node-sdks

Patch Changes

• Convert data streams to use livekit-ffi exposed data streams interface - #697 (@1egoman)

• update e2ee defaults to confirm to required FFI fields and update protocol - #708 (@lukasIO)

Minor Changes

• Add EgressClient.startEgress(), a unified convenience method that calls the v2 Egress.StartEgress RPC with a StartEgressRequest (or a partial init object). The existing per-type start helpers are unchanged. - #706 (@frostbyte73)

Patch Changes

• update e2ee defaults to confirm to required FFI fields and update protocol - #708 (@lukasIO)

Patch Changes

• fix: avoid dual package hazard by building single CJS version with an ESM wrapper - #700 (@lukasIO)

• Update to livekit ffi 0.12.73 - #698 (@1egoman)

Patch Changes

• Close audio streams when their track is unsubscribed. An unsubscribed track never receives eos from the FFI, so any AudioStream attached to it kept delivering frames — after a reconnect that meant the stale stream and the new subscription's stream both delivered the publisher's audio. - #701 (@davidzhao)

• bump FFI to 0.12.68 - #701 (@davidzhao)

Minor Changes

• Add LiveKitAPI, a unified entry point that exposes every server API through a property (api.room, api.egress, api.ingress, api.sip, api.agentDispatch, api.connector). It can be constructed with an API key and secret, or with a pre-signed token (no secret required, so it can run client-side), and falls back to the LIVEKIT_URL, LIVEKIT_API_KEY, LIVEKIT_API_SECRET, and LIVEKIT_TOKEN environment variables. Individual clients also accept a token via ClientOptions. - #695 (@davidzhao)

• Add SipCallError, a TwirpError subclass thrown by SipClient.createSipParticipant / transferSipParticipant when a call fails with a SIP status. It exposes the SIP response code and reason as sipStatusCode / sipStatus getters, while other error metadata remains available via metadata. - #695 (@davidzhao)

Patch Changes

• feat: auto failover APIs with LK Cloud - #686 (@davidzhao)

Patch Changes

• Fix AudioSource.waitForPlayout resolving immediately with audio still queued. The internal playout promise could be left resolved ("latched") by the drain timer firing during a gap between captures, or by clearQueue()/pause; a later waitForPlayout() then consumed the stale resolution and reported playout complete ~queueSizeMs early, clipping the tail of agent speech on every turn in downstream consumers. The waiter is now discarded when released and lazily re-created on the next captureFrame, mirroring python-sdks. As part of this, waitForPlayout() now resolves immediately when no audio is queued instead of blocking until the next release. - #693 (@chenghao-mou)

Minor Changes

• Support SIP media configs. - #687 (@dennwc)

Patch Changes

• Add callee dispatch rule support to createSipDispatchRule - #684 (@lukasIO)

Patch Changes

• Add initial support for frame processor usage directly on tracks - #671 (@1egoman)

Patch Changes

• Add deployment to CreateDispatchOptions - #675 (@lukasIO)

Patch Changes

• Add token revocation timestamp to RemoveParticipant - #668 (@cnderrauber)

Patch Changes

• bump ffi bindings, includes rpc v2 support - #666 (@stephen-derosa)

• Add method to query rtcStats on room via await room.getRtcStats() - #663 (@lukasIO)

Patch Changes

• Expose kindDetails on Participant and re-export the ParticipantKindDetail enum - #657 (@anunaym14)

• feat: handle track republish - #660 (@davidzhao)

• feat: add support for SimulateScenario - #659 (@davidzhao)

Patch Changes

• Bump @livekit/protocol to 1.45.6 and surface new fields on the agent dispatch and connector clients: - #657 (@anunaym14)

  • AgentDispatchClient.createDispatch: new restartPolicy option (cloud only)
  • ConnectorClient.dialWhatsAppCall / acceptWhatsAppCall: new ringingTimeout option (and waitUntilAnswered on accept)
  • ConnectorClient.disconnectWhatsAppCall: new optional disconnectReason parameter
  • Re-export JobRestartPolicy and DisconnectWhatsAppCallRequest_DisconnectReason

• Add canManageAgentSession to VideoGrant - #661 (@theomonnom)