Alessandro Ros and GitHub
d1fd3df27c
api: add user field to RTSP, RTMP, SRT, WebRTC conns and sessions ( #5104 ) ( #5565 )
2026-03-14 00:03:58 +01:00
Alessandro Ros and GitHub
35e1f486c9
add integrated packet dumper ( #5488 )
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this allows to dump any incoming and outgoing packet, to disk, in
pcapng format.
2026-02-22 13:34:05 +01:00
Alessandro Ros and GitHub
a56408db19
add always available streams ( #5335 )
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When the publisher or source of a stream is offline, the server can be
configured to fill gaps in the stream with a video that is played on
repeat until a publisher comes back online. This allows readers to stay
connected regardless of the state of the stream. The offline video and
any future online stream are concatenated without decoding or
re-encoding packets, using the original codec.
2026-01-31 14:44:58 +01:00
Alessandro Ros and GitHub
46b4293e35
api: improve openapi definition ( #5380 )
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add uint64, uint, uuid, move WebRTCICEServer into dedicated schema
2026-01-25 19:35:43 +01:00
Alessandro Ros and GitHub
ff187b6d8a
update golangci-lint configuration ( #5182 )
2025-11-11 23:57:52 +01:00
Alessandro Ros and GitHub
1ef5db5b3c
extend udpReadBufferSize to WebRTC UDP sockets ( #5130 )
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this allows to decrease packet losses without touching system parameters.
2025-10-29 11:45:15 +01:00
Alessandro Ros and GitHub
adc4a6ceb6
add udpReadBufferSize parameter ( #5129 )
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this allows to set a global UDP read buffer, applied to every UDP socket.
2025-10-29 11:28:22 +01:00
Alessandro Ros and GitHub
f81c50ee68
rtsp: support reading streams tunneled with HTTP or WebSocket ( #4986 )
2025-09-17 22:31:20 +02:00
Alessandro Ros and GitHub
d0430d8ea5
support ingesting RTP streams ( #1515 ) ( #4843 )
2025-08-09 16:12:10 +02:00
Alessandro Ros and GitHub
7feff1d1dc
support MPEG-TS over unix sockets ( #4388 ) ( #4389 ) ( #4828 )
2025-08-08 18:03:38 +02:00
Alessandro Ros and GitHub
b59c37ad4b
remove redundant definitions ( #4793 )
2025-07-27 12:01:52 +02:00
Alessandro Ros and GitHub
1083eea307
make RTP packet size compatible with RTSP/SRTP ( #4692 )
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when RTSP encryption is enabled, maximum RTP packet size is slightly
decreased to make room for SRTP.
2025-07-05 15:42:58 +02:00
Alessandro Ros and GitHub
bf9cd328a9
rpi: fix RTP packet size ( #4691 )
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UDP max payload size was not applied.
2025-07-05 15:31:28 +02:00
Alessandro Ros and GitHub
8ce49727d6
rpi: support exposing a secondary stream from the same camera ( #4426 )
2025-04-14 11:56:08 +02:00
Alessandro Ros and GitHub
14f1daa02a
move static source handler into dedicated package ( #4427 )
2025-04-13 21:35:12 +02:00
