asterisk_to_openai_rt/asterisk_to_openai_rt.js at main · infinitocloud/asterisk_to_openai_rt

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// Import required Node.js modules

const ari = require('ari-client'); // Asterisk REST Interface client

const WebSocket = require('ws'); // WebSocket library for OpenAI real-time API

const fs = require('fs'); // File system module for saving audio files

const dgram = require('dgram'); // UDP datagram for RTP audio streaming

const winston = require('winston'); // Logging library

const chalk = require('chalk'); // Colorizes console output

const async = require('async'); // Async utilities (used for RTP queue)

require('dotenv').config(); // Loads environment variables from .env file

// Configuration constants loaded from environment variables or defaults

const ARI_URL = 'http://127.0.0.1:8088'; // Asterisk ARI endpoint

const ARI_USER = 'asterisk'; // ARI username

const ARI_PASS = 'asterisk'; // ARI password

const ARI_APP = 'stasis_app'; // Stasis application name

const OPENAI_API_KEY = process.env.OPENAI_API_KEY; // OpenAI API key from .env

const REALTIME_URL = 'wss://api.openai.com/v1/realtime?model=gpt-4o-realtime-preview-2024-12-17'; // OpenAI real-time WebSocket URL

const RTP_PORT = 12000; // Local port for RTP audio reception

const MAX_CALL_DURATION = process.env.MAX_CALL_DURATION ? parseInt(process.env.MAX_CALL_DURATION) : 300000; // Max call duration in ms (default: 5 min)

const RTP_QUEUE_CONCURRENCY = parseInt(process.env.RTP_QUEUE_CONCURRENCY) || 50; // Concurrent RTP packet sends

const LOG_RTP_EVERY_N_PACKETS = parseInt(process.env.LOG_RTP_EVERY_N_PACKETS) || 100; // Log RTP stats every N packets

const ENABLE_RTP_LOGGING = process.env.ENABLE_RTP_LOGGING === 'true'; // Enable detailed RTP logging

const ENABLE_SENT_TO_OPENAI_RECORDING = process.env.ENABLE_SENT_TO_OPENAI_RECORDING === 'true'; // Controls saving of both .raw and .wav files

const VAD_THRESHOLD = process.env.VAD_THRESHOLD ? parseFloat(process.env.VAD_THRESHOLD) : 0.1; // Voice Activity Detection threshold

const VAD_PREFIX_PADDING_MS = process.env.VAD_PREFIX_PADDING_MS ? parseInt(process.env.VAD_PREFIX_PADDING_MS) : 300; // VAD prefix padding in ms

const VAD_SILENCE_DURATION_MS = process.env.VAD_SILENCE_DURATION_MS ? parseInt(process.env.VAD_SILENCE_DURATION_MS) : 500; // VAD silence duration in ms

const TARGET_RMS = 0.15; // Target Root Mean Square for audio normalization

const MIN_RMS = 0.001; // Minimum RMS to apply gain

// Counters for client/server event logging

let sentEventCounter = 0; // Tracks sent events to OpenAI

let receivedEventCounter = -1; // Tracks received events from OpenAI

// Configure Winston logger with timestamp and colorized output

const logger = winston.createLogger({

level: 'info', // Log level

format: winston.format.combine(

winston.format.timestamp(), // Add timestamp to logs

winston.format.printf(({ timestamp, level, message }) => {

const [origin] = message.split(' ', 1); // Extract message origin (Client/Server)

let counter;

let coloredMessage;

if (origin === '[Client]') {

counter = `C-${sentEventCounter.toString().padStart(4, '0')}`; // Client event counter

sentEventCounter++;

coloredMessage = chalk.cyanBright(message); // Cyan for client messages

} else if (origin === '[Server]') {

counter = `S-${receivedEventCounter.toString().padStart(4, '0')}`; // Server event counter

receivedEventCounter++;

coloredMessage = chalk.yellowBright(message); // Yellow for server messages

} else {

counter = 'N/A'; // No counter for general logs

coloredMessage = chalk.gray(message); // Gray for general logs

}

return `${counter} | ${timestamp} [${level.toUpperCase()}] ${coloredMessage}`; // Formatted log line

})

transports: [new winston.transports.Console()] // Output logs to console

});

// Helper functions for logging OpenAI events

const logClient = (msg) => logger.info(`[Client] ${msg}`); // Log client-side OpenAI events

const logServer = (msg) => logger.info(`[Server] ${msg}`); // Log server-side OpenAI events

