@ariaflowagents/livekit-plugin-transport-sip 0.9.10 → 0.9.12

package/dist/transport_adapter.js

@@ -3,32 +3,33 @@ import { SIPAudioInput } from './audio_input.js';
 import { SIPAudioOutput } from './audio_output.js';
 import { SIPTextOutput } from './text_output.js';
 export class SIPTransportAdapter extends TransportAdapter {
-    rtpSession;
     id;
     audioInput;
     audioOutput;
     textOutput;
     config;
+    _rtpSession;
     _isOpen = true;
     constructor(rtpSession, codec, options) {
         super();
-        this.rtpSession = rtpSession;
+        this._rtpSession = rtpSession;
         this.id = options.id;
         const packetDurationMs = options.packetDurationMs ?? 20;
-        // SIPAudioInput upsamples decoded G.711 (8kHz) to 16kHz for STT.
-        // Report the actual output rate so AriaFlowVoiceSession forwards
-        // the correct sample rate to the STT engine.
-        const outputSampleRate = 16000;
+        const outputSampleRate = options.outputSampleRate ?? 24000;
         this.config = {
             sampleRate: outputSampleRate,
             numChannels: codec.channels,
             encoding: codec.name === 'PCMU' ? 'mulaw' : 'alaw',
             samplesPerChannel: (outputSampleRate * packetDurationMs) / 1000,
         };
-        this.audioInput = new SIPAudioInput(rtpSession, codec.sampleRate, codec.channels);
+        this.audioInput = new SIPAudioInput(rtpSession, codec.sampleRate, codec.channels, outputSampleRate);
         this.audioOutput = new SIPAudioOutput(rtpSession, codec.sampleRate, packetDurationMs);
         this.textOutput = new SIPTextOutput();
     }
+    /** The underlying RTP session (for native bridge access). */
+    get rtpSession() {
+        return this._rtpSession;
+    }
     get isOpen() {
         return this._isOpen;
     }
@@ -38,7 +39,6 @@ export class SIPTransportAdapter extends TransportAdapter {
         this._isOpen = false;
         await this.audioInput.close();
         await this.audioOutput.close();
-        this.rtpSession.close();
+        this._rtpSession.close();
     }
 }
-//# sourceMappingURL=transport_adapter.js.map

package/dist/types.d.ts

@@ -43,4 +43,3 @@ export interface SIPServerOptions {
     /** Whether to register with SIP server (for WebSocket transport). Default: true. */
     shouldRegister?: boolean;
 }
-//# sourceMappingURL=types.d.ts.map

package/dist/types.js

@@ -1,2 +1 @@
 export {};
-//# sourceMappingURL=types.js.map

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@ariaflowagents/livekit-plugin-transport-sip",
-  "version": "0.9.10",
+  "version": "0.9.12",
   "description": "SIP/RTP transport adapter for LiveKit Agents. Run voice agents over traditional telephony with G.711 codec support.",
   "type": "module",
   "main": "dist/index.js",
@@ -14,23 +14,23 @@
   "files": [
     "dist"
   ],
-  "scripts": {
-    "build": "tsc",
-    "clean": "rm -rf dist",
-    "typecheck": "tsc --noEmit",
-    "test": "bun test test",
-    "test:typecheck": "tsc -p test/tsconfig.json --noEmit"
-  },
   "dependencies": {
     "node.js-sip": "^1.0.9"
   },
   "peerDependencies": {
-    "@ariaflowagents/livekit-plugin": "workspace:*",
-    "@livekit/agents": ">=1.0.0"
+    "@livekit/agents": ">=1.0.0",
+    "@ariaflowagents/livekit-plugin": "0.9.12"
   },
   "devDependencies": {
     "@types/node": "^22.10.0",
     "typescript": "^5.5.0"
   },
-  "license": "Apache-2.0"
-}
+  "license": "Apache-2.0",
+  "scripts": {
+    "build": "tsc",
+    "clean": "rm -rf dist",
+    "typecheck": "tsc --noEmit",
+    "test": "bun test test",
+    "test:typecheck": "tsc -p test/tsconfig.json --noEmit"
+  }
+}

package/dist/audio_input.d.ts.map

@@ -1 +0,0 @@
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package/dist/audio_input.js.map

