@relaymessenger/livekit 0.1.0-staging.0

package/LICENSE

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+MIT License
+
+Copyright (c) 2026 Companion Inc.
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

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+# `@relaymessenger/livekit`
+
+`@relaymessenger/livekit` connects Relay Calls to TypeScript LiveKit Agents.
+Relay stays responsible for the Call resource and signaling; the package owns
+the Node WebRTC peer and translates audio between Relay and LiveKit `AudioFrame`
+objects. Application code does not handle Cloudflare, SDP, ICE, or SFU
+credentials.
+
+Install it next to the Relay SDK and the LiveKit Agents runtime:
+
+```sh
+npm install @relaymessenger/sdk @relaymessenger/livekit \
+  @livekit/agents @livekit/rtc-node
+```
+
+## Answer a Relay Call
+
+Agents receive `call.created` through the normal Relay Webhook or acknowledged
+Agent WebSocket. Joining that Call's authenticated room answers it; there is no
+agent call URL or separate accept endpoint.
+
+```ts
+import { AgentSession } from "@livekit/agents";
+import Relay from "@relaymessenger/sdk";
+import { RelayLiveKitCall } from "@relaymessenger/livekit";
+
+const relay = new Relay({ apiKey: process.env.RELAY_AGENT_TOKEN! });
+
+async function answerCall(callId: string, session: AgentSession) {
+  const call = await RelayLiveKitCall.connect({ relay, callId });
+  call.attach(session);
+
+  call.transport.on("ended", () => {
+    void call.close();
+  });
+
+  return call;
+}
+```
+
+`RelayLiveKitCall.connect()` does not resolve until the WebRTC media peer has
+reached `connected`. `RelayAudioInput` converts the remote participant's PCM16
+audio to LiveKit frames. `RelayAudioOutput` publishes LiveKit TTS frames to
+Relay, pacing larger frames into 10 ms WebRTC source slices. `clearBuffer()`
+drops queued outbound audio that has not yet reached WebRTC.
+
+Use `setMuted(true)` to publish participant mute state, `end()` to end the Relay
+Call, and `close()` for local cleanup. If the signaling connection is replaced,
+`call.transport.reconnect()` keeps the existing media peer and replays the same
+audio publication so Relay can return its cached answer.
+
+## Provider-neutral transport
+
+The WebRTC/PCM layer is a separate public entry point for future adapters that
+do not use LiveKit Agents:
+
+```ts
+import { RelayCallTransport } from "@relaymessenger/livekit/transport";
+
+const room = relay.calls.room(callId);
+const transport = new RelayCallTransport({ room });
+
+transport.on("audio", ({ samples, sampleRate, channelCount }) => {
+  // Interleaved signed PCM16 from the remote Relay participant.
+});
+
+await transport.connect();
+await transport.writeAudio({
+  samples,
+  sampleRate: 48_000,
+  channelCount: 1,
+});
+```
+
+`RelayCallTransport` consumes the SDK's `CallRoom`; it does not duplicate the
+room protocol. The default engine is `werift` (pure TypeScript WebRTC) with
+`@evan/opus` (prebuilt Opus, WASM fallback), so no native WebRTC binding is
+loaded. Pass `engine: "wrtc"` to use the optional `@roamhq/wrtc` binding
+instead. The transport boundary remains provider-neutral for additional adapters. `@livekit/agents`
+and `@livekit/rtc-node` are peer dependencies so the host agent process owns
+those runtimes.

package/dist/engine-werift.d.ts

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+import { RTCPeerConnection } from "werift";
+import type { RelayAudioSourceLike, RelayWebRTCFactory } from "./transport.js";
+/** Opus on the wire is always 48 kHz; Relay sends stereo so callers may pass either channel count. */
+export declare const WERIFT_SAMPLE_RATE = 48000;
+export declare const WERIFT_CHANNEL_COUNT = 2;
+/** One RTP packet carries 20 ms: 960 frames at 48 kHz, timestamp advances by 960. */
+export declare const WERIFT_PACKET_MS = 20;
+export type WeriftAudioSourceData = Parameters<RelayAudioSourceLike["onData"]>[0];
+/** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
+export declare const toWireFormat: (data: {
+    samples: Int16Array;
+    sampleRate: number;
+    channelCount: number;
+}) => Int16Array;
+export declare const createWeriftPeerConnection: () => RTCPeerConnection;
+export declare const createWeriftWebRTCFactory: () => RelayWebRTCFactory;
+//# sourceMappingURL=engine-werift.d.ts.map

package/dist/engine-werift.d.ts.map

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package/dist/engine-werift.js

