@chanl/widget-sdk 0.2.0-canary.0

package/dist/voice/transports/webrtc.js

@@ -0,0 +1,318 @@
+"use strict";
+/**
+ * WebRTCTransport — native browser WebRTC for voice calls.
+ *
+ * Audio path:
+ *   Mic → getUserMedia → addTrack(pc) → RTP to server (no base64, no encoding)
+ *   Server → remote audio track → <audio> element (native playback)
+ *
+ * Messages (transcripts, config, events) go through RTCDataChannel.
+ * Signaling uses HTTP POST /offer + /candidates to the voice-bot-sdk.
+ *
+ * Advantages over WebSocket:
+ *   - No base64 encoding overhead (33% bandwidth savings)
+ *   - Native echo cancellation + noise suppression
+ *   - Lower latency (no JSON framing per audio chunk)
+ *   - Browser-native audio playback (no PCMPlayer)
+ */
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.WebRTCTransport = void 0;
+class WebRTCTransport {
+    /**
+     * @param callbacks — transport event callbacks
+     * @param signalingUrl — platform API base URL for signaling relay.
+     *   The transport POSTs to `{signalingUrl}/offer/{interactionId}` and
+     *   `{signalingUrl}/candidates`.
+     */
+    constructor(callbacks, signalingUrl) {
+        this.pc = null;
+        this.dc = null;
+        this.stream = null;
+        this.audioEl = null;
+        this.pingInterval = null;
+        this.micMuted = false;
+        this.audioDeafened = false;
+        this.debug = false;
+        this.callbacks = callbacks;
+        this.signalingUrl = signalingUrl;
+    }
+    // ── ITransport: connect ──────────────────────────────────────────────
+    async connect(config) {
+        this.debug = config.debug ?? false;
+        // The platform /offer relay resolves the room by interactionId, NOT the
+        // room's own id. Use interactionId; fall back to roomId only if absent.
+        // roomState is always present for WebRTCTransport (room-based flow only).
+        const interactionId = config.roomState.interactionId ?? config.roomState.roomId;
+        if (this.debug) {
+            console.debug('🌐 [WebRTCTransport] Connecting, interaction:', interactionId);
+        }
+        // 1. Get microphone
+        this.stream = await navigator.mediaDevices.getUserMedia({
+            audio: {
+                echoCancellation: true,
+                noiseSuppression: true,
+                autoGainControl: true,
+            },
+        });
+        if (this.debug)
+            console.debug('[WebRTCTransport] Mic acquired');
+        // 2. Create peer connection
+        this.pc = new RTCPeerConnection({
+            iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
+        });
+        // Add mic track
+        this.stream.getTracks().forEach((t) => this.pc.addTrack(t, this.stream));
+        // Add receive-only audio transceiver for bot audio
+        this.pc.addTransceiver('audio', { direction: 'recvonly' });
+        // 3. Create data channel for JSON messages
+        this.dc = this.pc.createDataChannel('pipecat');
+        this.dc.onopen = () => {
+            if (this.debug)
+                console.debug('[WebRTCTransport] Data channel open — ready');
+            // Keepalive ping every second
+            this.pingInterval = setInterval(() => {
+                try {
+                    this.dc?.send('ping');
+                }
+                catch { /* ignore */ }
+            }, 1000);
+            // Data channel open = connection ready for WebRTC
+            // (WebSocket gets a separate 'ready' message, but WebRTC is ready when DC opens)
+            this.callbacks.onReady();
+        };
+        this.dc.onmessage = (e) => this.handleDataChannelMessage(e);
+        // 4. Handle remote audio track — play via <audio> element
+        this.pc.ontrack = (e) => {
+            if (this.debug)
+                console.debug('[WebRTCTransport] Remote track:', e.track.kind);
+            if (e.track.kind === 'audio') {
+                if (!this.audioEl) {
+                    this.audioEl = document.createElement('audio');
+                    this.audioEl.autoplay = true;
+                    this.audioEl.setAttribute('playsinline', 'true');
+                    // Detached <audio> elements don't reliably autoplay in Chrome — must
+                    // be in the DOM. Hide off-screen so it never affects layout.
