero tracks ( #1991 )
2 years ago
fb1f8ff81d
webrtc: fix bitrate not being applied ( #1984 )
2 years ago
Dan Bason
Alessandro Ros
Volodymyr Borodin