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+/**
+ * Relay's default call participant engine: werift (pure-TypeScript WebRTC:
+ * ICE, DTLS, SRTP, RTP) plus `@evan/opus` (prebuilt libopus, WASM fallback).
+ *
+ * API sources read for this file, in node_modules/werift/lib/webrtc/src:
+ * - peerConnection.d.ts: `new RTCPeerConnection({ codecs: { audio: [...] } })`,
+ *   `addTransceiver(track, { direction })`, the W3C-style `ontrack`,
+ *   `onconnectionstatechange`, `localDescription`, `close()`, and the
+ *   `"icegatheringstatechange"` / `"connectionstatechange"` events emitted
+ *   through `EventTarget.addEventListener` (peerConnection.js:317, :330).
+ * - media/track.d.ts: `MediaStreamTrack({ kind })`, `writeRtp(RtpPacket)`,
+ *   `onReceiveRtp: Event<[RtpPacket, Extensions?]>` with `subscribe()`.
+ * - media/rtpSender.js:544-549: the sender rewrites ssrc and payloadType and
+ *   offsets sequenceNumber/timestamp, so the local track starts both at 0.
+ * - ../../rtp/src/codec/opus.d.ts: `OpusRtpPayload.deSerialize(buffer)`.
+ * - node_modules/@evan/opus/lib.d.ts: `Encoder.encode(pcm16)` returns one Opus
+ *   packet; `Decoder.decode(packet)` returns interleaved PCM16 bytes.
+ * Participant shapes copied from the headless phone that joined Relay's room
+ * through Cloudflare's SFU on 2026-09-21 (_runtime/bin/headless-phone-webrtc.mjs
+ * lines 93-160): `bundlePolicy: "max-bundle", iceServers: []` (Cloudflare
+ * supplies its ICE candidates in the answer), the local track is id
+ * "microphone" in stream "relay-call" (Relay-iOS RelayWebRTCAudioTransport),
+ * the first RTP packet carries the marker bit, empty payloads are skipped and a
+ * decode failure never tears the sink down.
+ */
+import { Decoder, Encoder } from "@evan/opus";
+import { MediaStreamTrack, OpusRtpPayload, RTCPeerConnection, RTCRtpCodecParameters, RtpHeader, RtpPacket, } from "werift";
+/** Opus on the wire is always 48 kHz; Relay sends stereo so callers may pass either channel count. */
+export const WERIFT_SAMPLE_RATE = 48_000;
+export const WERIFT_CHANNEL_COUNT = 2;
+/** One RTP packet carries 20 ms: 960 frames at 48 kHz, timestamp advances by 960. */
+export const WERIFT_PACKET_MS = 20;
+const PACKET_FRAMES = (WERIFT_SAMPLE_RATE * WERIFT_PACKET_MS) / 1000;
+const PACKET_SAMPLES = PACKET_FRAMES * WERIFT_CHANNEL_COUNT;
+/** Dynamic payload type; werift's sender replaces it with the negotiated one. */
+const LOCAL_PAYLOAD_TYPE = 111;
+/** Convert any PCM16 frame to interleaved 48 kHz stereo by linear interpolation. */
+export const toWireFormat = (data) => {
+    const { samples, sampleRate, channelCount } = data;
+    const inputFrames = Math.floor(samples.length / channelCount);
+    if (sampleRate === WERIFT_SAMPLE_RATE && channelCount === WERIFT_CHANNEL_COUNT) {
+        return samples;
+    }
+    const outputFrames = Math.round((inputFrames * WERIFT_SAMPLE_RATE) / sampleRate);
+    const out = new Int16Array(outputFrames * WERIFT_CHANNEL_COUNT);
+    const ratio = sampleRate / WERIFT_SAMPLE_RATE;
+    const leftIndex = 0;
+    const rightIndex = channelCount > 1 ? 1 : 0;
+    for (let frame = 0; frame < outputFrames; frame += 1) {
+        const position = frame * ratio;
+        const base = Math.min(Math.floor(position), inputFrames - 1);
+        const next = Math.min(base + 1, inputFrames - 1);
+        const fraction = position - Math.floor(position);
+        const left = samples[base * channelCount + leftIndex] * (1 - fraction)
+            + samples[next * channelCount + leftIndex] * fraction;
+        const right = samples[base * channelCount + rightIndex] * (1 - fraction)
+            + samples[next * channelCount + rightIndex] * fraction;
+        out[frame * 2] = Math.round(left);
+        out[frame * 2 + 1] = Math.round(right);
+    }
+    return out;
+};
+/**
+ * PCM16 in, paced 20 ms Opus RTP out. Frames are accumulated into whole
+ * packets; a timer drains one packet every 20 ms so the wire sees a steady
+ * cadence regardless of how the adapter chunks its writes.
+ */
+class WeriftAudioSource {
+    #encoder = new Encoder({
+        channels: WERIFT_CHANNEL_COUNT,
+        sample_rate: WERIFT_SAMPLE_RATE,
+        application: "voip",
+    });
+    #track = new MediaStreamTrack({ kind: "audio", id: "microphone", streamId: "relay-call" });
+    #packets = [];
+    #pending = new Int16Array(0);
+    #sequenceNumber = 1;
+    #timestamp = 0;
+    #first = true;
+    #pump;
+    #stopped = false;
+    createTrack() {
+        const track = this.#track;
+        const stop = track.stop;
+        track.stop = () => {
+            this.#stopped = true;
+            this.#packets.length = 0;
+            this.#pending = new Int16Array(0);
+            this.#stopPump();
+            stop();
+        };
+        return track;
+    }
+    onData(data) {
+        if (this.#stopped)
+            return;
+        const wire = toWireFormat(data);
+        const merged = new Int16Array(this.#pending.length + wire.length);
+        merged.set(this.#pending);
+        merged.set(wire, this.#pending.length);
+        let offset = 0;
+        while (merged.length - offset >= PACKET_SAMPLES) {
+            const frame = merged.subarray(offset, offset + PACKET_SAMPLES);
+            this.#packets.push(Buffer.from(this.#encoder.encode(frame)));
+            offset += PACKET_SAMPLES;
+        }
+        this.#pending = merged.slice(offset);
+        this.#startPump();
+    }
+    #startPump() {
+        if (this.#pump || this.#packets.length === 0)
+            return;
+        this.#sendNext();
+        if (this.#packets.length === 0)
+            return;
+        this.#pump = setInterval(() => {
+            this.#sendNext();
+            if (this.#packets.length === 0)
+                this.#stopPump();
+        }, WERIFT_PACKET_MS);
+        this.#pump.unref?.();
+    }
+    #stopPump() {
+        if (!this.#pump)
+            return;
+        clearInterval(this.#pump);
+        this.#pump = undefined;
+    }
+    #sendNext() {
+        const payload = this.#packets.shift();
+        if (!payload || this.#stopped)
+            return;
+        const header = new RtpHeader({
+            payloadType: LOCAL_PAYLOAD_TYPE,
+            sequenceNumber: this.#sequenceNumber,
+            timestamp: this.#timestamp,
+            ssrc: 0,
+            marker: this.#first,
+        });
+        this.#first = false;
+        this.#sequenceNumber = (this.#sequenceNumber + 1) & 0xffff;
+        this.#timestamp = (this.#timestamp + PACKET_FRAMES) >>> 0;
+        this.#track.writeRtp(new RtpPacket(header, payload));
+    }
+}
+/** Opus RTP in, PCM16 (48 kHz stereo) out, one `ondata` per packet. */
+class WeriftAudioSink {
+    ondata = null;
+    #decoder = new Decoder({ channels: WERIFT_CHANNEL_COUNT, sample_rate: WERIFT_SAMPLE_RATE });
+    #unsubscribe;
+    constructor(track) {
+        const { unSubscribe } = track.onReceiveRtp.subscribe((rtp) => {
+            const handler = this.ondata;
+            if (!handler || !rtp.payload?.length)
+                return;
+            const packet = OpusRtpPayload.deSerialize(rtp.payload);
+            let bytes;
+            try {
+                bytes = this.#decoder.decode(packet.payload);
+            }
+            catch {
+                return;
+            }
+            const samples = new Int16Array(bytes.byteLength >> 1);
+            new Uint8Array(samples.buffer).set(bytes.subarray(0, samples.byteLength));
+            handler({
+                samples,
+                sampleRate: WERIFT_SAMPLE_RATE,
+                bitsPerSample: 16,
+                channelCount: WERIFT_CHANNEL_COUNT,
+                numberOfFrames: samples.length / WERIFT_CHANNEL_COUNT,
+            });
+        });
+        this.#unsubscribe = unSubscribe;
+    }
+    stop() {
+        this.ondata = null;
+        this.#unsubscribe();
+    }
+}
+export const createWeriftPeerConnection = () => new RTCPeerConnection({
+    bundlePolicy: "max-bundle",
+    iceServers: [],
+    codecs: {
+        audio: [
+            new RTCRtpCodecParameters({
+                mimeType: "audio/opus",
+                clockRate: WERIFT_SAMPLE_RATE,
+                channels: WERIFT_CHANNEL_COUNT,
+            }),
+        ],
+    },
+});
+export const createWeriftWebRTCFactory = () => ({
+    createPeerConnection: () => createWeriftPeerConnection(),
+    createAudioSource: () => new WeriftAudioSource(),
+    createAudioSink: (track) => new WeriftAudioSink(track),
+});
+//# sourceMappingURL=engine-werift.js.map