+                    this.audioEl.style.position = 'fixed';
+                    this.audioEl.style.top = '-9999px';
+                    document.body.appendChild(this.audioEl);
+                }
+                this.audioEl.srcObject = e.streams[0];
+                this.audioEl.muted = this.audioDeafened;
+                this.audioEl.play().catch((err) => {
+                    if (this.debug)
+                        console.debug('[WebRTCTransport] audio.play() rejected:', err);
+                });
+            }
+        };
+        // 5. Collect ICE candidates
+        const candidates = [];
+        this.pc.onicecandidate = (e) => {
+            if (e.candidate) {
+                candidates.push({
+                    candidate: e.candidate.candidate,
+                    sdp_mid: e.candidate.sdpMid,
+                    sdp_mline_index: e.candidate.sdpMLineIndex,
+                });
+            }
+        };
+        this.pc.oniceconnectionstatechange = () => {
+            if (this.debug) {
+                console.debug('[WebRTCTransport] ICE:', this.pc?.iceConnectionState);
+            }
+            if (this.pc?.iceConnectionState === 'failed' ||
+                this.pc?.iceConnectionState === 'disconnected') {
+                this.callbacks.onError(new Error(`ICE ${this.pc.iceConnectionState}`));
+            }
+        };
+        // 6. Create SDP offer
+        const offer = await this.pc.createOffer();
+        await this.pc.setLocalDescription(offer);
+        // Wait for ICE gathering (up to 3s)
+        await new Promise((resolve) => {
+            if (this.pc.iceGatheringState === 'complete')
+                return resolve();
+            const onGather = () => {
+                if (this.pc?.iceGatheringState === 'complete') {
+                    this.pc.removeEventListener('icegatheringstatechange', onGather);
+                    resolve();
+                }
+            };
+            this.pc.addEventListener('icegatheringstatechange', onGather);
+            setTimeout(resolve, 3000);
+        });
+        if (this.debug) {
+            console.debug('[WebRTCTransport] ICE candidates:', candidates.length);
+        }
+        // 7. Send offer via platform signaling relay
+        //    POST /api/v1/bot/call/providers/webrtc/offer/{interactionId}
+        const offerResp = await fetch(`${this.signalingUrl}/offer/${interactionId}`, {
+            method: 'POST',
+            headers: { 'Content-Type': 'application/json' },
+            body: JSON.stringify({
+                sdp: this.pc.localDescription.sdp,
+                type: this.pc.localDescription.type,
+            }),
+        });
+        if (!offerResp.ok) {
+            const text = await offerResp.text().catch(() => '');
+            throw new Error(`WebRTC /offer failed [${offerResp.status}]: ${text}`);
+        }
+        const answer = await offerResp.json();
+        if (this.debug) {
+            console.debug('[WebRTCTransport] SDP answer, pc_id:', answer.pc_id);
+        }
+        await this.pc.setRemoteDescription(new RTCSessionDescription({ sdp: answer.sdp, type: answer.type }));
+        // 8. Send ICE candidates via platform relay.
+        //    interactionId is required so the platform can route to the same PCC
+        //    machine that handled /offer (PCC routes by sessionId, NestJS stores
+        //    that mapping on the WebsocketRoom).
+        if (candidates.length > 0) {
+            await fetch(`${this.signalingUrl}/candidates/${interactionId}`, {
+                method: 'POST',
+                headers: { 'Content-Type': 'application/json' },
+                body: JSON.stringify({ pc_id: answer.pc_id, candidates }),
+            });
+            if (this.debug) {
+                console.debug('[WebRTCTransport] ICE candidates sent');
+            }
+        }
+    }
+    // ── ITransport: disconnect ───────────────────────────────────────────
+    disconnect(_code, _reason) {
+        this.destroy();
+        this.callbacks.onClose(_code ?? 1000, _reason ?? 'client-stop');
+    }
+    // ── ITransport: sendMessage ──────────────────────────────────────────
+    sendMessage(message) {
+        if (!this.dc || this.dc.readyState !== 'open') {
+            if (this.debug) {
+                console.warn('[WebRTCTransport] Cannot send: data channel not open');
+            }
+            return;
+        }
+        // Audio messages are N/A for WebRTC (audio goes via RTP track)
+        if (message.type === 'audio')
+            return;
+        this.dc.send(JSON.stringify(message));
+    }
+    // ── ITransport: isConnected ──────────────────────────────────────────
+    isConnected() {