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package/dist/audio_output.d.ts.map

@@ -1 +0,0 @@
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package/dist/audio_output.js.map

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package/dist/codec/g711.d.ts

@@ -1,17 +0,0 @@
-/**
- * G.711 mu-law and A-law codec implementation.
- *
- * Converts between 16-bit signed linear PCM (AudioFrame format) and
- * 8-bit compressed telephony audio. Stateless per-sample operation.
- */
-export interface Codec {
-    readonly name: string;
-    readonly payloadType: number;
-    readonly sampleRate: number;
-    readonly channels: number;
-    decode(input: Uint8Array): Int16Array;
-    encode(input: Int16Array): Uint8Array;
-}
-export declare const PCMU: Codec;
-export declare const PCMA: Codec;
-//# sourceMappingURL=g711.d.ts.map

package/dist/codec/g711.d.ts.map

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package/dist/codec/g711.js

@@ -1,123 +0,0 @@
-/**
- * G.711 mu-law and A-law codec implementation.
- *
- * Converts between 16-bit signed linear PCM (AudioFrame format) and
- * 8-bit compressed telephony audio. Stateless per-sample operation.
- */
-// --- mu-law ---
-const MULAW_BIAS = 33;
-const MULAW_CLIP = 32635;
-const mulawExpTable = new Uint8Array([
-    0, 0, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 4,
-    4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-    5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7,
-]);
-function mulawEncodeSample(sample) {
-    const sign = (sample >> 8) & 0x80;
-    if (sign !== 0)
-        sample = -sample;
-    if (sample > MULAW_CLIP)
-        sample = MULAW_CLIP;
-    sample += MULAW_BIAS;
-    const exponent = mulawExpTable[(sample >> 7) & 0xff];
-    const mantissa = (sample >> (exponent + 3)) & 0x0f;
-    return ~(sign | (exponent << 4) | mantissa) & 0xff;
-}
-function mulawDecodeSample(mulaw) {
-    mulaw = ~mulaw & 0xff;
-    const sign = mulaw & 0x80;
-    const exponent = (mulaw >> 4) & 0x07;
-    const mantissa = mulaw & 0x0f;
-    let sample = ((mantissa << 3) + MULAW_BIAS) << exponent;
-    sample -= MULAW_BIAS;
-    return sign !== 0 ? -sample : sample;
-}
-// --- A-law ---
-const alawExpTable = new Uint8Array([
-    1, 1, 2, 2, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-    5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
-    6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
-    7, 7, 7,
-]);
-function alawEncodeSample(sample) {
-    const sign = (~sample >> 8) & 0x80;
-    if (sign === 0)
-        sample = -sample;
-    if (sample > 32767)
-        sample = 32767;
-    let compressedByte;
-    if (sample >= 256) {
-        const exponent = alawExpTable[(sample >> 8) & 0x7f];
-        const mantissa = (sample >> (exponent + 3)) & 0x0f;
-        compressedByte = (exponent << 4) | mantissa;
-    }
-    else {
-        compressedByte = sample >> 4;
-    }
-    return (compressedByte ^ (sign ^ 0x55)) & 0xff;
-}
-function alawDecodeSample(alaw) {
-    alaw ^= 0x55;
-    const sign = alaw & 0x80;
-    const exponent = (alaw >> 4) & 0x07;
-    const mantissa = alaw & 0x0f;
-    let sample;
-    if (exponent !== 0) {
-        sample = ((mantissa << 4) + 0x108) << (exponent - 1);
-    }
-    else {
-        sample = (mantissa << 4) + 8;
-    }
-    return sign !== 0 ? sample : -sample;
-}
-export const PCMU = {
-    name: 'PCMU',
-    payloadType: 0,
-    sampleRate: 8000,
-    channels: 1,
-    decode(input) {
-        const output = new Int16Array(input.length);
-        for (let i = 0; i < input.length; i++) {
-            output[i] = mulawDecodeSample(input[i]);
-        }
-        return output;
-    },
-    encode(input) {
-        const output = new Uint8Array(input.length);
-        for (let i = 0; i < input.length; i++) {
-            output[i] = mulawEncodeSample(input[i]);
-        }
-        return output;
-    },
-};
-export const PCMA = {
-    name: 'PCMA',
-    payloadType: 8,
-    sampleRate: 8000,
-    channels: 1,
-    decode(input) {
-        const output = new Int16Array(input.length);
-        for (let i = 0; i < input.length; i++) {
-            output[i] = alawDecodeSample(input[i]);
-        }
-        return output;
-    },
-    encode(input) {
-        const output = new Uint8Array(input.length);
-        for (let i = 0; i < input.length; i++) {
-            output[i] = alawEncodeSample(input[i]);
-        }
-        return output;
-    },
-};
-//# sourceMappingURL=g711.js.map