package/dist/engine-werift.js.map

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package/dist/index.d.ts

@@ -0,0 +1,4 @@
+export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, type RelayLiveKitAudio, type RelayLiveKitAudioOptions, type RelayLiveKitConnectOptions, } from "./livekit.js";
+export { RelayCallTransport, RelayCallTransportError, type RelayAudioFrame, type RelayCallEngine, type RelayCallTransportCloseEvent, type RelayCallTransportOptions, } from "./transport.js";
+export { createWeriftWebRTCFactory } from "./engine-werift.js";
+//# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

@@ -0,0 +1 @@
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package/dist/index.js

@@ -0,0 +1,4 @@
+export { RelayAudioInput, RelayAudioOutput, RelayLiveKitCall, createRelayLiveKitAudio, } from "./livekit.js";
+export { RelayCallTransport, RelayCallTransportError, } from "./transport.js";
+export { createWeriftWebRTCFactory } from "./engine-werift.js";
+//# sourceMappingURL=index.js.map

package/dist/index.js.map

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package/dist/livekit.d.ts

@@ -0,0 +1,57 @@
+import { AudioInput, AudioOutput, type AgentSession } from "@livekit/agents";
+import { AudioFrame } from "@livekit/rtc-node";
+import type Relay from "@relaymessenger/sdk";
+import type { CallRoomOptions } from "@relaymessenger/sdk";
+import { RelayCallTransport } from "./transport.js";
+/** LiveKit Agents input backed by the remote Relay participant's WebRTC audio. */
+export declare class RelayAudioInput extends AudioInput {
+    #private;
+    constructor(transport: RelayCallTransport);
+    setAttached(attached: boolean): void;
+    close(): Promise<void>;
+}
+/** LiveKit Agents output that publishes TTS PCM to the Relay participant. */
+export declare class RelayAudioOutput extends AudioOutput {
+    #private;
+    constructor(transport: RelayCallTransport, sampleRate?: number);
+    captureFrame(frame: AudioFrame): Promise<void>;
+    flush(): void;
+    clearBuffer(): void;
+    close(): void;
+}
+export interface RelayLiveKitAudio {
+    input: RelayAudioInput;
+    output: RelayAudioOutput;
+}
+export interface RelayLiveKitAudioOptions {
+    outputSampleRate?: number;
+}
+/** Build LiveKit Agents audio IO around an already-connected Relay transport. */
+export declare const createRelayLiveKitAudio: (transport: RelayCallTransport, options?: RelayLiveKitAudioOptions) => RelayLiveKitAudio;
+export interface RelayLiveKitConnectOptions extends RelayLiveKitAudioOptions {
+    relay: Relay;
+    callId: string;
+    room?: CallRoomOptions;
+    connectionTimeoutMs?: number;
+}
+type AgentSessionAudioTarget = Pick<AgentSession, "input" | "output">;
+/**
+ * First-party bridge between a Relay Call and a LiveKit Agents AgentSession.
+ * Developers work with Relay call IDs and LiveKit audio IO; media negotiation
+ * stays inside the transport.
+ */
+export declare class RelayLiveKitCall {
+    #private;
+    readonly transport: RelayCallTransport;
+    readonly input: RelayAudioInput;
+    readonly output: RelayAudioOutput;
+    private constructor();
+    static connect(options: RelayLiveKitConnectOptions): Promise<RelayLiveKitCall>;
+    attach(session: AgentSessionAudioTarget): void;
+    detach(): void;
+    setMuted(muted: boolean): void;
+    end(): void;
+    close(): Promise<void>;
+}
+export {};
+//# sourceMappingURL=livekit.d.ts.map

