@ariaflowagents/livekit-plugin-transport-sip-jssip 0.9.0

package/README.md

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+# @ariaflow/livekit-plugin-transport-sip-jssip
+
+SIP over WebSocket/WebRTC signaling utilities built on JsSIP.
+
+Use this package for:
+- Browser or WebRTC SIP endpoints
+- SIP over WebSocket (WS/WSS)
+- JsSIP session lifecycle handling
+
+This package focuses on signaling/session lifecycle. RTP trunk media bridging is provided by:
+- `@ariaflow/livekit-plugin-transport-sip`

package/dist/index.d.ts

@@ -0,0 +1,3 @@
+export { JsSIPSignaling } from './jssip_signaling.js';
+export type { JsSIPSignalingOptions, OnSessionCallback, OnByeCallback, } from './types.js';
+//# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

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package/dist/index.js

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+export { JsSIPSignaling } from './jssip_signaling.js';
+//# sourceMappingURL=index.js.map

package/dist/index.js.map

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+{"version":3,"file":"index.js","sourceRoot":"","sources":["../src/index.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,cAAc,EAAE,MAAM,sBAAsB,CAAC"}

package/dist/jssip_signaling.d.ts

@@ -0,0 +1,24 @@
+import type { RTCSession } from 'jssip/lib/RTCSession.js';
+import type { JsSIPSignalingOptions, OnByeCallback, OnSessionCallback } from './types.js';
+/**
+ * SIP signaling/session lifecycle utility for SIP over WebSocket (JsSIP).
+ *
+ * This class handles SIP signaling and WebRTC session lifecycle only.
+ * Media playout/capture should be handled via the RTCSession APIs.
+ */
+export declare class JsSIPSignaling {
+    private options;
+    private ua;
+    private activeSessions;
+    constructor(options: JsSIPSignalingOptions);
+    start(onSession: OnSessionCallback, onBye?: OnByeCallback): Promise<void>;
+    makeCall(targetUri: string, onSession?: OnSessionCallback, onBye?: OnByeCallback): Promise<string>;
+    hangup(callId: string): Promise<void>;
+    stop(): Promise<void>;
+    getSession(callId: string): RTCSession | undefined;
+    get isRegistered(): boolean;
+    get status(): 'disconnected' | 'connecting' | 'connected';
+    private attachSession;
+    private extractCallId;
+}
+//# sourceMappingURL=jssip_signaling.d.ts.map

package/dist/jssip_signaling.d.ts.map

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package/dist/jssip_signaling.js