package/dist/codec/g711.js.map

@@ -1 +0,0 @@
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package/dist/index.d.ts.map

@@ -1 +0,0 @@
-{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,cAAc,EAAE,MAAM,aAAa,CAAC;AAC7C,OAAO,EAAE,YAAY,EAAE,MAAM,oBAAoB,CAAC;AAClD,YAAY,EAAE,gBAAgB,EAAE,YAAY,EAAE,MAAM,YAAY,CAAC;AAEjE,OAAO,EAAE,mBAAmB,EAAE,MAAM,wBAAwB,CAAC;AAC7D,OAAO,EAAE,aAAa,EAAE,MAAM,kBAAkB,CAAC;AACjD,OAAO,EAAE,cAAc,EAAE,MAAM,mBAAmB,CAAC;AACnD,OAAO,EAAE,aAAa,EAAE,MAAM,kBAAkB,CAAC;AAEjD,OAAO,EAAE,IAAI,EAAE,IAAI,EAAE,KAAK,KAAK,EAAE,MAAM,2CAA2C,CAAC;AAEnF,OAAO,EAAE,UAAU,EAAE,KAAK,iBAAiB,EAAE,MAAM,sBAAsB,CAAC;AAC1E,OAAO,EAAE,cAAc,EAAE,cAAc,EAAE,KAAK,SAAS,EAAE,MAAM,qBAAqB,CAAC;AACrF,OAAO,EAAE,YAAY,EAAE,MAAM,wBAAwB,CAAC;AAEtD,OAAO,EAAE,6BAA6B,EAAE,MAAM,oBAAoB,CAAC"}

package/dist/index.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,cAAc,EAAE,MAAM,aAAa,CAAC;AAC7C,OAAO,EAAE,YAAY,EAAE,MAAM,oBAAoB,CAAC;AAGlD,OAAO,EAAE,mBAAmB,EAAE,MAAM,wBAAwB,CAAC;AAC7D,OAAO,EAAE,aAAa,EAAE,MAAM,kBAAkB,CAAC;AACjD,OAAO,EAAE,cAAc,EAAE,MAAM,mBAAmB,CAAC;AACnD,OAAO,EAAE,aAAa,EAAE,MAAM,kBAAkB,CAAC;AAEjD,OAAO,EAAE,IAAI,EAAE,IAAI,EAAc,MAAM,2CAA2C,CAAC;AAEnF,OAAO,EAAE,UAAU,EAA0B,MAAM,sBAAsB,CAAC;AAC1E,OAAO,EAAE,cAAc,EAAE,cAAc,EAAkB,MAAM,qBAAqB,CAAC;AACrF,OAAO,EAAE,YAAY,EAAE,MAAM,wBAAwB,CAAC;AAEtD,OAAO,EAAE,6BAA6B,EAAE,MAAM,oBAAoB,CAAC"}