package/dist/livekit.d.ts.map

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package/dist/livekit.js

@@ -0,0 +1,169 @@
+import { AudioInput, AudioOutput } from "@livekit/agents";
+import { AudioFrame } from "@livekit/rtc-node";
+import { TransformStream } from "node:stream/web";
+import { RelayCallTransport, } from "./transport.js";
+/** LiveKit Agents input backed by the remote Relay participant's WebRTC audio. */
+export class RelayAudioInput extends AudioInput {
+    #transport;
+    #writer;
+    #onAudio;
+    #attached = false;
+    #closed = false;
+    constructor(transport) {
+        super();
+        this.#transport = transport;
+        const channel = new TransformStream();
+        this.#writer = channel.writable.getWriter();
+        this.multiStream.addInputStream(channel.readable);
+        this.#onAudio = (frame) => {
+            if (this.#closed || !this.#attached)
+                return;
+            const audio = new AudioFrame(frame.samples, frame.sampleRate, frame.channelCount, frame.samples.length / frame.channelCount);
+            void this.#writer.write(audio).catch(() => undefined);
+        };
+        this.#transport.on("audio", this.#onAudio);
+    }
+    setAttached(attached) {
+        this.#attached = attached;
+    }
+    async close() {
+        if (this.#closed)
+            return;
+        this.#closed = true;
+        this.#transport.off("audio", this.#onAudio);
+        await this.#writer.close().catch(() => undefined);
+        await super.close();
+    }
+}
+/** LiveKit Agents output that publishes TTS PCM to the Relay participant. */
+export class RelayAudioOutput extends AudioOutput {
+    #transport;
+    #segmentDuration = 0;
+    #segmentStarted = false;
+    #segmentStartedAt = 0;
+    #closed = false;
+    constructor(transport, sampleRate = 48_000) {
+        super(sampleRate, undefined, { pause: false });
+        this.#transport = transport;
+    }
+    async captureFrame(frame) {
+        if (this.#closed)
+            throw new Error("Relay LiveKit audio output is closed.");
+        await super.captureFrame(frame);
+        if (!this.#segmentStarted) {
+            this.#segmentStarted = true;
+            this.#segmentStartedAt = Date.now();
+            this.onPlaybackStarted(this.#segmentStartedAt);
+        }
+        this.#segmentDuration += frame.samplesPerChannel / frame.sampleRate;
+        await this.#transport.writeAudio({
+            samples: frame.data,
+            sampleRate: frame.sampleRate,
+            channelCount: frame.channels,
+        });
+    }
+    flush() {
+        super.flush();
+        if (!this.#segmentStarted)
+            return;
+        this.onPlaybackFinished({
+            playbackPosition: this.#segmentDuration,
+            interrupted: false,
+        });
+        this.#resetSegment();
+    }
+    clearBuffer() {
+        this.#transport.clearAudio();
+        if (!this.#segmentStarted)
+            return;
+        this.onPlaybackFinished({
+            playbackPosition: Math.min(this.#segmentDuration, Math.max(0, (Date.now() - this.#segmentStartedAt) / 1_000)),
+            interrupted: true,
+        });
+        this.#resetSegment();
+    }
+    close() {
+        if (this.#closed)
+            return;
+        this.#closed = true;
+        this.clearBuffer();
+    }
+    #resetSegment() {
+        this.#segmentDuration = 0;
+        this.#segmentStarted = false;
+        this.#segmentStartedAt = 0;
+    }
+}
+/** Build LiveKit Agents audio IO around an already-connected Relay transport. */
+export const createRelayLiveKitAudio = (transport, options = {}) => ({
+    input: new RelayAudioInput(transport),
+    output: new RelayAudioOutput(transport, options.outputSampleRate),
+});
+/**
+ * First-party bridge between a Relay Call and a LiveKit Agents AgentSession.
+ * Developers work with Relay call IDs and LiveKit audio IO; media negotiation
+ * stays inside the transport.
+ */
+export class RelayLiveKitCall {
+    transport;
+    input;
+    output;
+    #session;
+    #closed = false;
+    constructor(transport, audio) {
+        this.transport = transport;
+        this.input = audio.input;
+        this.output = audio.output;
+    }
+    static async connect(options) {
+        const transportOptions = {
+            relay: options.relay,
+            callId: options.callId,
+            ...(options.room ? { room: options.room } : {}),
+            ...(options.webRTC ? { webRTC: options.webRTC } : {}),
+            ...(options.iceGatheringTimeoutMs === undefined
+                ? {}
+                : { iceGatheringTimeoutMs: options.iceGatheringTimeoutMs }),
+            ...(options.connectionTimeoutMs === undefined
+                ? {}
+                : { connectionTimeoutMs: options.connectionTimeoutMs }),
+        };
+        const transport = new RelayCallTransport(transportOptions);
+        await transport.connect();
+        return new RelayLiveKitCall(transport, createRelayLiveKitAudio(transport, options));
+    }
+    attach(session) {
+        if (this.#closed)
+            throw new Error("Relay LiveKit Call is closed.");
+        if (this.#session && this.#session !== session)
+            this.detach();
+        this.#session = session;
+        session.input.audio = this.input;
+        session.output.audio = this.output;
+    }
+    detach() {
+        if (!this.#session)
+            return;
+        if (this.#session.input.audio === this.input)
+            this.#session.input.audio = null;
+        if (this.#session.output.audio === this.output)
+            this.#session.output.audio = null;
+        this.#session = undefined;
+    }
+    setMuted(muted) {
+        this.transport.setMuted(muted);
+    }
+    end() {
+        this.transport.end();
+    }
+    async close() {
+        if (this.#closed)
+            return;
+        this.#closed = true;
+        this.detach();
+        this.output.close();
+        await this.input.close();
+        this.transport.close();
+    }
+}
+//# sourceMappingURL=livekit.js.map