+        return this.dc?.readyState === 'open';
+    }
+    // ── ITransport: audio controls ───────────────────────────────────────
+    muteMic() {
+        if (this.micMuted)
+            return;
+        this.micMuted = true;
+        this.stream?.getAudioTracks().forEach((t) => { t.enabled = false; });
+    }
+    async unmuteMic() {
+        if (!this.micMuted)
+            return;
+        this.micMuted = false;
+        this.stream?.getAudioTracks().forEach((t) => { t.enabled = true; });
+    }
+    deafen() {
+        if (this.audioDeafened)
+            return;
+        this.audioDeafened = true;
+        if (this.audioEl)
+            this.audioEl.muted = true;
+    }
+    undeafen() {
+        if (!this.audioDeafened)
+            return;
+        this.audioDeafened = false;
+        if (this.audioEl)
+            this.audioEl.muted = false;
+    }
+    setVolume(volume) {
+        if (this.audioEl)
+            this.audioEl.volume = Math.max(0, Math.min(1, volume));
+    }
+    clearAudioBuffer() {
+        // No-op for WebRTC — audio is real-time, no buffering to clear
+    }
+    // ── ITransport: destroy ──────────────────────────────────────────────
+    destroy() {
+        if (this.pingInterval) {
+            clearInterval(this.pingInterval);
+            this.pingInterval = null;
+        }
+        if (this.dc) {
+            try {
+                this.dc.close();
+            }
+            catch { /* ignore */ }
+            this.dc = null;
+        }
+        if (this.pc) {
+            this.pc.close();
+            this.pc = null;
+        }
+        if (this.stream) {
+            this.stream.getTracks().forEach((t) => t.stop());
+            this.stream = null;
+        }
+        if (this.audioEl) {
+            this.audioEl.srcObject = null;
+            this.audioEl.remove();
+            this.audioEl = null;
+        }
+    }
+    // ── Private: data channel message handling ───────────────────────────
+    handleDataChannelMessage(event) {
+        const raw = event.data;
+        // Ignore pong keepalive
+        if (raw === 'pong')
+            return;
+        try {
+            const msg = JSON.parse(raw);
+            // Handle RTVI format from pipecat data channel:
+            // {label:"rtvi-ai", type:"user-transcription", data:{text:"...", timestamp:"..."}}
+            if (msg.label === 'rtvi-ai') {
+                const converted = this.convertRTVIMessage(msg);
+                if (converted) {
+                    this.callbacks.onMessage(converted);
+                    return;
+                }
+            }
+            // Pass through to Audial for event routing
+            // (ready was already fired on DC open — skip duplicate)
+            this.callbacks.onMessage(msg);
+        }
+        catch {
+            if (this.debug) {
+                console.debug('[WebRTCTransport] Non-JSON DC message:', String(raw).slice(0, 80));
+            }
+        }
+    }
+    /**
+     * Convert pipecat RTVI-format data channel messages to Audial's message format.
+     *
+     * RTVI sends: {label:"rtvi-ai", type:"user-transcription"|"bot-transcription",
+     *              data:{text:"...", user_id:"", timestamp:"..."}}
+     * Audial expects: {type:"transcript", text:"...", sender:"user"|"assistant", timestamp:123}
+     */
+    convertRTVIMessage(msg) {
+        const type = msg.type;
+        const data = msg.data;
+        if (type === 'user-transcription' || type === 'bot-transcription') {
+            const sender = type === 'user-transcription' ? 'user' : 'assistant';
+            const text = data?.text || '';
+            if (!text)
+                return null;
+            return {
+                type: 'transcript',
+                text,
+                sender,
+                timestamp: Date.now(),
+            };
+        }
+        // config_loaded, error, and other RTVI types — pass through as-is
+        // Audial's generic 'message' event handler picks these up
+        return null;
+    }
+}
+exports.WebRTCTransport = WebRTCTransport;
+//# sourceMappingURL=webrtc.js.map

package/dist/voice/transports/webrtc.js.map

@@ -0,0 +1 @@
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package/dist/voice/transports/websocket.d.ts

@@ -0,0 +1,39 @@
+/**
+ * WebSocketTransport — legacy transport that sends base64-encoded PCM audio
+ * over a JSON WebSocket. Extracted from Audial.ts for the ITransport refactor.
+ *
+ * Audio path:
+ *   Mic → AudioRecorder (WAV → base64) → ws.send({type:'audio', data:...})
+ *   Server → {type:'audio', data: base64} → base64 decode → PCMPlayer
+ *
+ * Messages (transcripts, config, events) share the same WebSocket.
