Alessandro Ros
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b72f3577c8
|
print warning when the write queue is full (#2251)
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2 years ago |
|
Alessandro Ros
|
cf86dbb303
|
switch to gortsplib/v4 (#2244)
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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)
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2 years ago |
|
Alessandro Ros
|
72b1d233df
|
normalize channels and methods (#2127)
needed by #2068
|
2 years ago |
|
Alessandro Ros
|
e3d4856b4f
|
update gortsplib (#2126)
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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 |