package/dist/jssip_signaling.d.ts

@@ -1,76 +0,0 @@
-/**
- * SIP Signaling implementation using JsSIP library for WebSocket transport.
- *
- * @experimental
- *
- * WARNING: JsSIP is a WebRTC-only library. Its `session.answer()` API is a
- * WebRTC API and cannot bridge to the raw RTP pipeline used by `SIPAudioInput`
- * and `SIPAudioOutput`. This file is kept for reference but the WebSocket SIP
- * signaling path is architecturally incompatible with the RTP transport.
- *
- * For production use, use `sip_signaling.ts` (UDP-based, compatible with RTP).
- *
- * JsSIP supports:
- * - SIP over WebSocket (secure WSS and non-secure WS)
- * - WebRTC for media (NOT compatible with raw RTP)
- * - Full SIP protocol compliance (INVITE, ACK, BYE, etc.)
- */
-import type { SIPServerOptions } from './types.js';
-/**
- * Callback for incoming INVITE requests.
- */
-export type OnInviteCallback = (callId: string, remoteSdp: string, rtpPort: number) => void;
-/**
- * Callback for BYE requests (remote hangup).
- */
-export type OnByeCallback = (callId: string) => void;
-/**
- * SIP Signaling handler using JsSIP over WebSocket.
- *
- * This implementation uses the JsSIP library for WebSocket-based SIP
- * signaling, commonly used with WebRTC and modern SIP providers.
- */
-export declare class JsSIPSignaling {
-    private options;
-    private ua;
-    private activeSessions;
-    private nextRtpPort;
-    private localAddress;
-    constructor(options: SIPServerOptions);
-    /**
-     * Start the SIP user agent and begin listening for calls over WebSocket.
-     *
-     * @param onInvite - Callback when a new INVITE is received
-     * @param onBye - Callback when a BYE is received (remote hangup)
-     */
-    start(onInvite: OnInviteCallback, onBye?: OnByeCallback): Promise<void>;
-    /**
-     * Handle an incoming INVITE request.
-     */
-    private handleIncomingInvite;
-    /**
-     * Hang up an active call.
-     */
-    hangup(callId: string): Promise<void>;
-    /**
-     * Make an outgoing call.
-     */
-    makeCall(targetUri: string, onInvite?: OnInviteCallback, onBye?: OnByeCallback): Promise<string>;
-    /**
-     * Stop the SIP user agent and hang up all active calls.
-     */
-    stop(): Promise<void>;
-    /**
-     * Get the RTP port allocated for a specific call.
-     */
-    getRtpPort(callId: string): number | undefined;
-    /**
-     * Check if the user agent is registered.
-     */
-    get isRegistered(): boolean;
-    /**
-     * Get the user agent status.
-     */
-    get status(): 'disconnected' | 'connecting' | 'connected';
-}
-//# sourceMappingURL=jssip_signaling.d.ts.map

