Alessandro Ros
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20bb9b90cd
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support G711 tracks with multiple channels and custom sample rates (#2891)
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1 year ago |
|
Alessandro Ros
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7437ee7a09
|
update golangci-lint (#2868)
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1 year ago |
|
Alessandro Ros
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598fadc9fb
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api: add 'query' field to RTMP, RTSP, SRT and WebRTC clients (#2689) (#2844)
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1 year ago |
|
Alessandro Ros
|
11988249df
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move servers into internal/servers (#2792)
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1 year ago |
|
Alessandro Ros
|
ce45498769
|
move hooks into dedicated package (#2746)
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1 year ago |
|
Alessandro Ros
|
813611057d
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add runOnUnDemand hook (#2645)
|
2 years ago |
|
Alessandro Ros
|
43d41c070b
|
move static sources into dedicated package (#2616)
|
2 years ago |
|
Alessandro Ros
|
64eb90738a
|
webrtc: return detailed errors in responses (#2594)
|
2 years ago |
|
Alessandro Ros
|
99bc327d67
|
move protocol-related code into internal/protocols (#2572)
|
2 years ago |
|
Alessandro Ros
|
28452acf56
|
move webrtc utilities into internal/webrtc (#2559)
|
2 years ago |
|
Alessandro Ros
|
3a5bb06e26
|
add environment variable MTX_QUERY to some hooks (#2483) (#2522)
|
2 years ago |
|
Alessandro Ros
|
95ab9375c7
|
support recording to MPEG-TS (#2505)
|
2 years ago |
|
Alessandro Ros
|
c37fd38f5c
|
webrtc: print lost packets (#2468)
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2 years ago |
|
Alessandro Ros
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64d9060560
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add additional environment variables to custom commands (#1414) (#2356)
new variables: MTX_CONN_TYPE, MTX_CONN_ID, MTX_SOURCE_TYPE, MTX_SOURCE_ID, MTX_READER_TYPE, MTX_READ_ID
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2 years ago |
|
Alessandro Ros
|
ed77560811
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add runOnDisconnect, runOnNotReady, runOnUnread (#1464) (#2355)
|
2 years ago |
|
Alessandro Ros
|
f07886db5f
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print the reason why a source is started or stopped (#2322)
|
2 years ago |
|
Alessandro Ros
|
95baade478
|
srt: fix memory leak during reader disconnection (#2273)
|
2 years ago |
|
Alessandro Ros
|
5fb7f4e846
|
force all readers to use an asynchronous writer (#2265)
needed by #2255
|
2 years ago |
|
Alessandro Ros
|
b72f3577c8
|
print warning when the write queue is full (#2251)
|
2 years ago |
|
Alessandro Ros
|
cf86dbb303
|
switch to gortsplib/v4 (#2244)
|
2 years ago |
|
Alessandro Ros
|
bf8e69ea89
|
rename readBufferCount into writeQueueSize (#2248)
|
2 years ago |
|
Alessandro Ros
|
dd91abae9b
|
api: add transport to RTSP sessions (#2151)
|
2 years ago |
|
Alessandro Ros
|
bc3084ae7b
|
support proxying WebRTC streams (#2142)
|
2 years ago |
|
Alessandro Ros
|
72b1d233df
|
normalize channels and methods (#2127)
needed by #2068
|
2 years ago |
|
Alessandro Ros
|
e3d4856b4f
|
update gortsplib (#2126)
|
2 years ago |
|
Alessandro Ros
|
db3862cf0d
|
move stream in a dedicated package (#2121)
needed by #2068
|
2 years ago |
|
Alessandro Ros
|
b42154fa6a
|
return an error in case the random number generator fails (#2120)
|
2 years ago |
|
Alessandro Ros
|
9b491499bc
|
webrtc: speed up track detection (#2105)
|
2 years ago |
|
Alessandro Ros
|
1fa53b49d4
|
webrtc, hls: prevent brute-force attacks by waiting before sending responses (#2100)
|
2 years ago |
|
Alessandro Ros
|
0137734294
|
webrtc, hls: show IP in logs in case of failed authentication (#2099)
|
2 years ago |
|
Alessandro Ros
|
36298f8bc8
|
webrtc: send session ID to external auth server (#1981) (#2098)
|
2 years ago |
|
Alessandro Ros
|
af23609d47
|
api: fix crash when calling /v1/webrtcsessions/list just after session creation (#2097)
|
2 years ago |
|
Alessandro Ros
|
473c075d89
|
webrtc: fix memory leak during shutdown or session kick (#2079)
|
2 years ago |
|
Alessandro Ros
|
22b120ef22
|
update list of supported codecs inside error messages (#2058) (#2073)
|
2 years ago |
|
Alessandro Ros
|
5066ba403c
|
webrtc: fix race condition that caused random crashes during handshake (#2072)
|
2 years ago |
Volodymyr Borodin
|
47317ea8e5
|
api: add path to RTMP connections, RTSP sessions, WebRTC sessions (#1962) (#2022)
* api: add path to rtmp response
* add 'path' to RTSP and WebRTC sessions too
* add tests
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com>
|
2 years ago |
|
Alessandro Ros
|
1a748bb971
|
webrtc: allow using special characters in ICE server credentials (#1953) (#2000)
|
2 years ago |
|
Alessandro Ros
|
20a3b07d0a
|
webrtc: move codec and bitrate settings on client side (#1990)
|
2 years ago |
|
Alessandro Ros
|
79ee4e06f3
|
webrtc: add option to disable audio effects (#1908) (#1989)
|
2 years ago |
|
Alessandro Ros
|
4aef466103
|
webrtc: allow setting Opus bitrate (#1908) (#1985)
|
2 years ago |
|
Alessandro Ros
|
6663f7b474
|
webrtc: forbid publishing zero tracks (#1991)
|
2 years ago |
|
Alessandro Ros
|
fb1f8ff81d
|
webrtc: fix bitrate not being applied (#1984)
|
2 years ago |
|
Alessandro Ros
|
c46d2156b6
|
webrtc: fix memory leak when publishing or reading (#1884) (#1983)
|
2 years ago |
|
Alessandro Ros
|
99aa0d0ac9
|
webrtc: fix WHIP/WHEP implementation (#1857) (#1861)
offers and answers are now encoded in SDP in place of JSON; Location
header is set by the server.
This fixes compatibility with GStreamer and whipsink
|
2 years ago |
|
Alessandro Ros
|
b93eed64bc
|
api: add /get endpoints (#1577) (#1823)
this allows to get entities by ID or name after /list endpoints were
changed in v0.23.0.
|
2 years ago |
|
Alessandro Ros
|
503a131097
|
webrtc: return 404 when a stream is not present (#1805)
|
2 years ago |
|
Alessandro Ros
|
39c072edd6
|
change repository owner (#1801)
|
2 years ago |
|
Alessandro Ros
|
a14246d776
|
webrtc: support publishing with WHIP and reading with WHEP (#1800)
|
2 years ago |