+ */
+import type { ITransport, TransportCallbacks, TransportConnectConfig } from './transport';
+import type { OutboundClientMessage } from '../types';
+export declare class WebSocketTransport implements ITransport {
+    private ws;
+    private recorder;
+    private player;
+    private audioDeafened;
+    private micMuted;
+    private debug;
+    private callbacks;
+    constructor(callbacks: TransportCallbacks);
+    connect(config: TransportConnectConfig): Promise<void>;
+    disconnect(code?: number, reason?: string): void;
+    sendMessage(message: OutboundClientMessage): void;
+    isConnected(): boolean;
+    muteMic(): void;
+    unmuteMic(): Promise<void>;
+    deafen(): void;
+    undeafen(): void;
+    setVolume(volume: number): void;
+    clearAudioBuffer(): void;
+    destroy(): void;
+    private handleMessage;
+    private sendRaw;
+    private initRecorder;
+    private initPlayer;
+    private sendTwilioSimulationMessages;
+}
+//# sourceMappingURL=websocket.d.ts.map

package/dist/voice/transports/websocket.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"websocket.d.ts","sourceRoot":"","sources":["../../../src/voice/transports/websocket.ts"],"names":[],"mappings":"AAAA;;;;;;;;;GASG;AAEH,OAAO,KAAK,EACV,UAAU,EACV,kBAAkB,EAClB,sBAAsB,EACvB,MAAM,aAAa,CAAC;AACrB,OAAO,KAAK,EAEV,qBAAqB,EAItB,MAAM,UAAU,CAAC;AAKlB,qBAAa,kBAAmB,YAAW,UAAU;IACnD,OAAO,CAAC,EAAE,CAA0B;IACpC,OAAO,CAAC,QAAQ,CAA8B;IAC9C,OAAO,CAAC,MAAM,CAA0B;IACxC,OAAO,CAAC,aAAa,CAAS;IAC9B,OAAO,CAAC,QAAQ,CAAS;IACzB,OAAO,CAAC,KAAK,CAAS;IACtB,OAAO,CAAC,SAAS,CAAqB;gBAE1B,SAAS,EAAE,kBAAkB;IAMnC,OAAO,CAAC,MAAM,EAAE,sBAAsB,GAAG,OAAO,CAAC,IAAI,CAAC;IAyD5D,UAAU,CAAC,IAAI,SAAO,EAAE,MAAM,SAAgB,GAAG,IAAI;IAQrD,WAAW,CAAC,OAAO,EAAE,qBAAqB,GAAG,IAAI;IAMjD,WAAW,IAAI,OAAO;IAMtB,OAAO,IAAI,IAAI;IAMT,SAAS,IAAI,OAAO,CAAC,IAAI,CAAC;IAMhC,MAAM,IAAI,IAAI;IAMd,QAAQ,IAAI,IAAI;IAMhB,SAAS,CAAC,MAAM,EAAE,MAAM,GAAG,IAAI;IAI/B,gBAAgB,IAAI,IAAI;IAUxB,OAAO,IAAI,IAAI;IAuBf,OAAO,CAAC,aAAa;IAmCrB,OAAO,CAAC,OAAO;IAYf,OAAO,CAAC,YAAY;IAWpB,OAAO,CAAC,UAAU;IAalB,OAAO,CAAC,4BAA4B;CA4CrC"}

package/dist/voice/transports/websocket.js

@@ -0,0 +1,280 @@
+"use strict";
+/**
+ * WebSocketTransport — legacy transport that sends base64-encoded PCM audio
+ * over a JSON WebSocket. Extracted from Audial.ts for the ITransport refactor.
+ *
+ * Audio path:
+ *   Mic → AudioRecorder (WAV → base64) → ws.send({type:'audio', data:...})
+ *   Server → {type:'audio', data: base64} → base64 decode → PCMPlayer
+ *
+ * Messages (transcripts, config, events) share the same WebSocket.