package/dist/jssip_signaling.d.ts.map

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

@@ -1,255 +0,0 @@
-/**
- * SIP Signaling implementation using JsSIP library for WebSocket transport.
- *
- * @experimental
- *
- * WARNING: JsSIP is a WebRTC-only library. Its `session.answer()` API is a
- * WebRTC API and cannot bridge to the raw RTP pipeline used by `SIPAudioInput`
- * and `SIPAudioOutput`. This file is kept for reference but the WebSocket SIP
- * signaling path is architecturally incompatible with the RTP transport.
- *
- * For production use, use `sip_signaling.ts` (UDP-based, compatible with RTP).
- *
- * JsSIP supports:
- * - SIP over WebSocket (secure WSS and non-secure WS)
- * - WebRTC for media (NOT compatible with raw RTP)
- * - Full SIP protocol compliance (INVITE, ACK, BYE, etc.)
- */
-import { UA, WebSocketInterface } from 'jssip';
-/**
- * SDP offer for G.711 μ-law audio.
- */
-const G711_SDP_OFFER = (localAddress, rtpPort) => `v=0\r\n` +
-    `o=- ${Date.now()} ${Date.now()} IN IP4 ${localAddress}\r\n` +
-    `s=AriaFlow Voice Agent\r\n` +
-    `c=IN IP4 ${localAddress}\r\n` +
-    `t=0 0\r\n` +
-    `m=audio ${rtpPort} RTP/AVP 0 8\r\n` + // 0 = G.711 μ-law (PCMU), 8 = G.711 A-law (PCMA)
-    `a=rtpmap:0 PCMU/8000\r\n` +
-    `a=rtpmap:8 PCMA/8000\r\n` +
-    `a=fmtp:0\r\n` +
-    `a=fmtp:8\r\n` +
-    `a=ptime:20\r\n` +
-    `a=sendrecv\r\n`;
-/**
- * SIP Signaling handler using JsSIP over WebSocket.
- *
- * This implementation uses the JsSIP library for WebSocket-based SIP
- * signaling, commonly used with WebRTC and modern SIP providers.
- */
-export class JsSIPSignaling {
-    options;
-    ua = null;
-    activeSessions = new Map();
-    nextRtpPort;
-    localAddress;
-    constructor(options) {
-        this.options = options;
-        this.localAddress = options.localAddress;
-        this.nextRtpPort = options.rtpPortStart ?? 10000;
-    }
-    /**
-     * Start the SIP user agent and begin listening for calls over WebSocket.
-     *
-     * @param onInvite - Callback when a new INVITE is received
-     * @param onBye - Callback when a BYE is received (remote hangup)
-     */
-    async start(onInvite, onBye) {
-        // Create WebSocket URL
-        const wsProtocol = this.options.secureWebSocket === false ? 'ws' : 'wss';
-        const wsUrl = `${wsProtocol}://${this.options.wsServerHost || this.localAddress}:${this.options.wsServerPort || 8080}/ws`;
-        // Create SIP URI
-        const sipUri = `sip:${this.options.sipUsername || 'agent'}@${this.options.sipDomain || this.localAddress}`;
-        // Create JsSIP User Agent
-        this.ua = new UA({
-            uri: sipUri,
-            sockets: [new WebSocketInterface(wsUrl)],
-            password: this.options.sipPassword,
-            register: this.options.shouldRegister !== false, // Register by default
-        });
-        // Set up event handlers for incoming calls
-        this.ua.on('newRTCSession', (data) => {
-            this.handleIncomingInvite(data.session, onInvite, onBye);
-        });
-        // Set up connection event handlers
-        this.ua.on('connected', () => {
-            console.log(`[JsSIPSignaling] Connected to SIP server: ${wsUrl}`);
-        });
-        this.ua.on('disconnected', () => {
-            console.warn('[JsSIPSignaling] Disconnected from SIP server');
-        });
-        this.ua.on('registered', () => {
-            console.log(`[JsSIPSignaling] Registered as: ${sipUri}`);
-        });
-        this.ua.on('registrationFailed', () => {
-            console.error('[JsSIPSignaling] Registration failed');
-        });
-        // Start the user agent
-        this.ua.start();
-        console.log(`[JsSIPSignaling] Started with WebSocket: ${wsUrl}`);
-    }
-    /**
-     * Handle an incoming INVITE request.
-     */
-    async handleIncomingInvite(session, onInvite, onBye) {
-        const callId = session.request?.callId || `call-${Date.now()}`;
-        const remoteSdp = session.request?.body || '';
-        console.log(`[JsSIPSignaling] Received INVITE for call: ${callId}`);
-        console.log(`[JsSIPSignaling] From: ${session.remote_identity?.uri?.toString() || session}`);
-        // Allocate RTP port for this call
-        const rtpPort = this.nextRtpPort;
-        this.nextRtpPort += 2; // RTP uses even ports
-        // Generate SDP answer with our RTP port
-        const sdpAnswer = G711_SDP_OFFER(this.localAddress, rtpPort);
-        // Set up session state handlers
-        session.on('progress', () => {
-            console.log(`[JsSIPSignaling] Call ${callId} in progress`);
-        });
-        session.on('confirmed', () => {
-            console.log(`[JsSIPSignaling] Call ${callId} established`);
-        });
-        session.on('ended', () => {
-            console.log(`[JsSIPSignaling] Call ${callId} ended`);
-            this.activeSessions.delete(callId);
-            if (onBye) {
-                onBye(callId);
-            }
-        });
-        session.on('failed', (error) => {
-            console.error(`[JsSIPSignaling] Call ${callId} failed:`, error);
-            this.activeSessions.delete(callId);
