Which integrations are available for this server?

Provides VoIP calling capabilities through Asterisk PBX systems, enabling AI agents to make phone calls and handle voice conversations Enables VoIP calling through Cisco Unified Communications Manager (CUCM), allowing AI agents to make phone calls via enterprise Cisco phone systems Integrates with OpenAI's Real-Time Voice API and o3-mini model to power AI-driven phone conversations and generate sophisticated call instructions from simple briefs

How do I use CallCenter.js MCP?

1. Click on "Install Server". 2. Wait a few minutes for the server to deploy. Once ready, it will show a "Started" state. 3. In the chat, type @ followed by the MCP server name and your instructions, e.g., "@CallCenter.js MCP call my dentist to reschedule my 3pm appointment to next Tuesday" That's it! The server will respond to your query, and you can continue using it as needed. Here is a step-by-step guide with screenshots.

Ask DeepWiki

CallCenter.js MCP + CLI

An MCP Server, CLI tool, and API that makes phone calls on your behalf using VoIP.

Just tell Claude what you want to accomplish, and it will call and handle the conversation for you. This is essentially an MCP Server that bridges between OpenAI's Real-Time Voice API and your VoIP connection to call people on your behalf.

⚠️ Vibe-coded side project! Please do not use this in any kind of professional context. This is a side project coded in a weekend. There are no guard rails. Your MCP client can call any number with this, even if you don't ask it to. In fact, it has done so during testing - it called a random number during the night "for testing" and played back scary low-pitched noises - then claimed it called MY number. So YMMV, no warranties. See disclaimer below.

📞 Example: Order Pizza with Claude

You: "Can you call Tony's Pizza and order a large pepperoni pizza for delivery to 123 Main St? My name is John and my number is 555-0123."

Claude automatically calls the restaurant:

⏺ mcp__callcenter_js__simple_call(phone_number: "+15551234567", brief: "Call Tony's Pizza and order a large pepperoni pizza for delivery to 123 Main St. Customer name is John, phone number 555-0123", caller_name: "John") ⎿ # Simple Call Result **Status:** ✅ Success **Duration:** 3 minutes 24 seconds **Call ID:** abc123xyz ## Call Transcript [14:23:15] 🎤 HUMAN: Tony's Pizza, how can I help you? [14:23:15] 🤖 ASSISTANT: Hi! I'm calling on behalf of John to place a delivery order. [14:23:20] 🎤 HUMAN: Sure! What would you like? [14:23:20] 🤖 ASSISTANT: I'd like to order one large pepperoni pizza for delivery please. [14:23:25] 🎤 HUMAN: Large pepperoni, got it. What's the delivery address? [14:23:25] 🤖 ASSISTANT: The address is 123 Main Street. [14:23:30] 🎤 HUMAN: And your phone number? [14:23:30] 🤖 ASSISTANT: The phone number is 555-0123. [14:23:35] 🎤 HUMAN: Perfect! That'll be $18.99. We'll have it delivered in about 30 minutes. [14:23:40] 🤖 ASSISTANT: That sounds great! Thank you so much. [14:23:42] 🎤 HUMAN: You're welcome! Have a great day.

Pizza ordered successfully! 🍕

Related MCP server: Audio MCP Server

📚 Quick Context for the Uninitiated

VoIP (Voice over IP) is how you make phone calls over the internet instead of traditional phone lines. SIP (Session Initiation Protocol) is the language these systems speak to connect calls. Think of it as HTTP but for phone calls.

Fritz!Box is a popular German router/modem that happens to have a built-in phone system (PBX). If you have one, you already have everything you need to make VoIP calls - this tool just connects to it. Outside Germany, you might know similar devices from other brands, or use dedicated VoIP services like Asterisk, 3CX, or cloud providers.

MCP (Model Context Protocol) is Anthropic's standard for connecting AI assistants like Claude to external tools and services. It's what lets MCP clients actually do things instead of just talking about them.

