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