// Maps to track channel states and audio buffers

const extMap = new Map(); // Maps ExternalMedia channels to their bridges and SIP channels

const sipMap = new Map(); // Maps SIP channels to their WebSocket and bridge data

const rtpSender = dgram.createSocket('udp4'); // Single UDP socket for sending RTP packets

let rtpReceiver = dgram.createSocket('udp4'); // UDP socket for receiving RTP packets

let ariClient; // ARI client instance

const audioFromAsteriskMap = new Map(); // Buffers audio received from Asterisk

const audioToOpenAIMap = new Map(); // Buffers audio sent to OpenAI

const amplificationLogFrequency = new Map(); // Tracks last amplification log time per channel

const rmsLogFrequency = new Map(); // Tracks last RMS log time per channel

const rtpSentStats = new Map(); // Tracks RTP stats per channel

// Add an ExternalMedia channel to a bridge with retry logic

async function addExtToBridge(client, channel, bridgeId, retries = 5, delay = 500) {

try {

const bridge = await client.bridges.get({ bridgeId }); // Fetch bridge by ID

if (!bridge) throw new Error('Bridge not found');

await bridge.addChannel({ channel: channel.id }); // Add channel to bridge

logger.info(`ExternalMedia channel ${channel.id} added to bridge ${bridgeId}`);

} catch (err) {

if (retries) {

logger.info(`Retrying to add ExternalMedia channel ${channel.id} to bridge ${bridgeId} (${retries} attempts remaining)`);

await new Promise(r => setTimeout(r, delay)); // Wait before retrying

return addExtToBridge(client, channel, bridgeId, retries - 1, delay); // Recursive retry

}

logger.error(`Error adding ExternalMedia channel ${channel.id} to bridge ${bridgeId}: ${err.message}`);

}

// Start the RTP receiver to listen for audio from Asterisk

function startRTPReceiver() {

let packetCount = 0; // Count of received RTP packets

let totalBytes = 0; // Total bytes received

let startTime = Date.now(); // Start time for rate calculation

const audioBuffers = new Map(); // Temporary audio buffers per channel

const BUFFER_INTERVAL_MS = 200; // Interval to process audio chunks (ms)

rtpReceiver.on('listening', () => {

const address = rtpReceiver.address();

logger.info(`RTP Receiver listening on ${address.address}:${address.port}`);

});

// Handle incoming RTP packets

rtpReceiver.on('message', (msg, rinfo) => {

packetCount++;

totalBytes += msg.length;

if (packetCount >= 100) { // Log stats every 100 packets

const currentTime = Date.now();

const duration = (currentTime - startTime) / 1000;

const rate = (packetCount / duration).toFixed(2);

logger.info(`Received ${packetCount} RTP packets from ${rinfo.address}:${rinfo.port}, total bytes: ${totalBytes}, rate: ${rate} packets/s`);

packetCount = 0;

totalBytes = 0;

startTime = currentTime;

}

// Find channel ID based on RTP source

const channelId = [...sipMap.entries()].find(([_, data]) => data.rtpSource && data.rtpSource.address === rinfo.address && data.rtpSource.port === rinfo.port)?.[0];

if (channelId) {

const muLawData = msg.slice(12); // Extract μ-law payload (skip RTP header)

if (!audioFromAsteriskMap.has(channelId)) audioFromAsteriskMap.set(channelId, Buffer.alloc(0));

audioFromAsteriskMap.set(channelId, Buffer.concat([audioFromAsteriskMap.get(channelId), muLawData])); // Append to Asterisk audio buffer

const pcmBuffer24kHz = muLawToPcm24kHz(muLawData, channelId); // Convert to PCM 24kHz

if (!audioBuffers.has(channelId)) audioBuffers.set(channelId, Buffer.alloc(0));

audioBuffers.set(channelId, Buffer.concat([audioBuffers.get(channelId), pcmBuffer24kHz])); // Append to temporary buffer

// Set up interval to send audio to OpenAI

if (!sipMap.get(channelId).sendTimeout) {

sipMap.get(channelId).sendTimeout = setInterval(() => {

const buffer = audioBuffers.get(channelId);

if (buffer && buffer.length > 0) {

let sumSquares = 0;

for (let i = 0; i < buffer.length / 2; i++) { // Calculate RMS

const sample = buffer.readInt16LE(i * 2);

sumSquares += sample * sample;

}

const rms = Math.sqrt(sumSquares / (buffer.length / 2)) / 32768;

const now = Date.now();

if (rms < TARGET_RMS && rms > MIN_RMS) { // Normalize audio if RMS is low

const gain = Math.min(TARGET_RMS / rms, 2);

for (let i = 0; i < buffer.length / 2; i++) {

let sample = buffer.readInt16LE(i * 2);

sample = Math.round(sample * gain);

sample = Math.max(-32768, Math.min(32767, sample));

buffer.writeInt16LE(sample, i * 2);

}

if (!rmsLogFrequency.has(channelId) || now - rmsLogFrequency.get(channelId) >= 2000) {

logger.info(`Adjusted RMS from ${rms.toFixed(3)} to ~${TARGET_RMS} with gain ${gain.toFixed(2)} for channel ${channelId}`);

rmsLogFrequency.set(channelId, now);

