Alessandro Ros
3381b196fb
solve codec labels internally ( #5603 )
...
stop using gortsplib format.Format.Codec() for getting codec names.
2026-03-21 18:22:49 +01:00
Alessandro Ros
2b302e7940
api: provide track details ( #5307 ) ( #5333 ) ( #1726 ) ( #5585 )
2026-03-18 00:01:43 +01:00
eh
f98c9c59ca
rtsp: support unwrapping MPEG-TS tracks ( #5476 )
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this allows to use MPEG-TS tracks with other protocols and with the recording system.
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-03-17 19:14:04 +01:00
Alessandro Ros
c8e07b55d5
api: add new stats ( #5582 )
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- RTSPSession.outboundRTPPacketsDiscarded
- Path.inboundFramesInError
- SRTConn.outboundFramesDiscarded
- WebRTCSession.outboundFramesDiscarded
- RTMPConn.outboundFramesDiscarded
- HLSMuxer.outboundFramesDiscarded
2026-03-17 17:22:15 +01:00
Alessandro Ros
adf45596a3
api: rename WebRTC stats to match RTSP ones ( #5581 )
2026-03-16 23:29:25 +01:00
Alessandro Ros
dfce20fefa
api: add RTSP reported lost packets ( #5198 ) ( #5579 )
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The new outboundRtpPacketsReportedLost property allows to track RTP
packets that have been reported lost by readers. Furthermore, stats now
have a "inbound" or "outbound" prefix to improve readability.
2026-03-16 22:10:18 +01:00
Alessandro Ros
3bad7045c1
api: add missing enums and move all enums in dedicated components ( #5576 )
2026-03-15 22:09:00 +01:00
Alessandro Ros
c6bb332664
api: add deprecated fields to the OpenAPI definition ( #5575 )
2026-03-15 19:28:06 +01:00
Alessandro Ros
d1fd3df27c
api: add user field to RTSP, RTMP, SRT, WebRTC conns and sessions ( #5104 ) ( #5565 )
2026-03-14 00:03:58 +01:00
Roman Sirokov
9b36d50b8d
optionally validate JWT iss and aud claims ( #5569 )
2026-03-13 22:38:40 +01:00
Alessandro Ros
549300cbd4
prevent using alwaysAvailableFile and alwaysAvailableTracks together ( #5529 )
2026-02-28 18:31:41 +01:00
Alessandro Ros
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
87fbfebf06
webrtc: add whepBearerTokenParameter ( #3796 ) ( #5486 )
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this allows to pass Authorization: Bearer to servers that require it.
2026-02-19 11:57:39 +01:00
Alessandro Ros
f991821a5f
webrtc: allow tuning WHEP timeouts ( #5027 ) ( #5479 )
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add whepHandshakeTimeout, whepTrackGatherTimeout, whepSTUNGatherTimeout
2026-02-18 16:29:25 +01:00
Alessandro Ros
832cc05f39
api: add conns to RTSP Sessions ( #5454 )
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this allows to find connections associated with a RTSP session.
2026-02-12 15:45:34 +01:00
Alessandro Ros
c250e34075
support X-Forwarded-Proto in playback server ( #4970 ) ( #5445 )
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allow reverse proxies to change the schema of URLs returned by the
server through the X-Forwarded-Proto header.
2026-02-11 21:47:02 +01:00
Alessandro Ros
4f859fb40b
add authHTTPFingerprint ( #5413 ) ( #5422 )
2026-02-05 16:51:29 +01:00
Alessandro Ros
74eaa11d3a
rtsp: add rtspUDPSourcePortRange param ( #5363 ) ( #5397 )
2026-01-31 16:21:53 +01:00
Alessandro Ros
92f9ee7b78
deprecate fallback ( #5388 )
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The fallback feature worked with RTSP only and did not allow readers to
resume the original stream. It has been replaced by alwaysAvailable.
2026-01-31 15:29:24 +01:00
Alessandro Ros
115dab1533
api: add available, availableTime, online, onlineTime properties ( #5391 )
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these replace ready and readyTime and allow to monitor the new
"always-available" feature.
2026-01-31 14:54:10 +01:00
Alessandro Ros
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
69fdd18d86
docs: update ( #5308 )
2026-01-28 11:52:42 +01:00
Alessandro Ros
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
Eugene Marushchenko
9e9fae9a10
add structured logging ( #5219 )
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2025-12-27 20:42:06 +01:00
Alessandro Ros
3de05c1330
api: always reply with JSON in case of success or failure ( #5252 )
...
Reply with "status": "ok" in case of success, and with "status":
"error" in case of error. This makes the API more accessible and user
friendly.
2025-12-07 10:37:55 +01:00
Omar Ferro
e168d2f3f4
apidocs: fix compatibility with OpenAPI 3.0 ( #5210 )
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double-precision numbers are now represented as double.
2025-11-22 23:05:08 +01:00
KHuynh
ade0cddeb3
support multiple CORS origins ( #5150 )
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Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2025-11-21 02:00:46 +01:00
Alessandro Ros
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
2690ef8e71
api: add /info endpoint ( #945 ) ( #5045 )
2025-09-30 10:39:27 +02:00
Alessandro Ros
558d1c3818
api: add tunnel and profile to RTSP conns and sessions ( #4985 )
2025-09-17 22:18:21 +02:00
Alessandro Ros
462fb2bd0f
allow setting UDP read buffer size ( #3308 ) ( #4846 )
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new parameters: rtspUDPReadBufferSize, rtpUDPReadBufferSize, mpegtsUDPReadBufferSize
2025-08-11 12:21:00 +02:00
Alessandro Ros
d0430d8ea5
support ingesting RTP streams ( #1515 ) ( #4843 )
2025-08-09 16:12:10 +02:00
Alessandro Ros
1f3014aeac
rename apidocs into api ( #4844 )
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this is to follow standard Golang project layout
2025-08-09 16:05:34 +02:00