package/dist/livekit.js.map

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package/dist/transport.d.ts

@@ -0,0 +1,66 @@
+import type { CallRoomEndedFrame, CallRoomOptions, CallRoomStateFrame, Relay } from "@relaymessenger/sdk";
+export interface RelayAudioFrame {
+    /** Interleaved signed PCM16 samples. */
+    samples: Int16Array;
+    sampleRate: number;
+    channelCount: number;
+}
+export type RelayCallEngine = "werift" | "wrtc";
+export interface RelayCallTransportOptions {
+    relay: Relay;
+    callId: string;
+    room?: CallRoomOptions;
+    /**
+     * WebRTC engine. `"werift"` (default) is pure TypeScript plus prebuilt Opus
+     * and loads no native binding. `"wrtc"` loads `@roamhq/wrtc` on demand.
+     */
+    engine?: RelayCallEngine;
+    /** Maximum wait for the WebRTC peer to become connected. Defaults to 15 seconds. */
+    connectionTimeoutMs?: number;
+}
+export interface RelayCallTransportCloseEvent {
+    code: number;
+    reason: string;
+    wasClean: boolean;
+}
+type TransportEventMap = {
+    audio: [RelayAudioFrame];
+    connected: [];
+    roomState: [CallRoomStateFrame];
+    ended: [CallRoomEndedFrame];
+    error: [Error];
+    close: [RelayCallTransportCloseEvent];
+};
+type TransportEvent = keyof TransportEventMap;
+type TransportListener<K extends TransportEvent> = (...args: TransportEventMap[K]) => void;
+export declare class RelayCallTransportError extends Error {
+    readonly code?: string;
+    constructor(message: string, code?: string);
+}
+/**
+ * Provider-neutral Node WebRTC bridge for Relay Call rooms.
+ *
+ * The class owns Relay media negotiation internally. Higher-level adapters only
+ * exchange PCM16 frames and call lifecycle events.
+ */
+export declare class RelayCallTransport {
+    #private;
+    constructor(options: RelayCallTransportOptions);
+    on<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
+    off<K extends TransportEvent>(event: K, listener: TransportListener<K>): this;
+    connect(): Promise<void>;
+    /** Replace only the room signaling socket and replay the identical publication. */
+    reconnect(): Promise<void>;
+    /**
+     * Feed interleaved PCM16 audio into Relay. Frames are split into 10 ms WebRTC
+     * source slices and paced in real time, so adapters may push larger chunks.
+     */
+    writeAudio(frame: RelayAudioFrame): Promise<void>;
+    /** Drop queued outgoing PCM. At most one already-submitted 10 ms slice remains. */
+    clearAudio(): void;
+    setMuted(muted: boolean): void;
+    end(): void;
+    close(): void;
+}
+export {};
+//# sourceMappingURL=transport.d.ts.map

