cae9920f00
hls: improve muxer performance ( #5660 )
...
use a mutex instead of a channel to get current instance.
2026-04-21 20:16:36 +02:00
9e077744fd
metrics: improve performance ( #5663 )
...
use string.Builder instead of string concatenation
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-04-21 19:37:31 +02:00
2589c99639
return a custom error when body size limit is exceeded ( #5675 )
2026-04-21 19:00:30 +02:00
4d3026c914
bump hls.js to v1.6.16 ( #5665 )
2026-04-19 21:41:47 +02:00
caeccdceff
prevent out-of-memory errors ( #5674 )
...
impose a maximum size on body of incoming HTTP requests and responses.
2026-04-19 21:39:08 +02:00
67caaea4fe
playback: return errors as JSON ( #5656 )
...
this is aligned with all other HTTP-based services.
2026-04-09 17:11:03 +02:00
c7826d406b
hls: return JSON with error message in case path conf is not available ( #5655 )
...
this behavior is aligned with WebRTC one.
2026-04-08 23:26:04 +02:00
14e0a8f55b
expose token passed as query parameter to HTTP authentication too ( #5649 )
...
this allows to parse tokens coming from RTSP and RTMP without additional effort.
2026-04-06 18:10:29 +02:00
6d6ebee80d
deprecate authJWTInHTTPQuery and disable JWTs in query parameters ( #5648 )
...
This fixes a long standing security flaw. Even though it's a breaking
change, few users should be impacted since this feature has been discouraged
for some time.
2026-04-06 18:03:22 +02:00
4472bcfc4b
use "token" as query parameter key to pass tokens ( #5647 )
...
the legacy "jwt" query parameter key is still supported.
2026-04-06 17:51:59 +02:00
6a2bccd25b
add docsorder linter ( #5637 )
2026-04-04 17:01:15 +02:00
d4c6f95291
dump unencrypted TLS sessions ( #5624 )
...
when dumpPackets is true, embed TLS master keys into the dump, in a
format which is natively compatible with Wireshark.
2026-04-04 14:46:43 +02:00
f453b59cd6
improve listener labels ( #5635 )
...
add a label after every "listener opened on :XXX" message that mentions
protocols of every listener.
2026-04-03 16:35:31 +02:00
489a69af38
bump mediamtx-rpicamera to v2.5.5 ( #5629 )
2026-03-31 22:34:55 +02:00
0eb7089ed6
use safer atomic structs instead of atomic functions ( #5622 )
2026-03-31 11:30:50 +02:00
7418e51031
prevent directory traversal attacks ( #5602 )
...
Path names are used as part of paths in several components: in the
recorder, in the playback server and in every HTTP-based component
(WebRTC, HLS, API). Special characters that allow to escape from the
intended directory are now forbidden in order to prevent directory
traversal attacks.
2026-03-23 20:16:12 +01:00
d5d1de0bd2
webrtc: strip TWCC extension of incoming RTP packets ( #5146 ) ( #5605 )
...
The TWCC extension is used as part of the WebRTC congestion control
algorithm placed between the publisher and the server. If this
extension is routed untouched from the server to readers, it messes
with the congestion control algorithm present between the server and
each reader. Remove it.
2026-03-23 14:14:46 +01:00
8568d8c57c
webrtc: fix random absolute timestamps with Opus, G711 and LPCM ( #5597 )
...
When rewriting audio RTP timestamps in WebRTC egress, NTP was
derived using regenerated packet timestamps minus the incoming
RTP base timestamp.
That mixed timestamp domains and could shift absolute time by an
arbitrary offset while still exposing mapping as available.
Fix by using a consistent outgoing RTP domain in rewritten audio
paths:
- snapshot outgoing base timestamp before rewriting each unit
- compute NTP from (outgoing packet timestamp - outgoing base
timestamp)
This keeps RTP<->NTP mapping coherent for sender reports and prevents
random absolute-time offsets in WebRTC loopback with
useAbsoluteTimestamp.
---------
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-03-21 19:46:39 +01:00
3381b196fb
solve codec labels internally ( #5603 )
...
stop using gortsplib format.Format.Codec() for getting codec names.
2026-03-21 18:22:49 +01:00
0f42f087ab
test: add docslinks linter ( #5601 )
...
this checks links in the documentation.
2026-03-21 13:04:14 +01:00
2b302e7940
api: provide track details ( #5307 ) ( #5333 ) ( #1726 ) ( #5585 )
2026-03-18 00:01:43 +01:00
f98c9c59ca
rtsp: support unwrapping MPEG-TS tracks ( #5476 )
...
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
c8e07b55d5
api: add new stats ( #5582 )
...
- RTSPSession.outboundRTPPacketsDiscarded
- Path.inboundFramesInError
- SRTConn.outboundFramesDiscarded
- WebRTCSession.outboundFramesDiscarded
- RTMPConn.outboundFramesDiscarded
- HLSMuxer.outboundFramesDiscarded
2026-03-17 17:22:15 +01:00
adf45596a3
api: rename WebRTC stats to match RTSP ones ( #5581 )
2026-03-16 23:29:25 +01:00
02e05f1f16
metrics: add two missing SRT stats ( #5580 )
2026-03-16 22:18:16 +01:00
dfce20fefa
api: add RTSP reported lost packets ( #5198 ) ( #5579 )
...
