@ariaflowagents/livekit-plugin-transport-sip 0.9.5 → 0.9.7

package/dist/index.d.ts

@@ -9,4 +9,5 @@ export { PCMU, PCMA, type Codec } from '@ariaflowagents/livekit-plugin/codec/g71
 export { RtpSession, type RtpSessionOptions } from './rtp/rtp_session.js';
 export { parseRtpPacket, buildRtpPacket, type RtpPacket } from './rtp/rtp_packet.js';
 export { JitterBuffer } from './rtp/jitter_buffer.js';
+export { createSipNativeAudioTransport } from './native_bridge.js';
 //# sourceMappingURL=index.d.ts.map

package/dist/index.d.ts.map

@@ -1 +1 @@
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+{"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

@@ -8,4 +8,5 @@ export { PCMU, PCMA } from '@ariaflowagents/livekit-plugin/codec/g711';
 export { RtpSession } from './rtp/rtp_session.js';
 export { parseRtpPacket, buildRtpPacket } from './rtp/rtp_packet.js';
 export { JitterBuffer } from './rtp/jitter_buffer.js';
+export { createSipNativeAudioTransport } from './native_bridge.js';
 //# sourceMappingURL=index.js.map

package/dist/index.js.map

@@ -1 +1 @@
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+{"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/native_bridge.d.ts

@@ -0,0 +1,12 @@
+import type { NativeAudioTransport } from '@ariaflowagents/livekit-plugin';
+import type { RtpSession } from './rtp/rtp_session.js';
+/**
+ * RTP (G.711 payload) session ↔ PCM s16le for the native LiveKit runner.
+ * Decoded RTP PCM is 8 kHz; resampled to the runner rate (default 24 kHz).
+ */
+export declare function createSipNativeAudioTransport(rtpSession: RtpSession, options?: {
+    runnerSampleRate?: number;
+    codecSampleRate?: number;
+    packetDurationMs?: number;
+}): NativeAudioTransport;
+//# sourceMappingURL=native_bridge.d.ts.map

package/dist/native_bridge.d.ts.map

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

@@ -0,0 +1,130 @@
+import { AudioFrame } from '@ariaflowagents/livekit-plugin';
+import { createResampler } from '@ariaflowagents/livekit-plugin/utils/resample';
+const RTP_PCM_RATE = 8000;
+const RUNNER_DEFAULT_RATE = 24000;
+function bytesToInt16Le(bytes) {
+    const n = Math.floor(bytes.byteLength / 2);
+    const out = new Int16Array(n);
+    for (let i = 0; i < n; i++) {
+        out[i] = bytes[i * 2] | (bytes[i * 2 + 1] << 8);
+    }
+    return out;
+}
+/** Copy PCM samples into a plain array (avoids TS 5.9 Int16Array<ArrayBuffer> vs ArrayBufferLike issues). */
+function pcmToNumbers(data) {
+    const len = data.length;
+    const out = new Array(len);
+    for (let i = 0; i < len; i++)
+        out[i] = data[i];
+    return out;
+}
+function appendPcm(dst, data) {
+    for (let i = 0; i < data.length; i++)
+        dst.push(data[i]);
+}
+/**
+ * RTP (G.711 payload) session ↔ PCM s16le for the native LiveKit runner.
+ * Decoded RTP PCM is 8 kHz; resampled to the runner rate (default 24 kHz).
+ */
+export function createSipNativeAudioTransport(rtpSession, options) {
+    const runnerRate = options?.runnerSampleRate ?? RUNNER_DEFAULT_RATE;
+    const codecRate = options?.codecSampleRate ?? RTP_PCM_RATE;
+    const packetDurationMs = options?.packetDurationMs ?? 20;
+    const samplesPerPacket = (codecRate * packetDurationMs) / 1000;
+    const toRunner = createResampler(codecRate, runnerRate);
+    const fromRunner = createResampler(runnerRate, codecRate);
+    const audioHandlers = new Set();
+    const closeHandlers = new Set();
+    let outboundRemainder = [];
+    const emitClose = () => {
+        for (const h of closeHandlers) {
+            try {
+                h();
+            }
+            catch {
+                /* ignore */
+            }
+        }
+    };
+    const onRtpPcm = (pcm) => {
+        const inFrame = new AudioFrame(pcm, codecRate, 1, pcm.length);
+        try {
+            for (const outFrame of toRunner.push(inFrame)) {
+                const samples = pcmToNumbers(outFrame.data);
+                const bytes = new Uint8Array(samples.length * 2);
+                const view = new DataView(bytes.buffer);
+                for (let i = 0; i < samples.length; i++)
+                    view.setInt16(i * 2, samples[i], true);
+                for (const h of audioHandlers) {
+                    try {
+                        h(bytes);
+                    }
+                    catch {
+                        /* ignore */
+                    }
+                }
+            }
+        }
+        catch {
+            /* ignore */
+        }
+    };
+    rtpSession.on('audio', onRtpPcm);
+    const flushOutbound = () => {
+        for (const outFrame of fromRunner.flush()) {
+            appendPcm(outboundRemainder, outFrame.data);
+        }
+        while (outboundRemainder.length >= samplesPerPacket) {
+            const chunk = outboundRemainder.splice(0, samplesPerPacket);
+            rtpSession.sendAudio(new Int16Array(chunk));
+        }
+        if (outboundRemainder.length > 0) {
+            const padded = new Int16Array(samplesPerPacket);
+            padded.set(outboundRemainder);
+            rtpSession.sendAudio(padded);
+            outboundRemainder = [];
+        }
+    };
+    return {
+        sendAudio(data) {
+            let int16;
+            try {
+                int16 = bytesToInt16Le(data);
+            }
+            catch {
+                return;
+            }
+            const inFrame = new AudioFrame(int16, runnerRate, 1, int16.length);
+            try {
+                for (const outFrame of fromRunner.push(inFrame)) {
+                    appendPcm(outboundRemainder, outFrame.data);
+                    while (outboundRemainder.length >= samplesPerPacket) {
+                        const chunk = outboundRemainder.splice(0, samplesPerPacket);
+                        rtpSession.sendAudio(new Int16Array(chunk));
+                    }
+                }
+            }
+            catch {
+                /* ignore */
+            }
+        },
+        onAudio(handler) {
+            audioHandlers.add(handler);
+        },
+        onClose(handler) {
+            closeHandlers.add(handler);
+        },
+        close() {
+            try {
+                flushOutbound();
+            }
+            catch {
+                /* ignore */
+            }
+            rtpSession.off('audio', onRtpPcm);
+            emitClose();
+            rtpSession.close();
+        },
+    };
+}
+//# sourceMappingURL=native_bridge.js.map

package/dist/native_bridge.js.map

@@ -0,0 +1 @@
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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@ariaflowagents/livekit-plugin-transport-sip",
-  "version": "0.9.5",
+  "version": "0.9.7",
   "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",
@@ -19,7 +19,7 @@
   },
   "peerDependencies": {
     "@livekit/agents": ">=1.0.0",
-    "@ariaflowagents/livekit-plugin": "0.9.5"
+    "@ariaflowagents/livekit-plugin": "0.9.7"
   },
   "devDependencies": {
     "@types/node": "^22.10.0",