-        });
-        // Accept the invitation with our SDP
-        try {
-            await session.answer({
-                extraHeaders: ['Content-Type: application/sdp'],
-            });
-            // Track the active session
-            this.activeSessions.set(callId, { session, rtpPort });
-            // Notify the application about the new call
-            onInvite(callId, remoteSdp, rtpPort);
-        }
-        catch (error) {
-            console.error(`[JsSIPSignaling] Error accepting call ${callId}:`, error);
-            try {
-                session.terminate();
-            }
-            catch {
-                // Ignore terminate errors
-            }
-        }
-    }
-    /**
-     * Hang up an active call.
-     */
-    async hangup(callId) {
-        const activeSession = this.activeSessions.get(callId);
-        if (!activeSession) {
-            console.warn(`[JsSIPSignaling] No active session for call: ${callId}`);
-            return;
-        }
-        try {
-            await activeSession.session.terminate();
-            this.activeSessions.delete(callId);
-            console.log(`[JsSIPSignaling] Hung up call: ${callId}`);
-        }
-        catch (error) {
-            console.error(`[JsSIPSignaling] Error hanging up call ${callId}:`, error);
-        }
-    }
-    /**
-     * Make an outgoing call.
-     */
-    async makeCall(targetUri, onInvite, onBye) {
-        if (!this.ua) {
-            throw new Error('[JsSIPSignaling] User agent not started');
-        }
-        // Allocate RTP port for this call
-        const rtpPort = this.nextRtpPort;
-        this.nextRtpPort += 2;
-        // Generate SDP offer with our RTP port
-        const sdpOffer = G711_SDP_OFFER(this.localAddress, rtpPort);
-        // Create the call session
-        const session = this.ua.call(targetUri, {
-            extraHeaders: ['Content-Type: application/sdp'],
-            eventHandlers: {
-                progress: () => {
-                    const callId = session.request?.callId || `call-${Date.now()}`;
-                    console.log(`[JsSIPSignaling] Outgoing call ${callId} in progress`);
-                },
-                confirmed: () => {
-                    const callId = session.request?.callId || `call-${Date.now()}`;
-                    console.log(`[JsSIPSignaling] Outgoing call ${callId} accepted`);
-                    this.activeSessions.set(callId, { session, rtpPort });
-                    if (onInvite) {
-                        onInvite(callId, session.request?.body || '', rtpPort);
-                    }
-                },
-                ended: () => {
-                    const callId = session.request?.callId || `call-${Date.now()}`;
-                    console.log(`[JsSIPSignaling] Outgoing call ${callId} ended`);
-                    this.activeSessions.delete(callId);
-                    if (onBye) {
-                        onBye(callId);
-                    }
-                },
-                failed: (error) => {
-                    const callId = session.request?.callId || `call-${Date.now()}`;
-                    console.error(`[JsSIPSignaling] Outgoing call ${callId} failed:`, error);
-                    this.activeSessions.delete(callId);
-                },
-            },
-        });
-        return session.request?.callId || `call-${Date.now()}`;
-    }
-    /**
-     * Stop the SIP user agent and hang up all active calls.
-     */
-    async stop() {
-        if (!this.ua) {
-            return;
-        }
-        // Hang up all active calls
-        const hangupPromises = Array.from(this.activeSessions.entries()).map(async ([callId, { session }]) => {
-            try {
-                await session.terminate();
-            }
-            catch (error) {
-                console.error(`[JsSIPSignaling] Error hanging up call ${callId} during shutdown:`, error);
-            }
-        });
-        await Promise.allSettled(hangupPromises);
-        this.activeSessions.clear();
-        // Stop the user agent
-        this.ua.stop();
-        this.ua = null;
-        console.log('[JsSIPSignaling] Stopped');
-    }
-    /**
-     * Get the RTP port allocated for a specific call.
-     */
-    getRtpPort(callId) {
-        return this.activeSessions.get(callId)?.rtpPort;
-    }
-    /**
-     * Check if the user agent is registered.
-     */
-    get isRegistered() {
-        return this.ua?.isRegistered() ?? false;
-    }
-    /**
-     * Get the user agent status.
-     */
-    get status() {
-        if (!this.ua)
-            return 'disconnected';
-        const uaStatus = this.ua.status;
-        // JsSIP UA status: 0 = Init, 1 = Ready, 2 = UserRegistered, 3 = Unregistered
-        if (uaStatus === 0)
-            return 'connecting';
-        if (uaStatus === 1 || uaStatus === 2)
-            return 'connected';
-        return 'disconnected';
-    }
-}
-//# sourceMappingURL=jssip_signaling.js.map

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package/dist/native_bridge.d.ts.map

@@ -1 +0,0 @@
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