package/dist/transport.d.ts.map

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package/dist/transport.js

@@ -0,0 +1,417 @@
+const DEFAULT_ICE_GATHERING_TIMEOUT_MS = 10_000;
+const DEFAULT_CONNECTION_TIMEOUT_MS = 15_000;
+const AUDIO_SLICE_MS = 10;
+export class RelayCallTransportError extends Error {
+    code;
+    constructor(message, code) {
+        super(message);
+        this.name = "RelayCallTransportError";
+        if (code !== undefined)
+            this.code = code;
+    }
+}
+const loadWebRTCFactory = async (engine) => {
+    if (engine === "werift") {
+        const { createWeriftWebRTCFactory } = await import("./engine-werift.js");
+        return createWeriftWebRTCFactory();
+    }
+    const wrtc = await import("@roamhq/wrtc");
+    return {
+        createPeerConnection: () => new wrtc.RTCPeerConnection({
+            bundlePolicy: "max-bundle",
+            iceServers: [],
+        }),
+        createAudioSource: () => new wrtc.nonstandard.RTCAudioSource(),
+        createAudioSink: (track) => new wrtc.nonstandard.RTCAudioSink(track),
+    };
+};
+const cloneSamples = (samples) => {
+    const copy = new Int16Array(samples.length);
+    copy.set(samples);
+    return copy;
+};
+const delay = (milliseconds) => new Promise((resolve) => {
+    const timer = setTimeout(resolve, milliseconds);
+    timer.unref?.();
+});
+/**
+ * Provider-neutral Node WebRTC bridge for Relay Call rooms.
+ *
+ * The class owns Relay media negotiation internally. Higher-level adapters only
+ * exchange PCM16 frames and call lifecycle events.
+ */
+export class RelayCallTransport {
+    #room;
+    #providedFactory;
+    #engine;
+    #iceGatheringTimeoutMs;
+    #connectionTimeoutMs;
+    #listeners = new Map();
+    #factory;
+    #peer;
+    #audioSource;
+    #localTrack;
+    #remoteSink;
+    #publishTransceiver;
+    #publishFrame;
+    #initialAnswerSdp;
+    #negotiationTail = Promise.resolve();
+    #outputTail = Promise.resolve();
+    #audioGeneration = 0;
+    #reportedConnected = false;
+    #readySettled = false;
+    #readyResolve;
+    #readyReject;
+    #ready;
+    #handlersAttached = false;
+    #closed = false;
+    constructor(options) {
+        if (!options.callId.trim())
+            throw new Error("callId is required.");
+        this.#room = options.roomClient ?? options.relay.calls.room(options.callId, options.room);
+        this.#providedFactory = options.webRTC;
+        this.#engine = options.engine ?? "werift";
+        if (this.#engine !== "werift" && this.#engine !== "wrtc") {
+            throw new Error('engine must be "werift" or "wrtc".');
+        }
+        this.#iceGatheringTimeoutMs = options.iceGatheringTimeoutMs ?? DEFAULT_ICE_GATHERING_TIMEOUT_MS;
+        this.#connectionTimeoutMs = options.connectionTimeoutMs ?? DEFAULT_CONNECTION_TIMEOUT_MS;
+        if (!Number.isFinite(this.#iceGatheringTimeoutMs) || this.#iceGatheringTimeoutMs <= 0) {
+            throw new Error("iceGatheringTimeoutMs must be greater than zero.");
+        }
+        if (!Number.isFinite(this.#connectionTimeoutMs) || this.#connectionTimeoutMs <= 0) {
+            throw new Error("connectionTimeoutMs must be greater than zero.");
+        }
+        this.#ready = new Promise((resolve, reject) => {
+            this.#readyResolve = resolve;
+            this.#readyReject = reject;
+        });
+        void this.#ready.catch(() => undefined);
+    }
+    on(event, listener) {
+        let listeners = this.#listeners.get(event);
+        if (!listeners) {
+            listeners = new Set();
+            this.#listeners.set(event, listeners);
+        }
+        listeners.add(listener);
+        return this;
+    }
+    off(event, listener) {
+        this.#listeners.get(event)?.delete(listener);
+        return this;
+    }
+    async connect() {
+        if (this.#closed)
+            throw new Error("Relay Call transport is closed.");
+        this.#attachRoomHandlers();
+        await this.#room.connect();
+        if (this.#peer) {
+            await this.#waitForConnection();
+            return;
+        }
+        this.#factory = this.#providedFactory ?? await loadWebRTCFactory(this.#engine);
+        this.#audioSource = this.#factory.createAudioSource();
+        this.#localTrack = this.#audioSource.createTrack();
+        if (this.#localTrack.kind !== "audio") {
+            throw new RelayCallTransportError("The WebRTC binding created a non-audio Relay track.");
+        }
+        const peer = this.#factory.createPeerConnection();
+        this.#peer = peer;
+        this.#publishTransceiver = peer.addTransceiver(this.#localTrack, { direction: "sendonly" });
+        peer.onconnectionstatechange = () => this.#connectionStateChanged();
+        peer.ontrack = (event) => this.#remoteTrack(event.track);
+        try {
+            await this.#publishLocalAudio();
+            await this.#waitForConnection();
+        }
+        catch (error) {
+            this.close();
+            throw error;
+        }
+    }
+    /** Replace only the room signaling socket and replay the identical publication. */
+    async reconnect() {
+        if (this.#closed)
+            throw new Error("Relay Call transport is closed.");
+        if (!this.#publishFrame)
+            throw new Error("Relay Call transport is not connected.");
+        await this.#room.reconnect();
+        this.#room.send(this.#publishFrame);
+    }
+    /**
+     * Feed interleaved PCM16 audio into Relay. Frames are split into 10 ms WebRTC
+     * source slices and paced in real time, so adapters may push larger chunks.
+     */
+    writeAudio(frame) {
+        if (this.#closed)
+            return Promise.reject(new Error("Relay Call transport is closed."));
+        if (!this.#audioSource)
+            return Promise.reject(new Error("Relay Call transport is not connected."));
+        if (!Number.isInteger(frame.sampleRate) || frame.sampleRate <= 0 || frame.sampleRate % 100 !== 0) {
+            return Promise.reject(new Error("Relay audio sampleRate must be a positive multiple of 100."));
+        }
+        if (!Number.isInteger(frame.channelCount) || frame.channelCount <= 0) {
+            return Promise.reject(new Error("Relay audio channelCount must be a positive integer."));
+        }
+        if (frame.samples.length % frame.channelCount !== 0) {
+            return Promise.reject(new Error("Relay audio samples must contain complete interleaved frames."));
+        }
+        const captured = {
+            samples: cloneSamples(frame.samples),
+            sampleRate: frame.sampleRate,
+            channelCount: frame.channelCount,
+        };
+        const generation = this.#audioGeneration;
+        const queued = this.#outputTail.then(() => this.#writeAudio(captured, generation));
+        this.#outputTail = queued.catch(() => undefined);
+        return queued;
+    }
+    /** Drop queued outgoing PCM. At most one already-submitted 10 ms slice remains. */
+    clearAudio() {
+        this.#audioGeneration += 1;
+    }
+    setMuted(muted) {
+        this.#room.userUpdate({ muted });
+    }
+    end() {
+        this.#room.end();
+    }
+    close() {
+        if (this.#closed)
+            return;
+        this.#closed = true;
+        this.#rejectReady(new RelayCallTransportError("Relay Call transport closed before media connected."));
+        this.clearAudio();
+        this.#shutdownMedia();
+        this.#room.close();
+    }
+    async #publishLocalAudio() {
+        const peer = this.#requirePeer();
+        const offer = await peer.createOffer();
+        await peer.setLocalDescription(offer);
+        await this.#waitForIceGathering(peer);
+        const description = this.#localDescription("offer");
+        const mid = this.#publishTransceiver?.mid;
+        if (!mid)
+            throw new RelayCallTransportError("Relay audio publication has no WebRTC MID.");
+        this.#publishFrame = {
+            type: "offer",
+            session_description: description,
+            tracks: [{ mid, name: "audio" }],
+        };
+        this.#room.send(this.#publishFrame);
+    }
+    #attachRoomHandlers() {
+        if (this.#handlersAttached)
+            return;
+        this.#handlersAttached = true;
