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102 Commits
Author SHA1 Message Date
Alessandro RosandGitHub 0eb7089ed6 use safer atomic structs instead of atomic functions (#5622) 2026-03-31 11:30:50 +02:00
Alessandro RosandGitHub 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
t-animalGitHubaler9
 		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 RosandGitHub 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 RosandGitHub adf45596a3 api: rename WebRTC stats to match RTSP ones (#5581) 2026-03-16 23:29:25 +01:00
Alessandro RosandGitHub 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 RosandGitHub 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
bluenviron-botandGitHub a07526d57c bump Golang to 1.26 (#5460) 2026-02-21 14:48:53 +01:00
Alessandro RosandGitHub 98a9cf8238 improve tests (#5497) 2026-02-21 14:25:32 +01:00
Alessandro RosandGitHub f991821a5f webrtc: allow tuning WHEP timeouts (#5027) (#5479) 
add whepHandshakeTimeout, whepTrackGatherTimeout, whepSTUNGatherTimeout
 2026-02-18 16:29:25 +01:00
Alessandro RosandGitHub 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
dependabot[bot]GitHubdependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>aler9
 		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
Yaroslav MolochkoGitHubaler9
 		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 RosandGitHub 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 RosandGitHub 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
dependabot[bot]GitHubdependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
 		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
Alessandro RosandGitHub 55f93eddde rtsp: call rtsp.ToStream before advertising the stream (#5288) 2025-12-24 20:37:01 +01:00
Alessandro RosandGitHub 9e82b5233d fix 1-minute offset when recording a H264 WebRTC track (#5275) (#5278) 2025-12-19 13:32:14 +01:00
Alessandro RosandGitHub 2d02ca0df7 webrtc: use rtpReceiver and rtpSender to count packets (#5193) 2025-11-14 17:10:50 +01:00
Alessandro RosandGitHub d02fb4ed8e move constant (#5190) 2025-11-12 18:09:53 +01:00
Alessandro RosandGitHub ff187b6d8a update golangci-lint configuration (#5182) 2025-11-11 23:57:52 +01:00
Alessandro RosandGitHub 5be9b0eb97 rename variable (#5173) 2025-11-09 22:19:17 +01:00
Alessandro RosandGitHub 1ef5db5b3c extend udpReadBufferSize to WebRTC UDP sockets (#5130) 
this allows to decrease packet losses without touching system parameters.
 2025-10-29 11:45:15 +01:00
Alessandro RosandGitHub 0cdae40fe3 estimate absolute timestamp more precisely (#5078) 
When the absolute timestamp of incoming frames was not available, it
was filled with the current timestamp, which is influenced by latency
over time.

This mechanism is replaced by an algorithm that detects when latency is
the lowest, stores the current timestamp and uses it as reference
throughout the rest of the stream.
 2025-10-12 11:02:14 +02:00
Alessandro RosandGitHub f5f03562d3 rewrite Unit definition (#5079) 
Stream units now share the same struct, with a specialized payload.
 2025-10-11 12:18:51 +02:00
Alessandro RosandGitHub 691d7290fb webrtc: fix packet corruption when reading G711 (#5048) 
when a reader is reading a G711 track, stream units are modified
globally, affecting every other reader. This fixes the issue.
 2025-09-30 10:42:24 +02:00
Alessandro RosandGitHub 2024783eec webrtc: fix clippy audio when reading Opus (#3878) (#5047) 
Opus timestamp is now recomputed from scratch.
 2025-09-30 10:35:56 +02:00
Alessandro RosandGitHub cd80814009 do not include recorder and HLS muxer in sent bytes (#4380) (#5039) 
in API (/paths/list, /paths/get) and metrics (paths_bytes_sent), the
amount of sent bytes was increased even in case of writes to the
recorder and HLS muxer, which are not generating network traffic. This
fixes the issue.
 2025-09-29 09:28:04 +02:00
Alessandro RosandGitHub e3b8af8933 switch to gortsplib/v5 (#4978) 2025-09-16 13:10:34 +02:00
Alessandro RosandGitHub e0f4748839 modernize code (#4947) 2025-09-07 16:08:47 +02:00
Alessandro RosandGitHub c80220eb7c webrtc: solve domains in webrtcAdditionalHosts on server-side (#4817) (#4866) 2025-08-12 15:49:38 +02:00
Alessandro RosandGitHub b627128d0f remove context from webrtc.PeerConnection arguments (#4854) 
contexts are useless since there's already PeerConnection.Close().
 2025-08-12 15:19:59 +02:00
Alessandro RosandGitHub 5ae934887d remove custom forks of pion/webrtc and pion/ice (#4861) 
this fixes IPv6 reliability issues and allows to receive upstream
updates in a more linear way.
 2025-08-12 14:30:08 +02:00
Alessandro RosandGitHub 03623799f5 use slices.Contains when possible (#4859) 2025-08-12 12:28:20 +02:00
Alessandro RosandGitHub 6d4dfff959 webrtc: fix clock rate of outgoing RTCP receiver reports (#4852) 2025-08-11 13:59:10 +02:00
dependabot[bot]GitHubdependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
 		9e073cd34f build(deps): bump github.com/bluenviron/gortsplib/v4 (#4850) 
Bumps [github.com/bluenviron/gortsplib/v4](https://github.com/bluenviron/gortsplib) from 4.16.0 to 4.16.1.
- [Commits](https://github.com/bluenviron/gortsplib/compare/v4.16.0...v4.16.1)