+ */
+var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
+    if (k2 === undefined) k2 = k;
+    var desc = Object.getOwnPropertyDescriptor(m, k);
+    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
+      desc = { enumerable: true, get: function() { return m[k]; } };
+    }
+    Object.defineProperty(o, k2, desc);
+}) : (function(o, m, k, k2) {
+    if (k2 === undefined) k2 = k;
+    o[k2] = m[k];
+}));
+var __setModuleDefault = (this && this.__setModuleDefault) || (Object.create ? (function(o, v) {
+    Object.defineProperty(o, "default", { enumerable: true, value: v });
+}) : function(o, v) {
+    o["default"] = v;
+});
+var __importStar = (this && this.__importStar) || (function () {
+    var ownKeys = function(o) {
+        ownKeys = Object.getOwnPropertyNames || function (o) {
+            var ar = [];
+            for (var k in o) if (Object.prototype.hasOwnProperty.call(o, k)) ar[ar.length] = k;
+            return ar;
+        };
+        return ownKeys(o);
+    };
+    return function (mod) {
+        if (mod && mod.__esModule) return mod;
+        var result = {};
+        if (mod != null) for (var k = ownKeys(mod), i = 0; i < k.length; i++) if (k[i] !== "default") __createBinding(result, mod, k[i]);
+        __setModuleDefault(result, mod);
+        return result;
+    };
+})();
+var __importDefault = (this && this.__importDefault) || function (mod) {
+    return (mod && mod.__esModule) ? mod : { "default": mod };
+};
+Object.defineProperty(exports, "__esModule", { value: true });
+exports.WebSocketTransport = void 0;
+const audio_recorder_1 = require("../audio-recorder");
+const pcm_player_1 = __importDefault(require("pcm-player"));
+const base64js = __importStar(require("base64-js"));
+class WebSocketTransport {
+    constructor(callbacks) {
+        this.ws = null;
+        this.recorder = null;
+        this.player = null;
+        this.audioDeafened = false;
+        this.micMuted = false;
+        this.debug = false;
+        this.callbacks = callbacks;
+    }
+    // ── ITransport: connect ──────────────────────────────────────────────
+    async connect(config) {
+        this.debug = config.debug ?? false;
+        // roomState is always present for WebSocketTransport (room-based flow only).
+        const wsUrl = config.roomState.websocketUrl;
+        if (!wsUrl) {
+            throw new Error('WebSocketTransport: no websocketUrl in roomState');
+        }
+        if (this.debug) {
+            console.debug('🔌 [WSTransport] Connecting:', wsUrl);
+        }
+        // Init audio components
+        this.initPlayer();
+        this.initRecorder();
+        // Start recording (mic capture)
+        await this.recorder?.start();
+        // Open WebSocket
+        this.ws = new WebSocket(wsUrl);
+        this.ws.binaryType = 'arraybuffer';
+        this.ws.onopen = () => {
+            if (this.debug)
+                console.debug('[WSTransport] WS open');
+            // Send PCC-compatibility simulation messages
+            this.sendTwilioSimulationMessages(config.roomState);
+            // Send audio config start message
+            const start = {
+                type: 'start',
+                input_audio_config: { sampling_rate: 48000, audio_encoding: 'linear16' },
+                output_audio_config: { sampling_rate: 24000, audio_encoding: 'linear16' },
+            };
+            this.sendRaw(start);
+        };
+        this.ws.onmessage = (event) => {
+            this.handleMessage(event);
+        };
+        this.ws.onerror = () => {
+            this.callbacks.onError(new Error('WebSocket error'));
+        };
+        this.ws.onclose = (event) => {
+            if (this.debug) {
+                console.debug('🔌 [WSTransport] WS closed:', { code: event.code, reason: event.reason });
+            }
+            this.callbacks.onClose(event.code, event.reason);
+        };
+    }
+    // ── ITransport: disconnect ───────────────────────────────────────────
+    disconnect(code = 1000, reason = 'client-stop') {
+        if (this.ws) {
+            this.ws.close(code, reason);
+        }
+    }
+    // ── ITransport: sendMessage ──────────────────────────────────────────
+    sendMessage(message) {
+        this.sendRaw(message);
+    }
+    // ── ITransport: isConnected ──────────────────────────────────────────
+    isConnected() {
+        return this.ws?.readyState === WebSocket.OPEN;
+    }
+    // ── ITransport: audio controls ───────────────────────────────────────
+    muteMic() {
+        if (this.micMuted)
+            return;
+        this.micMuted = true;
+        this.recorder?.stop();
+    }
+    async unmuteMic() {
+        if (!this.micMuted)
+            return;
+        this.micMuted = false;
+        await this.recorder?.start();
+    }
+    deafen() {
+        if (this.audioDeafened)
+            return;
+        this.audioDeafened = true;
+        this.player?.volume(0);
+    }
+    undeafen() {
+        if (!this.audioDeafened)
+            return;