+        this.#room.on("answer", (frame) => {
+            this.#queueNegotiation(() => this.#serverAnswer(frame));
+        });
+        this.#room.on("offer", (frame) => {
+            this.#queueNegotiation(() => this.#serverOffer(frame));
+        });
+        this.#room.on("roomState", (frame) => {
+            this.#emit("roomState", frame);
+        });
+        this.#room.on("error", (error) => {
+            if (error instanceof Error) {
+                this.#rejectReady(error);
+                this.#emit("error", error);
+            }
+            else
+                this.#serverError(error);
+        });
+        this.#room.on("ended", (frame) => {
+            this.#rejectReady(new RelayCallTransportError(`Relay Call ended before media connected (${frame.reason}).`));
+            this.clearAudio();
+            this.#shutdownMedia();
+            this.#emit("ended", frame);
+        });
+        this.#room.on("close", (event) => {
+            if (!this.#reportedConnected) {
+                this.#rejectReady(new RelayCallTransportError(`Relay Call room closed before media connected (${event.code}).`));
+            }
+            this.#emit("close", event);
+        });
+    }
+    #queueNegotiation(work) {
+        const run = this.#negotiationTail.then(work);
+        this.#negotiationTail = run.catch((error) => {
+            const parsed = error instanceof Error ? error : new Error(String(error));
+            this.#rejectReady(parsed);
+            this.#emit("error", parsed);
+        });
+    }
+    async #serverAnswer(frame) {
+        const peer = this.#requirePeer();
+        const sdp = frame.session_description.sdp;
+        // Reconnecting the signaling socket replays the exact initial offer. Relay
+        // returns its cached answer; applying that answer again in stable state is
+        // invalid WebRTC signaling, so recognize and ignore the replay.
+        if (peer.signalingState === "stable" && this.#initialAnswerSdp === sdp)
+            return;
+        await peer.setRemoteDescription(frame.session_description);
+        this.#initialAnswerSdp ??= sdp;
+    }
+    async #serverOffer(frame) {
+        const peer = this.#requirePeer();
+        await peer.setRemoteDescription(frame.session_description);
+        const answer = await peer.createAnswer();
+        await peer.setLocalDescription(answer);
+        await this.#waitForIceGathering(peer);
+        this.#room.send({ type: "answer", session_description: this.#localDescription("answer") });
+    }
+    #serverError(frame) {
+        const error = new RelayCallTransportError(frame.message, frame.code);
+        this.#rejectReady(error);
+        this.#emit("error", error);
+    }
+    #connectionStateChanged() {
+        const state = this.#peer?.connectionState;
+        if (state === "connected" && !this.#reportedConnected) {
+            this.#reportedConnected = true;
+            try {
+                this.#room.connected();
+                this.#resolveReady();
+                this.#emit("connected");
+            }
+            catch (error) {
+                const parsed = error instanceof Error ? error : new Error(String(error));
+                this.#rejectReady(parsed);
+                this.#emit("error", parsed);
+            }
+        }
+        else if (state === "failed") {
+            const error = new RelayCallTransportError("Relay WebRTC connection failed.");
+            this.#rejectReady(error);
+            this.#emit("error", error);
+        }
+    }
+    #remoteTrack(track) {
+        if (track.kind !== "audio" || !this.#factory)
+            return;
+        this.#remoteSink?.stop();
+        const sink = this.#factory.createAudioSink(track);
+        this.#remoteSink = sink;
+        sink.ondata = (data) => {
+            if (this.#closed || this.#remoteSink !== sink)
+                return;
+            const channelCount = data.channelCount ?? 1;
+            if (data.bitsPerSample !== undefined && data.bitsPerSample !== 16) {
+                this.#emit("error", new RelayCallTransportError(`Relay WebRTC delivered unsupported ${data.bitsPerSample}-bit audio.`));
+                return;
+            }
+            this.#emit("audio", {
+                samples: cloneSamples(data.samples),
+                sampleRate: data.sampleRate,
+                channelCount,
+            });
+        };
+    }
+    async #writeAudio(frame, generation) {
+        const source = this.#audioSource;
+        if (!source || generation !== this.#audioGeneration)
+            return;
+        const samplesPerChannel = frame.sampleRate / 100;
+        const sliceSamples = samplesPerChannel * frame.channelCount;
+        for (let offset = 0; offset < frame.samples.length; offset += sliceSamples) {
+            if (this.#closed || generation !== this.#audioGeneration)
+                return;
+            const remaining = Math.min(sliceSamples, frame.samples.length - offset);
+            const samples = new Int16Array(sliceSamples);
+            samples.set(frame.samples.subarray(offset, offset + remaining));
+            source.onData({
+                samples,
+                sampleRate: frame.sampleRate,
+                bitsPerSample: 16,
+                channelCount: frame.channelCount,
+                numberOfFrames: samplesPerChannel,
+            });
+            await delay(AUDIO_SLICE_MS);
+        }
+    }
+    async #waitForIceGathering(peer) {
+        if (peer.iceGatheringState === "complete")
+            return;
+        await new Promise((resolve, reject) => {
+            const timeout = setTimeout(() => {
+                cleanup();
+                reject(new RelayCallTransportError("Timed out gathering Relay WebRTC ICE candidates."));
+            }, this.#iceGatheringTimeoutMs);
+            timeout.unref?.();
+            const changed = () => {
+                if (peer.iceGatheringState !== "complete")
+                    return;
+                cleanup();
+                resolve();
+            };
+            const cleanup = () => {
+                clearTimeout(timeout);
+                peer.removeEventListener("icegatheringstatechange", changed);
+            };
+            peer.addEventListener("icegatheringstatechange", changed);
+            changed();
+        });
+    }
+    async #waitForConnection() {
+        if (this.#reportedConnected)
+            return;
+        await new Promise((resolve, reject) => {
+            const timeout = setTimeout(() => {
+                reject(new RelayCallTransportError("Timed out connecting Relay WebRTC media."));
+            }, this.#connectionTimeoutMs);
+            timeout.unref?.();
+            this.#ready.then(() => {
+                clearTimeout(timeout);
+                resolve();
+            }, (error) => {
+                clearTimeout(timeout);
+                reject(error);
+            });
+        });
+    }
+    #resolveReady() {
+        if (this.#readySettled)
+            return;
+        this.#readySettled = true;
+        this.#readyResolve?.();
+    }
+    #rejectReady(error) {
+        if (this.#readySettled)
+            return;
+        this.#readySettled = true;
+        this.#readyReject?.(error);
+    }
+    #localDescription(type) {
+        const local = this.#requirePeer().localDescription;
+        if (!local || local.type !== type || !local.sdp) {
+            throw new RelayCallTransportError(`Relay WebRTC did not produce a complete ${type} SDP.`);
+        }
+        return { type, sdp: local.sdp };
+    }
+    #requirePeer() {
+        if (!this.#peer)
+            throw new Error("Relay Call transport is not connected.");
+        return this.#peer;
+    }
+    #shutdownMedia() {
+        this.#remoteSink?.stop();
+        this.#remoteSink = undefined;
+        this.#localTrack?.stop();
+        this.#localTrack = undefined;
+        this.#peer?.close();
+        this.#peer = undefined;
+        this.#audioSource = undefined;
+    }
+    #emit(event, ...args) {
+        for (const listener of this.#listeners.get(event) ?? []) {
+            try {
+                listener(...args);
+            }
+            catch { /* adapters own listener failures */ }
+        }
+    }
+}
+//# sourceMappingURL=transport.js.map