}

const base64Audio = buffer.toString('base64'); // Convert to base64 for OpenAI

const channelData = sipMap.get(channelId);

if (channelData && channelData.ws && channelData.ws.readyState === WebSocket.OPEN) {

channelData.ws.send(JSON.stringify({ type: 'input_audio_buffer.append', audio: base64Audio })); // Send to OpenAI

if (!rmsLogFrequency.has(channelId) || now - rmsLogFrequency.get(channelId) >= 2000) {

logClient(`Sending audio chunk to OpenAI for channel ${channelId} | Size: ${(buffer.length / 1024).toFixed(2)} KB | RMS: ${rms.toFixed(3)}`);

rmsLogFrequency.set(channelId, now);

}

audioBuffers.set(channelId, Buffer.alloc(0)); // Clear buffer after sending

}

}, BUFFER_INTERVAL_MS);

}

});

rtpReceiver.on('error', (err) => {

logger.error(`RTP Receiver error: ${err.message}`);

});

rtpReceiver.bind(RTP_PORT, '127.0.0.1'); // Bind to local port

}

// Convert a single μ-law sample to 16-bit PCM

function muLawToPcm16(muLaw) {

muLaw = ~muLaw & 0xFF; // Invert bits and mask

const sign = (muLaw & 0x80) ? -1 : 1; // Extract sign

const exponent = (muLaw & 0x70) >> 4; // Extract exponent

const mantissa = muLaw & 0x0F; // Extract mantissa

let sample = (exponent === 0) ? (mantissa * 8 + 16) : (1 << (exponent + 3)) * (mantissa + 16) - 128; // Decode sample

sample = sign * sample;

return Math.max(-32768, Math.min(32767, sample)); // Clamp to 16-bit range

}

// Convert μ-law buffer to 24kHz PCM with interpolation

function muLawToPcm24kHz(muLawBuffer, channelId) {

const pcm8kHz = Buffer.alloc(muLawBuffer.length * 2); // Buffer for 8kHz PCM

let maxSampleBefore = 0; // Track max sample before clamping

let maxSampleAfter = 0; // Track max sample after clamping

// Convert μ-law to 8kHz PCM

for (let i = 0; i < muLawBuffer.length; i++) {

let sample = muLawToPcm16(muLawBuffer[i]);

maxSampleBefore = Math.max(maxSampleBefore, Math.abs(sample));

sample = Math.round(sample);

sample = Math.max(-32768, Math.min(32767, sample));

maxSampleAfter = Math.max(maxSampleAfter, Math.abs(sample));

pcm8kHz.writeInt16LE(sample, i * 2);

}

// Upsample to 24kHz with linear interpolation

const pcm24kHz = Buffer.alloc(muLawBuffer.length * 3 * 2);

let sumSquares = 0;

for (let i = 0; i < muLawBuffer.length; i++) {

const sample = pcm8kHz.readInt16LE(i * 2);

const prevSample = i > 0 ? pcm8kHz.readInt16LE((i - 1) * 2) : sample;

const nextSample = i < muLawBuffer.length - 1 ? pcm8kHz.readInt16LE((i + 1) * 2) : sample;

const interp1 = Math.round((prevSample * 0.5 + sample * 0.5)); // First interpolated sample

const interp2 = Math.round((sample * 0.75 + nextSample * 0.25)); // Second interpolated sample

pcm24kHz.writeInt16LE(prevSample, (i * 3) * 2);

pcm24kHz.writeInt16LE(interp1, (i * 3 + 1) * 2);

pcm24kHz.writeInt16LE(interp2, (i * 3 + 2) * 2);

sumSquares += prevSample * prevSample + interp1 * interp1 + interp2 * interp2; // For RMS calculation

}

const rms = Math.sqrt(sumSquares / (muLawBuffer.length * 3)) / 32768; // Calculate RMS

if (!audioToOpenAIMap.has(channelId)) audioToOpenAIMap.set(channelId, Buffer.alloc(0));

audioToOpenAIMap.set(channelId, Buffer.concat([audioToOpenAIMap.get(channelId), pcm24kHz])); // Append to OpenAI buffer

const now = Date.now();

if (!amplificationLogFrequency.has(channelId) || now - amplificationLogFrequency.get(channelId) >= 2000) {

logger.info(`Audio processed for channel ${channelId} | RMS: ${rms.toFixed(3)} | Max sample before: ${maxSampleBefore}, after: ${maxSampleAfter}`);

amplificationLogFrequency.set(channelId, now); // Update log frequency

}

return pcm24kHz;

