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CHANGELOG

All notable changes (from 2.13.0 onwards) will be documented in this file. Intermediate pre-release changes will only be registered separately in their respective tag's CHANGELOG. Final releases will consolidate all intermediate changes in chronological order.

For previous changes, see the release notes.

V2.22.2

  • fix(livekit): guarantee agent subscriptions are cleared up
  • fix(livekit): only create RTC agents when necessary

v2.22.1

  • fix: memory leaks in BBBWebRTCRecorder's API module
  • fix(livekit): minor listener leaks in livekit-controller
  • build(deps): livekit-server-sdk@v2.15.1 (up from v2.14.0)
  • build(deps): @livekit/rtc-node@0.13.25 (up from 0.13.20)
  • build(deps): eslint@8.57.1 (dev), nodemon@3.1.14 (dev)
  • build(deps): npm audit fix

v2.22.0

  • feat(livekit): add histogram metrics for mute/unmute duration
  • fix(livekit): honor user-scoped conditional recording flags
  • chore: disable Kurento by default in base/example config file

v2.21.2

  • fix(livekit): incorrect callerIdNum causes audio join failures
  • build(deps): js-yaml@v4.1.1
  • build(deps): express@v4.22.1

v2.21.1

  • fix(livekit): incorrect mute state with muteOnStart=false + eventSource=agent
  • fix(livekit): race condition on token generation (clients fail to connect)

v2.21.0

  • feat(livekit): add agent conn state and event metrics
  • feat: add docker matrix build to support arm64
  • fix(livekit): add conn retry logic for RTC state sync agents
  • fix(livekit): BBB state sync stalls on idle rooms
  • fix(mediasoup): incorrect media direction causes tab sharing + audio to fail
  • build: add environment variable to disable file logging
  • build: remove QEMU setup from publish Docker workflow
  • build(deps): livekit-server-sdk@2.14.0
  • build(deps): @livekit/rtc-node@0.13.20
  • build(deps): pino@10.1.0
  • build(deps): pino-pretty@13.1.2
  • build(deps): bump actions/checkout from 4 to 5

v2.20.0

  • feat(livekit): RTC-based event source (alternative to webhooks)
  • feat: getRecordings primitive in bbb-webrtc-recorder client
  • feat(livekit): add recording resync
  • build: livekit-server-sdk@2.13.2

v2.19.2

fix: bbb-webrtc-recorder heartbeat generates inaccurate crash alerts

v2.19.1

  • fix(livekit): forcefully unpublish screenshare when server ejects it
  • fix(screenshare): ignore eject requests in mediasoup module if using LiveKit
  • fix(livekit): discard camera/screen events if meeting is not using LiveKit

v2.19.0

  • feat(livekit): add manual subscription capabilities to internal RTC clients
  • feat(livekit): token gen for breakout audio-only transfers
  • feat(screenshare): add userId to screenshare RPCs
  • feat: add support for camera recording consent flag
  • feat(livekit): add flag to enable custom BBB permission checks
  • feat(livekit): add metric for track publish handler errors
  • feat(livekit): DTMF handling
  • feat(livekit): implement SIP speaking detection
  • fix(livekit): stop recording retry on track unavailable error
  • fix(livekit): track unretriable start failures in Prom as error
  • fix(livekit): add missing label to retry failure metric
  • fix(livekit): guarantee internal RTC agent is hidden on BBB
  • fix(livekit): incorrect RTC agent config parsing
  • fix(livekit): recording fails due to incomplete track_publish event
  • fix(livekit): don't create RTC clients for non-BBB meetings
  • fix(livekit): incorrect metadata override for external participants
  • fix: bbb-webrtc-recorder stop requests have no timeout
  • fix: camera and screen states in BBB become stale on broken stop calls
  • fix(livekit): incorrect parsing of dispatch and trunk rule options
  • fix(livekit): SIP users have hardcoded Phone prefix in their names
  • fix(livekit): unhandled EjectUserFromVoiceConfSysMsg
  • refactor(livekit): move RTC agent mgmt to its own module
  • build: livekit-server-sdk@2.13.0
  • build: @livekit/rtc-node@0.13.18
  • build: transitive dep brace-expansion audit fix