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
3bad7045c1
api: add missing enums and move all enums in dedicated components ( #5576 )
2026-03-15 22:09:00 +01:00
c6bb332664
api: add deprecated fields to the OpenAPI definition ( #5575 )
2026-03-15 19:28:06 +01:00
a7290a2069
webrtc: support publishing multiple video/audio renditions ( #5573 )
...
this allows to receive multiple video tracks from OBS Studio with the
new WebRTC Simulcast feature introduced in v32.1.0.
2026-03-14 16:59:05 +01:00
d1fd3df27c
api: add user field to RTSP, RTMP, SRT, WebRTC conns and sessions ( #5104 ) ( #5565 )
2026-03-14 00:03:58 +01:00
9b36d50b8d
optionally validate JWT iss and aud claims ( #5569 )
2026-03-13 22:38:40 +01:00
a6856e9e58
tests: fix race conditions ( #5571 )
2026-03-13 22:32:06 +01:00
979ddaf81d
webrtc: fix panic with WHIP POST authentication failures ( #5566 )
2026-03-12 12:14:49 +01:00
184899dfab
fix: avoid buffering HTTP response body in loggerWriter ( #5552 )
...
loggerWriter was shadow-copying every response byte into a bytes.Buffer
to report the body size, causing the entire response to be accumulated
in memory for the lifetime of each request. Replace the buffer with a
plain int counter since dump() only ever reported the byte count anyway.
2026-03-07 18:36:10 +01:00
daec21324f
rtsp: fix rtsps scheme not being used in requests ( #5236 ) ( #5544 )
2026-03-07 18:30:40 +01:00
72832369f6
inherit MPEG-4 audio type from alwaysAvailableFile ( #5539 )
2026-03-01 10:12:35 +01:00
3f1ff994b7
revert to Go 1.25 ( #5521 ) ( #5538 )
...
It seems like Go 1.26 is causing segmentation faults, related to
channels, on Windows.
2026-03-01 10:09:11 +01:00
05ab631996
refactor alwaysAvailable tests ( #5537 )
2026-03-01 00:20:20 +01:00
16fdb71ee9
fix alwaysAvailableFile restarting when a publisher fails ( #5536 )
...
when a publisher try to start an online stream and there's an error,
alwaysAvailableFile restarted without any reason.
2026-02-28 22:43:15 +01:00
2f91c8198c
fix audio from alwaysAvailableFile not being streamed ( #5535 )
2026-02-28 21:46:25 +01:00
10e0271755
fix corrupted video with alwaysAvailableFile ( #5534 )
...
when alwaysAvailableFile points to a file with a H265 or H264 track,
server is started, an online stream is published and then closed, video
was getting corrupted since the online video was overriding the
parameters of the offline video.
2026-02-28 21:06:07 +01:00
4ebfcde8e2
fix MPEG-4 audio configuration not matching error ( #5468 ) ( #5533 )
...
This happened when using alwaysAvailableFile and a MPEG-4 audio track.
2026-02-28 20:17:44 +01:00
5e80554d20
improve playback precision of alwaysAvailable offline segment ( #5530 )
2026-02-28 18:39:52 +01:00
549300cbd4
prevent using alwaysAvailableFile and alwaysAvailableTracks together ( #5529 )
2026-02-28 18:31:41 +01:00
13551f0d98
fix panic when setting writeQueueSize to zero ( #5360 ) ( #5527 )
2026-02-28 09:18:42 +01:00
3568c54a02
improve JSON decoder performance ( #5526 )
...
avoid decoding JSON twice.
2026-02-27 23:25:04 +01:00
4974cacb94
improve video/audio sync of alwaysAvailable ( #5443 ) ( #5508 )
...
store elapsed time once for the entire stream and start PTS of sub
streams from there.
2026-02-22 18:25:32 +01:00
0f46e4982d
build(deps): bump github.com/bluenviron/gortsplib/v5 from 5.3.2-0.20260222121945-18b06189ef23 to 5.3.2 ( #5504 )
...
* build(deps): bump github.com/bluenviron/gortsplib/v5
Bumps [github.com/bluenviron/gortsplib/v5](https://github.com/bluenviron/gortsplib ) from 5.3.2-0.20260222121945-18b06189ef23 to 5.3.2.
- [Commits](https://github.com/bluenviron/gortsplib/commits/v5.3.2 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortsplib/v5
dependency-version: 5.3.2
dependency-type: direct:production
update-type: version-update:semver-patch
...
Signed-off-by: dependabot[bot] <support@github.com >
* additional changes
---------
Signed-off-by: dependabot[bot] <support@github.com >
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-02-22 17:14:02 +01:00
2e429158a8
log packet sizes if UDPMaxPayloadSize is exceeded ( #4668 )
...
Co-authored-by: Alessandro Ros <aler9.dev@gmail.com >
2026-02-22 13:42:12 +01:00
35e1f486c9
add integrated packet dumper ( #5488 )
...
this allows to dump any incoming and outgoing packet, to disk, in
pcapng format.
2026-02-22 13:34:05 +01:00
Alessandro Ros