}

// Save PCM data as a WAV file

function saveWavFile(pcmData, filename, sampleRate) {

const bitsPerSample = 16; // 16-bit audio

const channels = 1; // Mono

const byteRate = sampleRate * channels * (bitsPerSample / 8);

const blockAlign = channels * (bitsPerSample / 8);

const dataSize = pcmData.length;

const fileSize = 36 + dataSize;

const buffer = Buffer.alloc(44 + dataSize); // WAV header + data

buffer.write('RIFF', 0);

buffer.writeUInt32LE(fileSize, 4);

buffer.write('WAVE', 8);

buffer.write('fmt ', 12);

buffer.writeUInt32LE(16, 16); // Subchunk size

buffer.writeUInt16LE(1, 20); // PCM format

buffer.writeUInt16LE(channels, 22);

buffer.writeUInt32LE(sampleRate, 24);

buffer.writeUInt32LE(byteRate, 28);

buffer.writeUInt16LE(blockAlign, 32);

buffer.writeUInt16LE(bitsPerSample, 34);

buffer.write('data', 36);

buffer.writeUInt32LE(dataSize, 40);

pcmData.copy(buffer, 44); // Copy PCM data

fs.writeFileSync(filename, buffer);

logger.info(`Saved audio as ${filename}`);

}

// Save raw μ-law data to a file

function saveRawFile(data, filename) {

fs.writeFileSync(filename, data);

logger.info(`Saved raw μ-law as ${filename}`);

}

// Convert 16-bit PCM sample to μ-law

function pcm16ToMuLaw(sample) {

const MAX = 32767;

const MU = 255;

const BIAS = 33;

sample = Math.max(-MAX, Math.min(MAX, sample)); // Clamp to 16-bit range

const sign = sample < 0 ? 0x80 : 0;

let absSample = Math.abs(sample);

if (absSample < 50) return 0x7F; // Silence threshold

absSample += BIAS;

const normalized = absSample / MAX;

const muLaw = Math.log(1 + MU * normalized) / Math.log(1 + MU); // μ-law compression

const quantized = Math.round(muLaw * 128);

const exponent = Math.min(Math.floor(quantized / 16), 7);

const mantissa = Math.min((quantized - (exponent * 16)), 15) & 0x0F;

return ~(sign | (exponent << 4) | mantissa) & 0xFF; // Invert bits

}

// Resample 24kHz PCM to 8kHz

function resamplePcm24kHzTo8kHz(pcm24kHz) {

const inSampleRate = 24000;

const outSampleRate = 8000;

const inSamples = pcm24kHz.length / 2;

const outSamples = Math.floor(inSamples * outSampleRate / inSampleRate);

const pcm8kHz = Buffer.alloc(outSamples * 2);

for (let i = 0; i < outSamples; i++) {

const srcPos = i * inSampleRate / outSampleRate;

const srcIndex = Math.floor(srcPos);

const frac = srcPos - srcIndex;

if (srcIndex + 1 < inSamples) {

const sample1 = pcm24kHz.readInt16LE(srcIndex * 2);

const sample2 = pcm24kHz.readInt16LE((srcIndex + 1) * 2);

const interpSample = Math.round(sample1 + frac * (sample2 - sample1)); // Linear interpolation

pcm8kHz.writeInt16LE(interpSample, i * 2);

} else if (srcIndex < inSamples) {

pcm8kHz.writeInt16LE(pcm24kHz.readInt16LE(srcIndex * 2), i * 2);

}

return pcm8kHz;

}

// Convert PCM buffer to μ-law, optionally resampling from 24kHz to 8kHz

function pcmToMuLaw(pcmBuffer, resample = false) {

const input = resample ? resamplePcm24kHzTo8kHz(pcmBuffer) : pcmBuffer;

const muLawBuffer = Buffer.alloc(input.length / 2);

const chunkSize = 1024;

for (let i = 0; i < input.length / 2; i += chunkSize) {

const end = Math.min(i + chunkSize, input.length / 2);

for (let j = i; j < end; j++) {

let sample = input.readInt16LE(j * 2);

sample = Math.max(-32767, Math.min(32767, Math.floor(sample * 0.95))); // Apply slight attenuation

muLawBuffer[j] = pcm16ToMuLaw(sample);

}

return muLawBuffer;

}

// Build RTP header for a packet

function buildRTPHeader(seq, timestamp, ssrc) {

const header = Buffer.alloc(12);

header[0] = 0x80; // Version 2, no padding, no extension

header[1] = 0x00; // Payload type (0 for μ-law)

header.writeUInt16BE(seq, 2); // Sequence number

header.writeUInt32BE(timestamp, 4); // Timestamp

header.writeUInt32BE(ssrc, 8); // Synchronization source

return header;

}

// Async queue for sending RTP packets

const rtpQueue = async.queue((task, callback) => {

rtpSender.send(task.packet, task.port, task.address, callback);

}, RTP_QUEUE_CONCURRENCY);

