build(bbb-webrtc-sfu): v2.14.0

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
This commit is contained in:
prlanzarin 2024-07-23 09:23:50 -03:00
parent 1c57db94bb
commit 6918d4e090

View File

@ -1 +1 @@
git clone --branch v2.13.3 --depth 1 https://github.com/bigbluebutton/bbb-webrtc-sfu bbb-webrtc-sfu
git clone --branch v2.14.0 --depth 1 https://github.com/bigbluebutton/bbb-webrtc-sfu bbb-webrtc-sfu