@@ -0,0 +1,119 @@
+import { UA, WebSocketInterface } from 'jssip';
+/**
+ * SIP signaling/session lifecycle utility for SIP over WebSocket (JsSIP).
+ *
+ * This class handles SIP signaling and WebRTC session lifecycle only.
+ * Media playout/capture should be handled via the RTCSession APIs.
+ */
+export class JsSIPSignaling {
+    options;
+    ua = null;
+    activeSessions = new Map();
+    constructor(options) {
+        this.options = options;
+    }
+    async start(onSession, onBye) {
+        const wsProtocol = this.options.secureWebSocket === false ? 'ws' : 'wss';
+        const wsUrl = `${wsProtocol}://${this.options.wsServerHost || this.options.localAddress}:${this.options.wsServerPort || 8080}/ws`;
+        const sipUri = `sip:${this.options.sipUsername || 'agent'}@${this.options.sipDomain || this.options.localAddress}`;
+        this.ua = new UA({
+            uri: sipUri,
+            sockets: [new WebSocketInterface(wsUrl)],
+            password: this.options.sipPassword,
+            register: this.options.shouldRegister !== false,
+        });
+        this.ua.on('newRTCSession', (data) => {
+            const session = data.session;
+            const callId = this.extractCallId(session);
+            this.attachSession(callId, session, onBye);
+            onSession(callId, session);
+        });
+        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', (event) => {
+            console.error('[JsSIPSignaling] Registration failed:', event);
+        });
+        this.ua.start();
+    }
+    async makeCall(targetUri, onSession, onBye) {
+        if (!this.ua) {
+            throw new Error('[JsSIPSignaling] User agent not started');
+        }
+        const session = this.ua.call(targetUri, {
+            mediaConstraints: { audio: true, video: false },
+        });
+        const callId = this.extractCallId(session);
+        this.attachSession(callId, session, onBye);
+        onSession?.(callId, session);
+        return callId;
+    }
+    async hangup(callId) {
+        const session = this.activeSessions.get(callId);
+        if (!session) {
+            return;
+        }
+        try {
+            await session.terminate();
+        }
+        finally {
+            this.activeSessions.delete(callId);
+        }
+    }
+    async stop() {
+        if (!this.ua) {
+            return;
+        }
+        const hangupPromises = Array.from(this.activeSessions.values()).map(async (session) => {
+            try {
+                await session.terminate();
+            }
+            catch (error) {
+                console.error('[JsSIPSignaling] Error terminating session during shutdown:', error);
+            }
+        });
+        await Promise.allSettled(hangupPromises);
+        this.activeSessions.clear();
+        this.ua.stop();
+        this.ua = null;
+    }
+    getSession(callId) {
+        return this.activeSessions.get(callId);
+    }
+    get isRegistered() {
+        return this.ua?.isRegistered() ?? false;
+    }
+    get status() {
+        if (!this.ua)
+            return 'disconnected';
+        const uaStatus = this.ua.status;
+        if (uaStatus === 0)
+            return 'connecting';
+        if (uaStatus === 1 || uaStatus === 2)
+            return 'connected';
+        return 'disconnected';
+    }
+    attachSession(callId, session, onBye) {
+        this.activeSessions.set(callId, session);
+        session.on('ended', () => {
+            this.activeSessions.delete(callId);
+            onBye?.(callId);
+        });
+        session.on('failed', (error) => {
+            console.error(`[JsSIPSignaling] Session ${callId} failed:`, error);
+            this.activeSessions.delete(callId);
+            onBye?.(callId);
+        });
+    }
+    extractCallId(session) {
+        const req = session.request;
+        return req?.call_id || req?.callId || `call-${Date.now()}`;
+    }
+}
+//# sourceMappingURL=jssip_signaling.js.map

package/dist/jssip_signaling.js.map

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package/dist/types.d.ts

@@ -0,0 +1,14 @@
+import type { RTCSession } from 'jssip/lib/RTCSession.js';
+export interface JsSIPSignalingOptions {
+    localAddress: string;
+    wsServerHost?: string;
+    wsServerPort?: number;
+    secureWebSocket?: boolean;
+    sipUsername?: string;
+    sipPassword?: string;
+    sipDomain?: string;
+    shouldRegister?: boolean;
+}
+export type OnSessionCallback = (callId: string, session: RTCSession) => void;
+export type OnByeCallback = (callId: string) => void;
+//# sourceMappingURL=types.d.ts.map

package/dist/types.d.ts.map

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package/dist/types.js

@@ -0,0 +1,2 @@
+export {};
+//# sourceMappingURL=types.js.map

package/dist/types.js.map

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+{"version":3,"file":"types.js","sourceRoot":"","sources":["../src/types.ts"],"names":[],"mappings":""}

package/package.json

@@ -0,0 +1,27 @@
+{
+  "name": "@ariaflowagents/livekit-plugin-transport-sip-jssip",
+  "version": "0.9.0",
+  "description": "SIP over WebSocket/WebRTC signaling utilities using JsSIP for LiveKit transport integrations.",
+  "type": "module",
+  "main": "dist/index.js",
+  "types": "dist/index.d.ts",
+  "exports": {
+    ".": {
+      "types": "./dist/index.d.ts",
+      "import": "./dist/index.js"
+    }
+  },
+  "files": ["dist"],
+  "scripts": {
+    "build": "tsc",
+    "clean": "rm -rf dist"
+  },
+  "dependencies": {
+    "jssip": "^3.13.5"
+  },
+  "devDependencies": {
+    "@types/node": "^22.10.0",
+    "typescript": "^5.5.0"
+  },
+  "license": "Apache-2.0"
+}