// Send an RTP packet with μ-law data

async function sendAudioPacket(muLawData, port, address, seq, timestamp, ssrc) {

const startTime = process.hrtime.bigint();

const header = buildRTPHeader(seq, timestamp, ssrc);

const rtpPacket = Buffer.concat([header, muLawData]);

await new Promise((resolve, reject) => {

rtpQueue.push({ packet: rtpPacket, port, address }, (err) => {

const elapsedMs = Number(process.hrtime.bigint() - startTime) / 1e6;

if (ENABLE_RTP_LOGGING && seq % LOG_RTP_EVERY_N_PACKETS === 0) {

logger.info(`Sent packet seq=${seq}, timestamp=${timestamp}, elapsed=${elapsedMs.toFixed(2)}ms`);

}

if (err) {

logger.error(`Error sending RTP packet: ${err.message}`);

reject(err);

} else {

resolve();

}

});

}

// Stream audio to Asterisk via RTP

const MAX_BUFFER_SIZE = 1024 * 1024; // Max buffer size (1MB)

async function streamAudio(channelId, rtpSource, initialBuffer = Buffer.alloc(0)) {

const samplesPerPacket = 80; // 10 ms at 8000 Hz

const packetIntervalNs = BigInt(10 * 1e6); // 10 ms in nanoseconds

const { address, port } = rtpSource;

logger.info(`Initializing RTP stream to ${address}:${port} for channel ${channelId}`);

let rtpSequence = Math.floor(Math.random() * 65535); // Random initial sequence

let rtpTimestamp = 0; // Initial timestamp

const rtpSSRC = Math.floor(Math.random() * 4294967295); // Random SSRC

let streamStartTime = process.hrtime.bigint();

let isStreaming = true;

let totalBytesSent = 0;

let totalPacketsSent = 0;

let stopRequested = false;

let lastBufferSize = 0; // Previous buffer size

let wasSending = false; // Track if we were sending data

let muLawBuffer = Buffer.alloc(0); // Buffer for μ-law data

let offset = 0; // Offset in buffer

if (!rtpSentStats.has(channelId)) {

rtpSentStats.set(channelId, { packets: 0, bytes: 0, startTime: null }); // Initialize stats

}

// Send a batch of RTP packets

const sendPackets = async (data, packetCount, isSilence = false) => {

let blockStartTime = process.hrtime.bigint();

let nextPacketTime = blockStartTime;

for (let i = 0; i < packetCount && !stopRequested; i++) {

const bytesToSend = Math.min(samplesPerPacket, data.length - (i * samplesPerPacket));

const packetData = data.slice(i * samplesPerPacket, i * samplesPerPacket + bytesToSend);

const packetDataPadded = bytesToSend < samplesPerPacket ? Buffer.concat([packetData, Buffer.alloc(samplesPerPacket - bytesToSend, 0x7F)]) : packetData;

await sendAudioPacket(packetDataPadded, port, address, rtpSequence, rtpTimestamp, rtpSSRC);

if (i === 0 && !streamStartTime) streamStartTime = process.hrtime.bigint();

rtpSequence = (rtpSequence + 1) % 65536;

rtpTimestamp += 80;

totalBytesSent += packetDataPadded.length;

totalPacketsSent += 1;

const stats = rtpSentStats.get(channelId);

stats.packets += 1;

stats.bytes += packetDataPadded.length;

if (!stats.startTime) stats.startTime = Date.now();

nextPacketTime += packetIntervalNs;

const now = process.hrtime.bigint();

if (now < nextPacketTime) {

const delayMs = Number(nextPacketTime - now) / 1e6;

await new Promise(resolve => setTimeout(resolve, delayMs)); // Maintain timing

}

};

const silencePacket = Buffer.alloc(samplesPerPacket, 0x7F); // Silence packet

await sendPackets(silencePacket, 10, true); // Send initial silence

logger.info(`RTP stream fully initialized for channel ${channelId}`);

// Process PCM chunks into μ-law

const processFallback = async (pcmChunk) => {

const muLawData = pcmToMuLaw(pcmChunk, true);

muLawBuffer = Buffer.concat([muLawBuffer, muLawData]);

if (muLawBuffer.length > MAX_BUFFER_SIZE) {

muLawBuffer = muLawBuffer.slice(muLawBuffer.length - MAX_BUFFER_SIZE); // Trim buffer

}

};

// Main streaming loop

const streamLoop = async () => {

while (isStreaming && !stopRequested) {

if (!sipMap.has(channelId)) {

logger.info(`Channel ${channelId} no longer active, stopping RTP stream`);

break;

}

const currentBufferSize = muLawBuffer.length - offset;

if (currentBufferSize >= samplesPerPacket) {

const packetCount = Math.floor(currentBufferSize / samplesPerPacket);

await sendPackets(muLawBuffer.slice(offset, offset + packetCount * samplesPerPacket), packetCount);

offset += packetCount * samplesPerPacket;

if (muLawBuffer.length - offset > MAX_BUFFER_SIZE / 2) {

muLawBuffer = muLawBuffer.slice(offset);

offset = 0; // Reset offset

}

wasSending = true;

