Alessandro Ros
c21c969a8c
revert #4583 ( #4606 )
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This reverts commit 500d18b6c6
2025-06-03 20:27:53 +02:00
Alessandro Ros
500d18b6c6
rtmp: fix timeout when publishing with GLive T80 ( #4002 ) ( #4583 )
2025-06-03 18:26:00 +02:00
bluenviron-bot
870b99f69f
bump hls.js to v1.6.5 ( #4601 )
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2025-06-03 07:53:46 +02:00
github-actions[bot]
eeee743d6c
bump hls.js to v1.6.4 ( #4586 )
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2025-05-31 09:48:14 +02:00
github-actions[bot]
945b6d5cb3
bump hls.js to v1.6.3 ( #4584 )
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2025-05-30 09:17:15 +02:00
Alessandro Ros
189c99d5fc
improve tests ( #4581 )
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2025-05-29 20:37:44 +02:00
Alessandro Ros
5d203b4d98
webrtc: prevent routine leaks ( #4558 )
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wait for all routines to exit before assuming a WebRTC connection is closed.
2025-05-26 11:33:46 +02:00
Alessandro Ros
5d4653ba4e
rtsp: fix authentication regression ( #4557 )
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since #4267 it was impossible to perform authentication when protocol
is RTSP and credentials are hashed.
2025-05-26 11:08:27 +02:00
Alessandro Ros
1b9dfbd367
rtmp: support connecting to sources that require standard credentials ( #4530 )
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2025-05-15 14:23:03 +02:00
Alessandro Ros
f97213ae6e
support passing JWTs through the password field ( #4516 )
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This is safer than passing JWTs through query parameters, unfortunately support is limited.
2025-05-10 22:54:24 +02:00
Alessandro Ros
a85d8f1391
rtmp: fix connect command when reading ( #4512 )
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when reading, the "connect" command should not contain fpad,
capabilities, audioCodecs, videoCodecs, videoFunction.
2025-05-10 15:26:48 +02:00
Alessandro Ros
defee1eed9
hls: fix toggling hlsAlwaysRemux after server is started ( #4503 )
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When hlsAlwaysRemux was switched from false to true, through API or hot
reloading, muxers of existing paths were not created. This fixes the
issue.
2025-05-09 22:50:11 +02:00
Alessandro Ros
77c77c7f2f
webrtc: prevent payload type conflict in read page ( #4456 )
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2025-04-21 15:24:01 +02:00
Alessandro Ros
0c501e2507
webrtc: improve JavaScript classes ( #4455 )
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* add close() method
* move support functions into classes as static methods
* convert arrow functions into standard functions
* make most methods private
* add JSDocs
2025-04-21 15:01:46 +02:00
Xiaobo Liu
44aa4ba50f
webrtc: simplify webrtcTurnSecretExpiration ( #4453 )
2025-04-21 14:21:45 +02:00
Alessandro Ros
77a3c7ae6e
webrtc: route original absolute timestamp of packets ( #1300 ) ( #4415 )
2025-04-12 11:34:27 +02:00
Alessandro Ros
49bcd35afd
bump gortsplib ( #4416 )
2025-04-12 11:29:37 +02:00
github-actions[bot]
e7388e1125
bump hls.js to v1.6.2 ( #4411 )
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Co-authored-by: mediamtx-bot <bot@mediamtx>
2025-04-11 11:14:42 +02:00
github-actions[bot]
070ffe739b
bump hls.js to v1.6.1 ( #4398 )
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2025-04-05 18:33:56 +02:00
Alessandro Ros
848a8b1031
webrtc: fix codec collision in Chrome ( #4394 )
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Fix error "A BUNDLE group contains a codec collision for
payload_type='96. All codecs must share the same type, encoding name,
clock rate and parameters. (INVALID_PARAMETER)"
2025-04-04 14:31:37 +02:00
Alessandro Ros
8b98c02903
rtsp: route original absolute timestamp of packets ( #1300 ) ( #4378 )
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2025-03-29 19:08:42 +01:00
github-actions[bot]
6250d6c485
bump hls.js to v1.6.0 ( #4374 )
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2025-03-28 08:48:10 +01:00
Alessandro Ros
f851cb6961
add missing Start() call; fix wrong sprintf ( #4369 )
2025-03-26 12:47:09 +01:00
Alessandro Ros
986e270862
count and log all discarded frames, decode errors, lost packets ( #4363 )
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Discarded frames, decode errors and lost packets were logged
individually, then there was a mechanism that prevented more than 1 log
entry per second from being printed, resulting in inaccurate reports.
Now discarded frames, decode errors and lost packets are accurately
counted, and their count is printed once every second.
2025-03-25 21:59:58 +01:00
Alessandro Ros
416ac1357e
bump mediacommon and gortsplib ( #4364 )
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2025-03-24 17:25:18 +01:00
Alessandro Ros
b329c4bbe8
replace New* with Initialize() ( #4345 )
2025-03-16 15:34:53 +01:00
Alessandro Ros
371b9585a6
change error message when stream is not found ( #4301 )
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Switch from 'no one is publishing to path' to 'no stream is available
on path' since the stream might be provided from a static source too.