🚀 What This Enables

🔌 MCP Server - Use directly in Claude Code or any MCP client (most popular usage)
🖥️ CLI Tool - Command-line interface for direct phone calls
📚 TypeScript API - Programmatic library for building voice applications

Built as a bridge between OpenAI's Real-Time Voice API and VoIP networks, with multiple codec support (G.722, G.711), and expanded SIP protocol support for broad VoIP compatibility. Compatible with the latest gpt-realtime model released August 28, 2025.

🏗️ System Architecture

graph TB subgraph "User Interface" A[Claude Code/MCP Client] B[CLI Tool] C[TypeScript API] end subgraph "CallCenter.js Core" D[MCP Server] E[VoiceAgent] F[Call Brief Processor o3-mini model] end subgraph "Communication Layer" G[SIP Client Provider Support] H[Audio Bridge RTP Streaming] end subgraph "Audio Processing" I[G.722 Codec 16kHz Wideband] J[G.711 Codec 8kHz Fallback] end subgraph "External Services" K[OpenAI Real-Time API gpt-realtime model] L[VoIP Network Fritz!Box/Asterisk/etc] end A --> D B --> E C --> E D --> E E --> F E --> G E --> H F --> K G --> L H --> I H --> J H --> K style F fill:#e1f5fe style K fill:#fff3e0 style L fill:#f3e5f5

⚠️ Vibe-coded project! Developed and tested on Fritz!Box (a German router with built-in VoIP) only. Other provider configs are research-based but untested. YMMV, no warranties. See disclaimer below.

100% vibe-coded

🔌 MCP Client Integration (Most Popular!)

Perfect for when your coding agent needs to call library authors to complain about their documentation! 😄

Quick Setup

# Add to Claude Code with one command: claude mcp add --env SIP_USERNAME=your_actual_extension \ --env SIP_PASSWORD="your_actual_password" \ --env SIP_SERVER_IP=192.168.1.1 \ --env OPENAI_API_KEY="sk-your_actual_openai_key" \ --env USER_NAME="Your Actual Name" \ -- callcenter.js npx -- github:gerkensm/callcenter.js-mcp --mcp

Then just ask your MCP Client to make calls:

"Can you call the pizza place and order a large pepperoni? My number is 555-0123."

Your MCP Client will automatically handle the entire conversation using the AI Voice Agent! 🤖📞

✨ Key Features

🎙️ Multiple Codec Support: G.722 wideband (16kHz) + G.711 fallback for broad compatibility
🧩 Compiler-Free WASM Codec: Ships with a prebuilt G.722 WebAssembly module so npx users get wideband audio without installing build tools (native addon still available for max performance)
🤖 AI-Powered Conversations: Uses OpenAI's Real-Time Voice API with the latest gpt-realtime model (released August 28, 2025) for actual calls, with o3-mini model for instruction generation
🌍 Automatic Language Detection: Intelligently detects conversation language from call briefs and configures transcription accordingly
🎭 Auto Voice Selection: New 'auto' mode where o3-mini selects optimal voice based on call context (formality, industry, goals)
🔊 Voice Characteristics: Full support for all 10 OpenAI Realtime API voices with gender and personality awareness
🌐 Expanded SIP Support: Configurations for common SIP providers (Fritz!Box tested, others experimental)
🔧 Smart Configuration: Auto-detects provider requirements and optimizes settings
📞 Enterprise-Ready: Supports advanced SIP features (STUN/TURN, session timers, transport fallback)
🔄 Robust Connection Management: Automatic reconnection with intelligent error handling
✅ Built-in Validation: Comprehensive configuration validation with network testing
🎯 Provider Profiles: Pre-configured settings for popular SIP systems
🔌 MCP Server: Integrate with MCP clients like Claude Code
📚 TypeScript API: Programmatic library for building voice applications
📝 Call Brief Processing: Natural language call instructions using o3-mini model with structured JSON output
🎵 Optional Call Recording: Stereo WAV recording with caller/AI separation

