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.
- 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
- refactor: remove unused MuteUserCmdMsg processor in bbb-gw
- fix(livekit): various adjustments to egress handling
- build: livekit-server-sdk@v2.9.3 (up from v2.6.2)
- build: @livekit/rtc-node@0.12.1 (up from 0.9.2)
- build: express@4.21.2 (up from 4.21.1)
- feat(livekit): add server domain to metadata
- fix(livekit): muteOnStart state inconsistency
- fix(livekit): treat screen share audio as separate tracks
- feat(livekit): sync screen share state with BBB
- fix(livekit): improve state sync and external user handling
- feat: add state sync and recording RPCs for LiveKit
- fix(core): onEvent is not a transaction, treat is as such
- fix: reorganize module forking to work around FIFO sched issues
- refactor(livekit): reorganize configs, add env var mappings
- refactor(livekit): handle mute requests server-side
- build: express@4.21.1
- build: mcs-js@0.0.21
- build: bump cross-spawn from 7.0.3 to 7.0.6 (transitive)
- feat: add
enabledflag to SFU modules, default to true - fix(livekit): do not sync stale egress instances
- build(livekit): add LiveKit submodule to base configuration
- feat(livekit): add track egress support
- feat(livekit): webhooks module
- feat(livekit): server side BBB <-> LiveKit event sync
- refactor(livekit): rename GenerateWebRtcToken* to GenerateLiveKitToken*
- feat: base-manager may opt out of connecting to mcs-core
- build(livekit-server-sdk): v2.6.2
- feat(livekit): add support for SIP trunking
- feat(mediasoup): pipe mediasoup logs to the application logger
- refactor(mediasoup): review log levels and metadata
- 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
- refactor(audio): set FLOWING logs to INFO level
- build(mediasoup): v3.14.11
- 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
- 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
- fix(audio): user is deafened when transferring to breakout rooms
- build(mediasoup): 3.13.24
- feat: add incrementBy util to prometheus-agent
- feat(core): add event callback and dispatch metrics
- fix: another edge case where subprocesses fail to recover
- fix: subprocesses fail to recover from multiple crashes
- 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)