v2.18.3

  • fix(livekit): false positive in recording error metric
  • refactor(livekit): cleanup recordings directly on meeting end

v2.18.2

  • fix(livekit): improvements to recording retry loop and event mgmt

v2.18.1

  • chore(livekit): bbb-webrtc-recorder as the default recording adapter

v2.18.0

  • feat(livekit): support recording LiveKit streams through bbb-webrtc-recorder
  • feat(livekit): add webrtc-recorder status and error metrics to LK module
  • feat(livekit): add recorder reqs/resps counter metrics
  • fix(livekit): make dial-in pin requirement configurable (default: false)
  • fix(livekit): create inbound trunks based on voice bridge number
  • fix(livekit): clean trunks and dispatch rules on restarts
  • fix(livekit): attach room metadata to inbound trunks and dispatch rules
  • fix(livekit): link dispatch rules to respective inbound trunks
  • fix(livekit): expose inbound trunks and dispatch rules configs
  • fix: restart loop on boot due to early requests
  • fix(livekit): guarantee uniqueness of "external" LK users

v2.17.1

  • build(medisoup): 3.14.14 (without extra debug logs)

v2.17.0

  • feat(livekit): add track egress support
  • feat(livekit): webhooks module
  • feat(livekit): server side BBB <-> LiveKit event sync
  • feat(livekit): add server domain to metadata
  • feat: base-manager may opt out of connecting to mcs-core
  • feat: add enabled flag to SFU modules, default to true
  • feat: add state sync and recording RPCs for LiveKit
  • feat(livekit): sync screen share state with BBB
  • feat(livekit): handle camera ejections
  • feat(livekit): add egress retry routines and Prometheus metrics
  • feat(livekit): add support for recordFullDurationMedia=false
  • feat(livekit): make egress retry timers configurable
  • fix(livekit): do not sync stale egress instances
  • fix(livekit): improve state sync and external user handling
  • fix(core): onEvent is not a transaction, treat is as such
  • fix: reorganize module forking to work around FIFO sched issues
  • fix(livekit): treat screen share audio as separate tracks
  • fix(livekit): muteOnStart state inconsistency
  • fix(livekit): various adjustments to egress handling
  • fix(livekit): ignore participants other than STANDARD and SIP
  • fix(livekit): do not send audio events to BBB if bridge is unused
  • fix(livekit): clean up rooms on meeting end
  • fix(livekit): add error handling to ParticipantLeft callback
  • fix(livekit): unhandled exception when recordFullDurationMedia=false
  • fix(livekit): retry egress regardless of error type
  • fix(livekit): handle egress shutdowns and assorted fixes
  • fix(livekit): temporarily disable custom maxParticipants limit
  • fix(freeswitch): UA fails due to unescaped "/" or malformed URI user
  • fix(audio): TLO callerID matching is too strict
  • fix(audio): channel incorrectly muted when transferring to breakout room
  • fix(audio): handle global audio bridge failures
  • fix(audio): make fs-consumer-bridge disconn handler verify current IDs
  • chore(audio): add option to control when global audio restarts
  • refactor(livekit): rename GenerateWebRtcToken* to GenerateLiveKitToken*
  • build(livekit): add LiveKit submodule to base configuration
  • build: mcs-js@0.0.21
  • build: bump cross-spawn from 7.0.3 to 7.0.6 (transitive)
  • build: express@4.21.2 (up from 4.21.1)
  • build: livekit-server-sdk@2.10.2 (pinned)
  • build: @livekit/rtc-node@0.13.6 (pinned)
  • build(docker): Automatic build pipeline for docker images
  • build(docker): Update build environment and Use multi stage builds
  • build(docker): add .github folder to dockerignore
  • build(docker): only push Docker images for release tags

v2.16.0

  • feat(livekit): add support for SIP trunking
  • feat(mediasoup): pipe mediasoup logs to the application logger
  • refactor(mediasoup): review log levels and metadata

