@memberjunction/ai-bridge-livekit-native

The real native LiveKit room client for the MemberJunction LiveKit Realtime Bridge — the piece that makes an agent actually talk and hear in a live LiveKit room. It wraps @livekit/rtc-node (LiveKit's Node WebRTC participant) behind the NativeRoomModule contract that @memberjunction/ai-bridge-livekit's LiveKitNativeMeetingSdk expects.

realtime model ──IRealtimeSession──► AIBridgeEngine transport seam ──► LiveKitBridge
                                                                          │ SetSdkFactory(BindLiveKitNative())
                                                                          ▼
                                                            LiveKitNativeMeetingSdk  (the adapter — in ai-bridge-livekit)
                                                                          │ NativeModuleSpecifier
                                                                          ▼
                                                  @memberjunction/ai-bridge-livekit-native  ◄── THIS PACKAGE
                                                                          │
                                                                          ▼
                                                                  @livekit/rtc-node  (the real WebRTC participant)

What it does

• Voice out — publishAudio(pcm) captures the agent's synthesized PCM onto a published LiveKit audio track (AudioSource → LocalAudioTrack), so every participant hears the agent.
• Hearing in — each remote participant's subscribed audio track is read via an AudioStream and surfaced as a diarized { data, participantIdentity, name } frame.
• Roster + chat — participant connect/disconnect events and the reliable data channel.

Video/screen publish are documented one-time-warned no-ops in this voice-MVP wrapper (LiveKit supports them; they need a VideoSource + frame-format work that's out of scope here).

Use it

Point the bridge's NativeModuleSpecifier straight at this package and the engine does the rest (the LiveKit native binding is the registered default for DriverClass = 'LiveKitBridge'):

// the session Configuration the LiveKit coordinator passes to the bridge:
{
  "NativeModuleSpecifier": "@memberjunction/ai-bridge-livekit-native",
  "AccessToken": "<pre-signed LiveKit join token, minted upstream>",
  "BotDisplayName": "Sage",
  "RoomName": "demo-room"
}

Install the addon on the server that runs the agent bot:

npm install @livekit/rtc-node    # native addon — the agent bot host needs it

@livekit/rtc-node is an optionalDependency loaded lazily, so this package builds and unit-tests without the addon (tests inject a fake module). When the addon is absent at runtime, connect() throws an actionable "install @livekit/rtc-node" error.

🔴 Sample rates (read this before the live test)

The realtime model emits/consumes PCM at a specific rate, and @livekit/rtc-node resamples for you only if told the right rate. Defaults are 24 kHz mono (OpenAI-Realtime-compatible: xAI Grok Voice, etc.). Gemini Live wants 16 kHz inbound. A mismatch produces chipmunk / garbled audio — it is the #1 live-test failure mode, not a logic bug.

To override, write a one-line module and point NativeModuleSpecifier at it:

// my-livekit-native-gemini.ts
import { CreateLiveKitRtcNodeModule } from '@memberjunction/ai-bridge-livekit-native';
export default CreateLiveKitRtcNodeModule({ InboundSampleRate: 16000, OutboundSampleRate: 24000 });

API

• CreateLiveKitRtcNodeModule(opts?) → a NativeRoomModule. Options: OutboundSampleRate, InboundSampleRate, Channels, Loader (inject a fake @livekit/rtc-node for tests).
• LiveKitRtcNodeRoomClient — the NativeRoomClient implementation (connect / publishAudio / onAudioFrame / roster / publishData / disconnect).
• Pure helpers: pcmToInt16, int16ToArrayBuffer, participantsToArray — unit-tested directly.
• defaultRtcNodeLoader — the lazy @livekit/rtc-node loader (with the actionable absent-addon error).

⚠️ The exact @livekit/rtc-node surface is pinned from the SDK docs and marked with // VERIFY against @livekit/rtc-node notes in livekit-rtc-node-room.ts (Room/AudioSource/AudioStream/AudioFrame ctors, RoomEvent/TrackKind member names, connect/publishData/publishTrack signatures). A live test against a real LiveKit server should confirm them.

Testing

cd packages/AI/RealtimeBridge/Providers/LiveKitNative && npm run test

14 tests against a fake @livekit/rtc-node (no addon, no network): the pure PCM helpers, the connect→publish-track flow, both audio directions (outbound captureFrame at the configured rate + inbound AudioStream→diarized frame), participant events, roster, data publish, disconnect teardown, the sample-rate overrides, the video/screen no-ops, and the loader's present/absent branches.