package/dist/transport.js.map

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package/package.json

@@ -0,0 +1,77 @@
+{
+  "name": "@relaymessenger/livekit",
+  "version": "0.1.0-staging.0",
+  "description": "Relay WebRTC audio transport and LiveKit Agents integration for Node.js.",
+  "type": "module",
+  "sideEffects": false,
+  "main": "./dist/index.js",
+  "types": "./dist/index.d.ts",
+  "exports": {
+    ".": {
+      "types": "./dist/index.d.ts",
+      "import": "./dist/index.js",
+      "default": "./dist/index.js"
+    },
+    "./transport": {
+      "types": "./dist/transport.d.ts",
+      "import": "./dist/transport.js",
+      "default": "./dist/transport.js"
+    },
+    "./package.json": "./package.json"
+  },
+  "files": [
+    "dist/",
+    "README.md",
+    "LICENSE"
+  ],
+  "scripts": {
+    "clean": "node -e \"require('node:fs').rmSync('dist',{recursive:true,force:true})\"",
+    "build": "tsc -p tsconfig.build.json",
+    "check": "tsc -p tsconfig.json --noEmit",
+    "test": "vitest run",
+    "validate": "npm run check && npm test && npm run build",
+    "prepack": "npm run validate"
+  },
+  "engines": {
+    "node": ">=22.22.3"
+  },
+  "license": "MIT",
+  "dependencies": {
+    "@evan/opus": "1.0.3",
+    "@relaymessenger/sdk": "0.3.6-staging.14",
+    "werift": "0.24.4"
+  },
+  "optionalDependencies": {
+    "@roamhq/wrtc": "^0.10.0"
+  },
+  "peerDependencies": {
+    "@livekit/agents": "^1.9.0",
+    "@livekit/rtc-node": "^0.13.34"
+  },
+  "devDependencies": {
+    "@livekit/agents": "^1.9.0",
+    "@livekit/rtc-node": "^0.13.34",
+    "@types/node": "^26.6.2",
+    "typescript": "^5.9.3",
+    "vitest": "^4.0.0"
+  },
+  "publishConfig": {
+    "access": "public",
+    "registry": "https://registry.npmjs.org/",
+    "tag": "staging",
+    "provenance": true
+  },
+  "repository": {
+    "type": "git",
+    "url": "git+https://github.com/RelayMessenger/Relay-SDK.git",
+    "directory": "packages/livekit"
+  },
+  "homepage": "https://github.com/RelayMessenger/Relay-SDK/tree/main/packages/livekit#readme",
+  "keywords": [
+    "relay",
+    "livekit",
+    "webrtc",
+    "voice",
+    "agents"
+  ]
+}