+        this.audioDeafened = false;
+        this.player?.volume(1);
+    }
+    setVolume(volume) {
+        this.player?.volume(volume);
+    }
+    clearAudioBuffer() {
+        if (this.player) {
+            this.player.destroy();
+            this.player = null;
+        }
+        this.initPlayer();
+    }
+    // ── ITransport: destroy ──────────────────────────────────────────────
+    destroy() {
+        // Stop recorder and release mic
+        this.recorder?.stop();
+        this.recorder = null;
+        // Destroy player
+        if (this.player) {
+            this.player.destroy();
+            this.player = null;
+        }
+        // Cleanup WebSocket
+        if (this.ws) {
+            this.ws.onopen = null;
+            this.ws.onmessage = null;
+            this.ws.onerror = null;
+            this.ws.onclose = null;
+            this.ws = null;
+        }
+    }
+    // ── Private: message handling ────────────────────────────────────────
+    handleMessage(event) {
+        try {
+            const msg = JSON.parse(event.data);
+            // Handle audio internally — decode base64 and feed to PCMPlayer
+            if (msg.type === 'audio') {
+                if (this.player && !this.audioDeafened) {
+                    try {
+                        const dataBase64 = msg.data;
+                        const byteArray = base64js.toByteArray(dataBase64);
+                        const int16Array = new Int16Array(byteArray.buffer);
+                        this.player.volume(1);
+                        this.player.feed(int16Array);
+                    }
+                    catch (e) {
+                        if (this.debug) {
+                            console.error('[WSTransport] Error processing audio:', e);
+                        }
+                    }
+                }
+            }
+            // Pass ALL messages (including audio) to Audial for event routing
+            this.callbacks.onMessage(msg);
+        }
+        catch (error) {
+            if (this.debug) {
+                console.error('[WSTransport] Error parsing message:', error);
+            }
+            this.callbacks.onError(error instanceof Error ? error : new Error(String(error)));
+        }
+    }
+    // ── Private: send raw JSON ───────────────────────────────────────────
+    sendRaw(message) {
+        if (!this.ws || this.ws.readyState !== WebSocket.OPEN) {
+            if (this.debug) {
+                console.warn('[WSTransport] Cannot send: WS not open');
+            }
+            return;
+        }
+        this.ws.send(JSON.stringify(message));
+    }
+    // ── Private: audio init ──────────────────────────────────────────────
+    initRecorder() {
+        this.recorder = new audio_recorder_1.AudioRecorder({
+            onData: (base64) => {
+                if (this.audioDeafened)
+                    return;
+                if (!this.ws || this.ws.readyState !== WebSocket.OPEN)
+                    return;
+                this.sendRaw({ type: 'audio', data: base64 });
+            },
+            onError: (error) => this.callbacks.onError(error),
+        });
+    }
+    initPlayer() {
+        this.player = new pcm_player_1.default({
+            inputCodec: 'Int16',
+            channels: 1,
+            sampleRate: 24000,
+            flushTime: 175,
+            fftSize: 1024,
+        });
+        this.player.volume(this.audioDeafened ? 0 : 1);
+    }
+    // ── Private: PCC compatibility ───────────────────────────────────────
+    sendTwilioSimulationMessages(roomState) {
+        const customParameters = roomState.connectionParameters;
+        if (!customParameters) {
+            if (this.debug) {
+                console.error('[WSTransport] No connectionParameters — cannot init');
+            }
+            return;
+        }
+        const roomId = roomState.roomId || Date.now().toString();
+        const connectedMessage = {
+            event: 'connected',
+            protocol: 'Call',
+            version: '1.0.0',
+        };
+        const startMessage = {
+            event: 'start',
+            sequenceNumber: '1',
+            start: {
+                accountSid: 'AC' + 'x'.repeat(32),
+                streamSid: `MZ${roomId}`,
+                callSid: `CA${roomId}`,
+                tracks: ['inbound', 'outbound'],
+                mediaFormat: {
+                    encoding: 'audio/x-mulaw',
+                    sampleRate: 8000,
+                    channels: 1,
+                },
+                customParameters,
+            },
+            streamSid: `MZ${roomId}`,
+        };
+        this.sendRaw(connectedMessage);
+        this.sendRaw(startMessage);
+        if (this.debug) {
+            console.debug('[WSTransport] Sent Twilio simulation messages');
+        }
+    }
+}
+exports.WebSocketTransport = WebSocketTransport;
+//# sourceMappingURL=websocket.js.map

package/dist/voice/transports/websocket.js.map

@@ -0,0 +1 @@
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