2025-03-03 18:09:22 +01:00
Alessandro Ros
c692f3b78c
webrtc: rewrite WHIP client ( #4299 )
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2025-03-01 17:01:57 +01:00
Alessandro Ros
5c6cf58d75
webrtc: fix connecting to TCP-only sources ( #4293 )
2025-03-01 11:07:50 +01:00
Alessandro Ros
386be42784
rtsp: rewrite authentication around ServerConn.VerifyCredentials ( #4267 )
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2025-02-18 17:54:13 +01:00
Alessandro Ros
d822c2a200
rtsp: support using relative URLs in sourceRedirect ( #4260 )
2025-02-18 15:23:07 +01:00
Alessandro Ros
244da930a1
switch to mediacommon/v2 ( #4259 )
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2025-02-17 14:54:58 +01:00
Jean-Philippe Bergeron
e8297478f3
Configurable webrtcSTUNGatherTimeout ( #4221 )
2025-02-07 16:34:17 +01:00
Alessandro Ros
1a1da5edda
fix freeze when getting metrics and a RTSP connection is being created ( #4179 ) ( #4216 )
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2025-02-02 14:03:17 +01:00
Alessandro Ros
a1c6da84dc
webrtc: fix detecting closure of some sessions ( #4204 ) ( #4212 )
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2025-02-01 13:43:57 +01:00
Alessandro Ros
5cb5dc4e12
remove useless checks ( #4213 )
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2025-01-31 20:57:50 +01:00
dependabot[bot]
c6a9d81d4c
build(deps): bump github.com/bluenviron/mediacommon ( #4193 )
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Bumps [github.com/bluenviron/mediacommon](https://github.com/bluenviron/mediacommon ) from 1.13.4-0.20250119173005-20b296ab1174 to 1.13.4.
- [Commits](https://github.com/bluenviron/mediacommon/commits/v1.13.4 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/mediacommon
dependency-type: direct:production
update-type: version-update:semver-patch
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Signed-off-by: dependabot[bot] <support@github.com>
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
2025-01-24 11:04:08 +01:00
github-actions[bot]
f40f491b09
bump hls.js to v1.5.20 ( #4185 )
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2025-01-22 07:15:45 +01:00
Alessandro Ros
c1e1c74f6e
webrtc: fix setting video width, height and rate when publishing a camera with browser ( #4167 ) ( #4171 )
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2025-01-18 18:57:56 +01:00
Alessandro Ros
8f6267deb8
bump pion/webrtc to v4 ( #4145 )
2025-01-13 23:19:29 +01:00
Alessandro Ros
8f04264fe5
webrtxc: fix MTX_QUERY not set when reading or publishing ( #4138 ) ( #3937 ) ( #4141 )
2025-01-11 17:29:48 +01:00
github-actions[bot]
5077fb2238
bump hls.js to v1.5.19 ( #4140 )
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2025-01-11 10:25:29 +01:00
aler9
1342431c15
webrtc: fix reconnecting to server after a failure ( #4022 )
2025-01-06 13:51:08 +01:00
Alessandro Ros
d4c29f8283
webrtc: switch to recvonly transceivers ( #4059 ) ( #4108 )
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This fixes compatibility with devices that support decoding AV1 but
don't support encoding it.
This was previously impossible to achieve due to a bug that prevented
video from being displayed when recvonly transceivers were in use and
audio was not present.
2025-01-04 16:36:03 +01:00
Alessandro Ros
b49acb1e00
accept durations expressed as days (i.e. '1d') ( #4094 )
2025-01-02 12:44:15 +01:00
Alessandro Ros
8cbbbc05c3
rtmp: support additional Enhanced-RTMP features ( #3685 ) ( #4088 )
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new features:
* support publishing Opus and AC-3 tracks
* support publishing more than 2 tracks. This is compatible with OBS multitrack video and OBS VOD audio track
2025-01-02 12:44:01 +01:00
Alessandro Ros
6425bd5a8e
srt: fix compatibility with Blackmagic Design hardware ( #3701 ) ( #4087 )
2025-01-02 12:43:37 +01:00
Alessandro Ros
dfb792eb90
rtsp: rename encryption, protocols, serverKey, serverCert into rtspEncryption, rtspProtocols, rtspServerKey, rtspServerCert ( #4078 )
2024-12-25 18:26:14 +01:00
Alessandro Ros
8cb584b514
api, metrics: fix RTC lowercase in webRTCSession and webRTCSource ( #4076 )
2024-12-25 17:47:24 +01:00
Alessandro Ros
28970b323e
fix MTX_CONN_TYPE value with RTSP connections ( #3967 ) ( #4075 )
2024-12-25 17:42:19 +01:00
