Alessandro Ros and GitHub
b5b63d02fc
support reading and publishing with Media-over-QUIC ( #5815 )
...
Media-over-QUIC is a streaming protocol built upon cutting edge
protocols (QUIC, HTTP3) and browser APIs (WebTransport, WebCodecs).
It's slightly faster than WebRTC, has an advanced data recovery
mechanism (placed at the frame level and not at the packet level), it
supports additional codecs (FLAC) and is less complicated to route.
2026-06-02 23:04:24 +02:00
c11810c2fc
build(deps): bump github.com/bluenviron/gortsplib/v5 from 5.5.3 to 5.5.4 ( #5822 )
...
* build(deps): bump github.com/bluenviron/gortsplib/v5 from 5.5.3 to 5.5.4
Bumps [github.com/bluenviron/gortsplib/v5](https://github.com/bluenviron/gortsplib ) from 5.5.3 to 5.5.4.
- [Commits](https://github.com/bluenviron/gortsplib/compare/v5.5.3...v5.5.4 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortsplib/v5
dependency-version: 5.5.4
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 >
2026-06-02 21:50:15 +02:00
Alessandro Ros and GitHub
640ddde51b
webrtc: fix connectivity after network changes ( #5097 ) ( #5818 )
2026-06-02 21:40:00 +02:00
Alessandro Ros and GitHub
7ab03cdc58
webrtc: rename incoming/outgoing into inbound/outbound ( #5816 )
2026-06-02 09:15:49 +02:00
Alessandro Ros and GitHub
0f5a76e253
modernize code ( #5814 )
2026-05-31 20:33:48 +02:00
Alessandro Ros and GitHub
2795962d8e
hls: support reading and publishing FLAC ( #5778 ) ( #5791 )
2026-05-21 17:00:59 +02:00
Alessandro Ros and GitHub
0edc3dbe86
rtmp: support reading and writing FLAC ( #5778 ) ( #5789 )
2026-05-21 14:50:06 +02:00
Alessandro Ros and GitHub
c05c14d9af
webrtc: support WHIP ICE restarts ( #5183 ) ( #5770 )
2026-05-15 19:15:42 +02:00
Alessandro Ros and GitHub
cdea2dc167
webrtc: support interacting with servers with no trickle ICE ( #5273 ) ( #5757 )
2026-05-12 21:07:16 +02:00
Alessandro Ros and GitHub
f542ccaf19
webrtc: fix checking POST responses ( #5758 )
2026-05-12 19:06:08 +02:00
eh and GitHub
fcf5eb092d
hls: support reading and writing KLV ( #5604 )
2026-04-30 12:14:00 +02:00
Alessandro Ros and GitHub
66a67188fe
tests: use testing.T.TempDir rather than os.TempDir ( #5709 )
2026-04-29 22:47:47 +02:00
Alessandro Ros and GitHub
f87d9e659e
hls: track sessions ( #962 ) ( #5683 )
...
sessions are now tracked through cookies or query parameters.
This provides the ability to inspect sessions through logs, metrics and
API, allows more precise tracking of outbound bytes, decreases load on
external HTTP authentication URLs since they are now called once per
session and not once per request.
2026-04-25 21:10:34 +02:00
Alessandro Ros and GitHub
2589c99639
return a custom error when body size limit is exceeded ( #5675 )
2026-04-21 19:00:30 +02:00
Alessandro Ros and GitHub
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
Alessandro Ros and GitHub
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
Alessandro Ros and GitHub
0eb7089ed6
use safer atomic structs instead of atomic functions ( #5622 )
2026-03-31 11:30:50 +02:00
Alessandro Ros and GitHub
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
Alessandro Ros and GitHub
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 and GitHub
adf45596a3
api: rename WebRTC stats to match RTSP ones ( #5581 )
2026-03-16 23:29:25 +01:00
Alessandro Ros and GitHub
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
Alessandro Ros and GitHub
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
Alessandro Ros and GitHub
a6856e9e58
tests: fix race conditions ( #5571 )
2026-03-13 22:32:06 +01:00
JulienCossec and GitHub
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
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
Alessandro Ros and GitHub
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
bluenviron-bot and GitHub
a07526d57c
bump Golang to 1.26 ( #5460 )
2026-02-21 14:48:53 +01:00
Alessandro Ros and GitHub
98a9cf8238
improve tests ( #5497 )
2026-02-21 14:25:32 +01:00
Alessandro Ros and GitHub
87fbfebf06
webrtc: add whepBearerTokenParameter ( #3796 ) ( #5486 )
...
this allows to pass Authorization: Bearer to servers that require it.