} else if (wasSending && currentBufferSize < samplesPerPacket) {

logger.info(`RTP buffer to Asterisk fully sent for channel ${channelId} | Remaining: ${currentBufferSize} bytes`);

wasSending = false;

}

lastBufferSize = currentBufferSize;

await new Promise(resolve => setImmediate(resolve)); // Yield control

}

const totalDuration = Number(process.hrtime.bigint() - streamStartTime) / 1e9;

logger.info(`Finished RTP stream for channel ${channelId} | Total duration: ${totalDuration.toFixed(2)}s | Total bytes sent: ${totalBytesSent} | Total packets: ${totalPacketsSent}`);

rtpSentStats.set(channelId, { packets: 0, bytes: 0, startTime: null }); // Reset stats

};

streamLoop();

// Stop the stream

const stop = async () => {

isStreaming = false;

stopRequested = true;

muLawBuffer = Buffer.alloc(0);

offset = 0;

logger.info(`RTP stream stopped for channel ${channelId}`);

};

return {

stop,

write: processFallback, // Method to write PCM data

muLawBuffer,

offset

};

}

// Start WebSocket connection to OpenAI real-time API

function startOpenAIWebSocket(channelId) {

logger.info(`Attempting to start OpenAI WebSocket for channel ${channelId}`);

const ws = new WebSocket(REALTIME_URL, {

headers: { 'Authorization': `Bearer ${OPENAI_API_KEY}`, 'OpenAI-Beta': 'realtime=v1' } // Authentication headers

});

let responseTimestamp = null; // Timestamp of response start

let responseTranscript = ''; // Accumulated transcript

let audioDeltaCount = 0; // Count of audio fragments

let transcriptDeltaCount = 0; // Count of transcript fragments

let audioReceivedLogged = false; // Flag for first audio log

let audioSentTime = null; // Time audio was sent to OpenAI

let callStartTime = null; // Call start time

let maxCallTimeoutId = null; // Timeout ID for max call duration

let totalPacketsSentThisResponse = 0; // Packets sent for current response

let totalPacketsSentSession = 0; // Total packets sent in session

let playbackComplete = false; // Playback completion flag

let streamHandler = null; // RTP stream handler

let isPlayingResponse = false; // Flag for active response playback

// Initialize RTP stream handler

const initializeStreamHandler = async () => {

const channelData = sipMap.get(channelId);

if (channelData && channelData.rtpSource) {

streamHandler = await streamAudio(channelId, channelData.rtpSource);

logger.info(`StreamHandler initialized for channel ${channelId} | Ready: ${streamHandler !== null}`);

} else {

logger.error(`Cannot initialize StreamHandler: No RTP source for channel ${channelId}`);

}

return streamHandler;

};

// WebSocket open event

ws.on('open', async () => {

callStartTime = Date.now();

logClient(`OpenAI WebSocket connection established for channel ${channelId}`);

await initializeStreamHandler();

ws.send(JSON.stringify({

type: 'session.update',

session: {

modalities: ['audio', 'text'], // Enable audio and text responses

voice: 'alloy', // Voice for OpenAI responses

instructions: 'Always respond with audio to any detected speech.',

turn_detection: {

type: 'server_vad', // Server-side Voice Activity Detection

threshold: VAD_THRESHOLD,

prefix_padding_ms: VAD_PREFIX_PADDING_MS,

silence_duration_ms: VAD_SILENCE_DURATION_MS

input_audio_transcription: { model: 'whisper-1' } // Transcription model

}

}));

logClient(`Session updated with VAD settings for channel ${channelId} | Threshold: ${VAD_THRESHOLD}, Prefix: ${VAD_PREFIX_PADDING_MS}ms, Silence: ${VAD_SILENCE_DURATION_MS}ms`);

// Set max call duration timeout

maxCallTimeoutId = setTimeout(async () => {

logClient(`Max call duration (${MAX_CALL_DURATION}ms) reached for channel ${channelId}, closing connection and hanging up`);

ws.close();

const channelData = sipMap.get(channelId);

if (channelData && channelData.bridge) {

try {

await ariClient.channels.hangup({ channelId: channelId });

logger.info(`Channel ${channelId} hung up due to max call duration`);

} catch (err) {

logger.error(`Failed to hang up channel ${channelId}: ${err.message}`);

}

}, MAX_CALL_DURATION);

});

// Handle incoming WebSocket messages from OpenAI

ws.on('message', async (data) => {

const response = JSON.parse(data.toString());

receivedEventCounter++;

const duration = audioSentTime ? ((Date.now() - audioSentTime) / 1000).toFixed(2) : 'N/A';

if (receivedEventCounter === 0) {

logServer(`First event received for channel ${channelId} | Type: ${response.type} | Duration: ${duration}s | Status: Received`);