v2.15.0

  • feat: add restartIce support for video/screenshare modules
  • refactor: rename ICE restart flag to restartIce, true by default
  • build: pino@9.3.2
  • build: config@3.3.12
  • build: ws@8.18.0
  • build: bufferutil@4.0.8
  • build: mcs-js@0.0.20
  • build: uuid@10.0.0
  • build: mediasoup-client@3.7.16
  • build: mediasoup@3.14.14
  • build: SIP.js@v0.7.5.14

v2.14.2

  • refactor(audio): set FLOWING logs to INFO level
  • build(mediasoup): v3.14.11

v2.14.1

  • fix(screenshare): presenter/viewer stop logs on all scenarios
  • refactor(screenshare): add presenter data to viewer logs
  • refactor(video): add video negotiation and flowing logs
  • build(mediasoup): 3.14.9

v2.14.0

  • feat(mediasoup): add least-loaded worker balancing strategy
  • feat(mediasoup): worker transposition (off by default)
  • feat(audio): dynamic global audio bridge mechanism
  • feat: livekit module, initial implementation
  • feat(audio): add signaling support for passive-sendrecv role
  • feat(freeswitch): overridable UA string
  • feat(audio): muteOnStart detection for conditional dialplans
  • feat(audio): mute passive-sendrecv clients on start
  • feat(audio): support for mute-and-hold on start
  • fix(audio): ignore TLO-incapable clients in hold/unhold metrics
  • fix(audio): mute/unmute stuck due to inconsistent hold status
  • fix(audio): hold/unhold loop when there are multiple sessions per user
  • fix(audio): muteOnStart sessions incorrectly muted on breakout transfers
  • fix(audio): header-provided userName incorrectly decoded
  • fix(audio): stuck unmute due to borked callerIdNum
  • fix(audio): correctly decode user name space chars
  • !build(npm): set min Node.js version to >=18.0.0
  • build: nodemon@3.1.3
  • build: ws@8.17.1
  • build(mediasoup): 3.14.8

v2.13.3

  • fix(audio): user is deafened when transferring to breakout rooms
  • build(mediasoup): 3.13.24

v2.13.2

  • feat: add incrementBy util to prometheus-agent
  • feat(core): add event callback and dispatch metrics
  • fix: another edge case where subprocesses fail to recover

v2.13.1

  • fix: subprocesses fail to recover from multiple crashes

v2.13.0

  • feat: add inbound queue size and job failure metrics
  • feat: add dry-run recording mode
  • feat: add time_to_mute/unmute metrics
  • feat: add warn logs for when hold/mute actions exceed max bucket time
  • feat(mediasoup): add mediasoup_ice_transport_protocol metric
  • feat(mediasoup): per-worker resource metrics
  • feat(mediasoup): add worker label to transport/router/prod/cons metrics
  • fix(audio): log and track metrics for hold/unhold timeouts
  • fix(bbb-webrtc-recorder): exception when removing nullish recording callbacks
  • fix(mediasoup): check for null producers
  • fix(screenshare): resolve subscriberAnswer job
  • fix(audio): prevent false positives in TLO toggle metrics
  • fix(test): wait for recorder to boot in stress test script
  • fix: set appropriate initial bitrates
  • fix(mediasoup): max bitrate for consumer-only transports not effective
  • fix(mediasoup): missing rtcp-fb and header exts in consumer-only offers
  • fix(audio): stricter adherence to router.mediaCodecs settings
  • fix(video): exception when destructuring null camera source
  • fix(mediasoup): only call consumer.changeProducer when appropriate
  • fix(mediasoup): capture icestatechange == disconnected
  • fix(mediasoup): invalid RTP header exts in default config
  • refactor: replace logger lib, Winston -> Pino
  • chore(mediasoup): expose webRtcTransport's iceConsentTimeout config
  • build: mediasoup-client@3.7.4
  • build: mediasoup@3.13.23
  • build: bump Docker and nvmrc to Node.js 20 (LTS)
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