@ariaflowagents/livekit-plugin-transport-sip 0.9.0

package/dist/transport_adapter.d.ts.map

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package/dist/transport_adapter.js

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+import { TransportAdapter, } from '@ariaflow/livekit-plugin';
+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;
+    _isOpen = true;
+    constructor(rtpSession, codec, options) {
+        super();
+        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;
+        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.audioOutput = new SIPAudioOutput(rtpSession, codec.sampleRate, packetDurationMs);
+        this.textOutput = new SIPTextOutput();
+    }
+    get isOpen() {
+        return this._isOpen;
+    }
+    async close() {
+        if (!this._isOpen)
+            return;
+        this._isOpen = false;
+        await this.audioInput.close();
+        await this.audioOutput.close();
+        this.rtpSession.close();
+    }
+}
+//# sourceMappingURL=transport_adapter.js.map

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package/dist/types.d.ts

@@ -0,0 +1,46 @@
+/**
+ * SIP transport type.
+ */
+export type SIPTransport = 'udp' | 'websocket';
+/**
+ * Configuration options for SIP server.
+ */
+export interface SIPServerOptions {
+    /**
+     * Transport type:
+     * - 'udp': production RTP telephony in this package
+     * - 'websocket': deprecated here, use @ariaflow/livekit-plugin-transport-sip-jssip
+     * Default: 'udp'
+     */
+    transport?: SIPTransport;
+    /** Port for SIP signaling (UDP transport). Default: 5060. */
+    sipPort?: number;
+    /** Protocol for UDP transport ('udp' or 'tcp'). Default: 'udp'. */
+    sipProtocol?: 'udp' | 'tcp';
+    /** Starting port for RTP media. Each call uses two consecutive ports. Default: 10000. */
+    rtpPortStart?: number;
+    /** Local IP address to advertise in SDP. */
+    localAddress: string;
+    /** Preferred codec. Default: 'PCMU' (G.711 μ-law). */
+    codec?: 'PCMU' | 'PCMA';
+    /**
+     * Send RTP on a fixed wall clock with silence fill when idle (see RtpSession).
+     * Default false.
+     */
+    continuousPacing?: boolean;
+    /** WebSocket server host (for WebSocket transport). Default: localAddress. */
+    wsServerHost?: string;
+    /** WebSocket server port (for WebSocket transport). Default: 8080. */
+    wsServerPort?: number;
+    /** Use secure WebSocket (WSS) instead of WS (for WebSocket transport). Default: true. */
+    secureWebSocket?: boolean;
+    /** SIP username for registration (for WebSocket transport). */
+    sipUsername?: string;
+    /** SIP password for registration (for WebSocket transport). */
+    sipPassword?: string;
+    /** SIP domain for registration (for WebSocket transport). Default: localAddress. */
+    sipDomain?: string;
+    /** Whether to register with SIP server (for WebSocket transport). Default: true. */
+    shouldRegister?: boolean;
+}
+//# sourceMappingURL=types.d.ts.map

package/dist/types.d.ts.map

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package/dist/types.js

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+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

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+{
+  "name": "@ariaflowagents/livekit-plugin-transport-sip",
+  "version": "0.9.0",
+  "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",
+  "types": "dist/index.d.ts",
+  "exports": {
+    ".": {
+      "types": "./dist/index.d.ts",
+      "import": "./dist/index.js"
+    }
+  },
+  "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"
+  },
+  "devDependencies": {
+    "@types/node": "^22.10.0",
+    "typescript": "^5.5.0"
+  },
+  "license": "Apache-2.0"
+}