}

switch (response.type) {

case 'session.created':

logServer(`Session created for channel ${channelId} | Duration: ${duration}s | Status: Received`);

break;

case 'input_audio_buffer.speech_started':

logServer(`Speech started detected for channel ${channelId} | Duration: ${duration}s | Status: Received`);

break;

case 'input_audio_buffer.speech_stopped':

logServer(`Speech stopped detected for channel ${channelId} | Duration: ${duration}s | Status: Received`);

audioSentTime = Date.now();

if (streamHandler) {

await streamHandler.stop();

logger.info(`Stopped RTP stream due to user speech for channel ${channelId}`);

streamHandler = null;

await initializeStreamHandler();

isPlayingResponse = false;

}

break;

case 'conversation.item.input_audio_transcription.completed':

logServer(`Sent audio transcribed for channel ${channelId} | Transcript: "${response.transcript.trim()}" | Duration: ${duration}s | Status: Received`);

break;

case 'response.audio.delta':

audioDeltaCount++;

if (!audioReceivedLogged) {

responseTimestamp = Date.now();

logServer(`Audio reception started for channel ${channelId} | Duration: ${duration}s | Status: Received`);

audioReceivedLogged = true;

}

isPlayingResponse = true;

const pcmChunk = Buffer.from(response.delta, 'base64'); // Decode audio chunk

logServer(`Audio delta received for channel ${channelId} | Size: ${(pcmChunk.length / 1024).toFixed(2)} KB`);

if (streamHandler) {

streamHandler.write(pcmChunk); // Send to Asterisk

totalPacketsSentThisResponse += pcmChunk.length / 160;

totalPacketsSentSession += pcmChunk.length / 160;

} else {

logger.error(`Failed to write audio delta: No StreamHandler for channel ${channelId}`);

}

break;

case 'response.audio_transcript.delta':

transcriptDeltaCount++;

responseTranscript += response.delta; // Accumulate transcript

break;

case 'response.audio_transcript.done':

logServer(`Response received for channel ${channelId} | Transcript: "${response.transcript.trim()}" | Duration: ${duration}s | Status: Received`);

responseTranscript = '';

break;

case 'response.done':

audioReceivedLogged = false;

isPlayingResponse = false;

const stats = rtpSentStats.get(channelId) || { packets: 0, bytes: 0, startTime: responseTimestamp };

const responseDuration = responseTimestamp ? ((Date.now() - responseTimestamp) / 1000).toFixed(2) : 'N/A';

logServer(`Response completed for channel ${channelId} | Duration: ${responseDuration}s | Audio Fragments: ${audioDeltaCount} | Text Fragments: ${transcriptDeltaCount} | RTP Packets: ${stats.packets} | RTP Bytes: ${stats.bytes}`);

audioDeltaCount = 0;

transcriptDeltaCount = 0;

totalPacketsSentThisResponse = 0;

if (ws.readyState === WebSocket.OPEN) {

ws.send(JSON.stringify({ type: 'input_audio_buffer.clear' })); // Clear OpenAI buffer

logClient(`Cleared OpenAI audio buffer for channel ${channelId}`);

}

break;

case 'error':

logServer(`Error received for channel ${channelId} | Message: ${response.error.message} | Code: ${response.error.code || 'N/A'} | Status: Error`);

if (streamHandler) {

await streamHandler.stop();

streamHandler = null;

}

break;

}

});

ws.on('error', (error) => {

logClient(`OpenAI WebSocket error for channel ${channelId} | Message: ${error.message} | Status: Error`);

if (streamHandler) {

streamHandler.stop();

streamHandler = null;

}

});

ws.on('close', () => {

if (maxCallTimeoutId) clearTimeout(maxCallTimeoutId);

if (streamHandler) {

streamHandler.stop();

streamHandler = null;

}

logClient(`OpenAI WebSocket connection closed for channel ${channelId} | Status: Finished`);

});

return { ws, getPlaybackComplete: () => playbackComplete, stopStream: () => streamHandler && streamHandler.stop() };

}

// Main async function to initialize ARI and handle events

(async () => {

try {

ariClient = await ari.connect(ARI_URL, ARI_USER, ARI_PASS); // Connect to ARI

logger.info(`Connected to ARI at ${ARI_URL}`);

await ariClient.start(ARI_APP); // Start Stasis app

logger.info(`ARI application "${ARI_APP}" started`);

startRTPReceiver(); // Start RTP receiver

// Handle new channel entering Stasis

ariClient.on('StasisStart', async (evt, channel) => {

logger.info(`StasisStart event received for channel ${channel.id}, name: ${channel.name}`);

if (channel.name && channel.name.startsWith('UnicastRTP')) { // ExternalMedia channel

logger.info(`ExternalMedia channel started: ${channel.id}`);

let mapping = extMap.get(channel.id);

if (!mapping) {

await new Promise(r => setTimeout(r, 500)); // Wait for mapping

mapping = extMap.get(channel.id);