2026-02-19 11:57:39 +01:00
Alessandro Ros and GitHub
f991821a5f
webrtc: allow tuning WHEP timeouts ( #5027 ) ( #5479 )
...
add whepHandshakeTimeout, whepTrackGatherTimeout, whepSTUNGatherTimeout
2026-02-18 16:29:25 +01:00
Alessandro Ros and GitHub
ddaafd4ef3
rtmp: prevent legacy clients from reading multiple video/audio tracks ( #5478 )
2026-02-18 11:02:58 +01:00
Alessandro Ros and GitHub
bbefa5bc8b
improve tests ( #5469 )
2026-02-15 12:50:26 +01:00
bluenviron-bot and GitHub
ea6977d918
bump golangci-lint to 2.9.0 ( #5459 )
2026-02-12 15:18:02 +01:00
88a15c7654
build(deps): bump github.com/bluenviron/gortmplib from 0.2.0 to 0.2.1 ( #5394 )
...
* build(deps): bump github.com/bluenviron/gortmplib from 0.2.0 to 0.2.1
Bumps [github.com/bluenviron/gortmplib](https://github.com/bluenviron/gortmplib ) from 0.2.0 to 0.2.1.
- [Commits](https://github.com/bluenviron/gortmplib/compare/v0.2.0...v0.2.1 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortmplib
dependency-version: 0.2.1
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 >
2026-01-31 15:57:00 +01:00
Alessandro Ros and GitHub
a56408db19
add always available streams ( #5335 )
...
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
06c7cc3d7f
fix race condition when closing HTTP servers ( #5390 )
...
when a HTTP server is closed, open connections are now immediately
closed and open routines are waited before the server is considered
closed.
2026-01-31 14:09:10 +01:00
aa6dc4ea9f
build(deps): bump github.com/pion/webrtc/v4 from 4.2.1 to 4.2.3 ( #5341 )
...
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 >
2026-01-20 18:21:05 +01:00
8901ac4e9c
webrtc: support reading KLV tracks ( #4722 )
...
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2026-01-16 23:40:21 +01:00
Alessandro Ros and GitHub
653d10fb75
use the same filtering process for every codec ( #5324 )
...
this allows to apply features that were previously implemented for
single codecs (like RTP packet resizing), to any codec, and simplifies
future development.
2026-01-16 14:03:13 +01:00
Alessandro Ros and GitHub
e4c60e1e62
tests: fix race condition ( #5338 )
2026-01-12 21:46:48 +01:00
Alessandro Ros and GitHub
cd34de8770
print last decode error ( #5319 )
...
decode errors are still grouped together, but the last one is now
printed.
2026-01-05 15:28:24 +01:00
62effa79ef
build(deps): bump github.com/bluenviron/gortmplib from 0.1.2 to 0.2.0 ( #5300 )
...
Bumps [github.com/bluenviron/gortmplib](https://github.com/bluenviron/gortmplib ) from 0.1.2 to 0.2.0.
- [Commits](https://github.com/bluenviron/gortmplib/compare/v0.1.2...v0.2.0 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortmplib
dependency-version: 0.2.0
dependency-type: direct:production
update-type: version-update:semver-minor
...
Signed-off-by: dependabot[bot] <support@github.com >
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
2025-12-28 17:29:27 +01:00
f843df4ab8
build(deps): bump github.com/bluenviron/mediacommon/v2 ( #5297 )
...
Bumps [github.com/bluenviron/mediacommon/v2](https://github.com/bluenviron/mediacommon ) from 2.5.3 to 2.6.0.
- [Commits](https://github.com/bluenviron/mediacommon/compare/v2.5.3...v2.6.0 )
---
updated-dependencies:
- dependency-name: github.com/bluenviron/mediacommon/v2
dependency-version: 2.6.0
dependency-type: direct:production
update-type: version-update:semver-minor
...
Signed-off-by: dependabot[bot] <support@github.com >
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
2025-12-28 15:44:36 +01:00
Alessandro Ros and GitHub
55f93eddde
rtsp: call rtsp.ToStream before advertising the stream ( #5288 )
2025-12-24 20:37:01 +01:00
Alessandro Ros and GitHub
9e82b5233d
fix 1-minute offset when recording a H264 WebRTC track ( #5275 ) ( #5278 )
2025-12-19 13:32:14 +01:00
ade0cddeb3
support multiple CORS origins ( #5150 )
...
Co-authored-by: aler9 <46489434+aler9@users.noreply.github.com >
2025-11-21 02:00:46 +01:00
Alessandro Ros and GitHub
2d02ca0df7
webrtc: use rtpReceiver and rtpSender to count packets ( #5193 )
2025-11-14 17:10:50 +01:00
Alessandro Ros and GitHub
d02fb4ed8e
move constant ( #5190 )
2025-11-12 18:09:53 +01:00
Alessandro Ros and GitHub
ff187b6d8a
update golangci-lint configuration ( #5182 )
2025-11-11 23:57:52 +01:00