---
updated-dependencies:
- dependency-name: github.com/bluenviron/gortsplib/v4
  dependency-version: 4.16.1
  dependency-type: direct:production
  update-type: version-update:semver-patch
...

Signed-off-by: dependabot[bot] <support@github.com>
Co-authored-by: dependabot[bot] <49699333+dependabot[bot]@users.noreply.github.com>
 2025-08-11 11:05:29 +02:00
Alessandro RosandGitHub eeb6ac3824 webrtc: re-enable ipv6 (#3227) (#4816) 2025-08-02 23:32:48 +02:00
Alessandro RosandGitHub 548cdbeb8f webrtc: fix crash introduced in #4795 (#4814) 2025-08-02 12:58:45 +02:00
Alessandro RosandGitHub 81abc2e008 webrtc: speed up candidate extraction (#4801) 2025-07-29 12:09:12 +02:00
Alessandro RosandGitHub 89e295eb4a metrics: add additional WebRTC metrics (#3304) (#4797) 
webrtc_sessions_rtp_packets_received, webrtc_sessions_rtp_packets_sent,
webrtc_sessions_rtp_packets_lost, webrtc_sessions_rtp_packets_jitter,
webrtc_sessions_rtcp_packets_received,
webrtc_sessions_rtcp_packets_sent.
 2025-07-29 11:17:12 +02:00
Alessandro RosandGitHub 534ea4d0c6 api: add additional WebRTC statistics (#4795) 
rtpPacketsReceived, rtpPacketsSent, rtpPacketsLost, rtpPacketsJitter,
rtcpPacketsReceived, rtcpPacketsSent
 2025-07-29 10:43:52 +02:00
Alessandro RosandGitHub d423a71aaa update linter settings (#4790) 2025-07-26 16:44:32 +02:00
Alessandro RosandGitHub 1083eea307 make RTP packet size compatible with RTSP/SRTP (#4692) 
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 RosandGitHub 3c703052f6 webrtc: fix writing tracks to some clients (#4602) 
some clients require PayloadType to be unique among all tracks, not
only among tracks of same kind.
 2025-06-03 16:23:38 +02:00
Alessandro RosandGitHub 823697210e webrtc: fix race condition after #4558 (#4564) 2025-05-27 15:17:31 +02:00
Alessandro RosandGitHub 5d203b4d98 webrtc: prevent routine leaks (#4558) 
wait for all routines to exit before assuming a WebRTC connection is closed.
 2025-05-26 11:33:46 +02:00
Alessandro RosandGitHub 77a3c7ae6e webrtc: route original absolute timestamp of packets (#1300) (#4415) 2025-04-12 11:34:27 +02:00
Alessandro RosandGitHub 49bcd35afd bump gortsplib (#4416) 2025-04-12 11:29:37 +02:00
Alessandro RosandGitHub 986e270862 count and log all discarded frames, decode errors, lost packets (#4363) 
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 RosandGitHub 416ac1357e bump mediacommon and gortsplib (#4364) 2025-03-24 17:25:18 +01:00