}

if (mapping) {

await addExtToBridge(ariClient, channel, mapping.bridgeId);

const channelData = sipMap.get(mapping.channelId);

if (channelData && !channelData.rtpSource) {

rtpReceiver.once('message', (msg, rinfo) => {

channelData.rtpSource = rinfo; // Assign RTP source

logger.info(`RTP Source assigned for channel ${mapping.channelId}: ${rinfo.address}:${rinfo.port}`);

});

}

return;

}

logger.info(`SIP channel started: ${channel.id}`);

try {

const bridge = await ariClient.bridges.create({ type: 'mixing,proxy_media' }); // Create mixing bridge

await bridge.addChannel({ channel: channel.id });

await channel.answer(); // Answer the call

logger.info(`Channel ${channel.id} answered`);

// Set up ExternalMedia channel

const extParams = {

app: ARI_APP,

external_host: `127.0.0.1:${RTP_PORT}`,

format: 'ulaw',

transport: 'udp',

encapsulation: 'rtp',

connection_type: 'client',

direction: 'both'

};

const extChannel = await ariClient.channels.externalMedia(extParams);

extMap.set(extChannel.id, { bridgeId: bridge.id, channelId: channel.id });

logger.info(`ExternalMedia channel ${extChannel.id} created and mapped to bridge ${bridge.id}`);

const { ws, getPlaybackComplete, stopStream } = startOpenAIWebSocket(channel.id);

sipMap.set(channel.id, { bridge, ws, channelId: channel.id, sendTimeout: null, getPlaybackComplete, stopStream });

} catch (e) {

logger.error(`Error in SIP channel ${channel.id}: ${e.message}`);

}

});

// Handle channel leaving Stasis (call end)

ariClient.on('StasisEnd', async (evt, channel) => {

if (channel.name && channel.name.startsWith('UnicastRTP')) {

extMap.delete(channel.id);

logger.info(`ExternalMedia channel ${channel.id} removed from map`);

} else {

const channelData = sipMap.get(channel.id);

if (channelData) {

try {

sipMap.delete(channel.id);

logger.info(`Channel ${channel.id} removed from sipMap at start of StasisEnd`);

if (channelData.sendTimeout) {

clearInterval(channelData.sendTimeout);

channelData.sendTimeout = null;

logger.info(`Send timeout cleared for channel ${channel.id}`);

}

if (channelData.stopStream) {

await channelData.stopStream();

logger.info(`StreamHandler stopped for channel ${channel.id} in StasisEnd`);

}

if (!channelData.getPlaybackComplete()) {

logger.info(`Channel ${channel.id} hung up, checking playback status before cleanup`);

await new Promise(resolve => setTimeout(resolve, 100)); // Brief delay

}

if (channelData.ws && channelData.ws.readyState === WebSocket.OPEN) {

channelData.ws.close();

logger.info(`WebSocket closed for channel ${channel.id} in StasisEnd`);

}

await channelData.bridge.destroy();

logger.info(`Bridge ${channelData.bridge.id} destroyed`);

} catch (e) {

logger.error(`Error during cleanup for channel ${channel.id}: ${e.message}`);

}

logger.info(`Channel ended: ${channel.id}`);

// Save audio files if enabled

if (ENABLE_SENT_TO_OPENAI_RECORDING && audioFromAsteriskMap.has(channel.id) && audioFromAsteriskMap.get(channel.id).length > 0) {

saveRawFile(audioFromAsteriskMap.get(channel.id), `asterisk_input_mulaw_raw_${channel.id}.raw`);

audioFromAsteriskMap.delete(channel.id);

}

if (ENABLE_SENT_TO_OPENAI_RECORDING && audioToOpenAIMap.has(channel.id) && audioToOpenAIMap.get(channel.id).length > 0) {

saveWavFile(audioToOpenAIMap.get(channel.id), `sent_to_openai_${channel.id}.wav`, 24000);

audioToOpenAIMap.delete(channel.id);

}

});

ariClient.on('error', (err) => logger.error(`ARI client error: ${err.message}`));

ariClient.on('close', () => logger.info('ARI WebSocket connection closed'));

} catch (err) {

logger.error(`ARI connection error: ${err.message}`);

process.exit(1); // Exit on connection failure

}

})();

// Handle uncaught exceptions

process.on('uncaughtException', (err) => {

logger.error(`Uncaught Exception: ${err.message}`);

cleanup();

process.exit(1);

});

// Handle SIGINT (Ctrl+C)

process.on('SIGINT', () => {

logger.info('Received SIGINT, cleaning up...');

cleanup();

process.exit(0);

});

// Cleanup function to close sockets and connections

function cleanup() {

sipMap.forEach((data, channelId) => {

if (data.ws) data.ws.close(); // Close WebSocket

if (data.sendTimeout) clearInterval(data.sendTimeout); // Clear send interval

if (data.stopStream) data.stopStream(); // Stop RTP stream

});

rtpSender.close(); // Close RTP sender socket

rtpReceiver.close(); // Close RTP receiver socket

}

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