🚀 Quick Start

Option 1: Run Instantly with npx (No Installation) ⚡

Fastest way to try it out:

# Set environment variables (or create .env file) export SIP_USERNAME="your_extension" export SIP_PASSWORD="your_password" export SIP_SERVER_IP="192.168.1.1" export OPENAI_API_KEY="sk-your-key-here" # Run directly from GitHub (no installation needed!) npx github:gerkensm/callcenter.js-mcp call "+1234567890" --brief "Call restaurant for reservation" --user-name "Your Name"

Or using a .env file:

# Create .env file cat > .env << EOF SIP_USERNAME=your_extension SIP_PASSWORD=your_password SIP_SERVER_IP=192.168.1.1 OPENAI_API_KEY=sk-your-key-here SIP_PROVIDER=fritz-box OPENAI_VOICE=auto EOF # Run from GitHub (loads .env automatically) npx github:gerkensm/callcenter.js-mcp call "+1234567890" --brief "Call restaurant"

Note: High-quality G.722 audio ships as a prebuilt WebAssembly module, so npx works even on machines without compilers. If you want the faster native addon instead, run npm run build:native (or npm run build:all) after cloning.

Option 2: Local Installation

Prerequisites

Node.js 20+
(Optional) Python 3.x + C/C++ build tools — only needed if you plan to rebuild the native addon instead of using the bundled WebAssembly codec
- macOS: Xcode Command Line Tools (xcode-select --install)
- Windows: Visual Studio Build Tools
- Linux: build-essential package
OpenAI API key

Note: The prebuilt WebAssembly codec already provides wideband G.722 audio out of the box. Rebuilding the native addon is optional and mainly useful for squeezing out a little more performance.

Installation

# Clone and install git clone https://github.com/gerkensm/callcenter.js-mcp cd callcenter.js-mcp npm install # Copy example configuration cp config.example.json config.json

Configuration

Edit config.json with your settings:

{ "sip": { "username": "your_sip_username", "password": "your_sip_password", "serverIp": "192.168.1.1", "serverPort": 5060, "provider": "fritz-box" }, "ai": { "openaiApiKey": "sk-your-openai-api-key-here", "voice": "alloy", "instructions": "You are a helpful AI assistant making phone calls on behalf of users.", "userName": "Your Name" } }

🎯 Usage Options

1. MCP Server (Claude Code Integration) ⭐

Most popular usage - integrates with Claude Code for seamless AI-powered calling. Perfect for when your coding agent needs to call library authors to complain about their documentation! 😄

Quick Setup with npx (Recommended)

Option 1: Using MCP Client CLI (Easiest)

# Replace with your ACTUAL credentials before running: claude mcp add --env SIP_USERNAME=your_actual_extension \ --env SIP_PASSWORD="your_actual_password" \ --env SIP_SERVER_IP=192.168.1.1 \ --env OPENAI_API_KEY="sk-your_actual_openai_key" \ --env USER_NAME="Your Actual Name" \ -- callcenter.js npx -- github:gerkensm/callcenter.js-mcp --mcp

⚠️ Important: Replace the placeholder values with your actual SIP credentials and OpenAI API key, or the server will fail to connect.

Option 2: Manual Configuration

Configure in Claude Code's MCP settings to automatically pull from GitHub:

{ "mcpServers": { "callcenter.js": { "command": "npx", "args": ["github:gerkensm/callcenter.js-mcp", "--mcp"], "env": { "SIP_USERNAME": "your_extension", "SIP_PASSWORD": "your_password", "SIP_SERVER_IP": "192.168.1.1", "OPENAI_API_KEY": "sk-your-key-here", "USER_NAME": "Your Name" } } } }

Alternative: Local Installation

For local development or if you prefer local installation:

npm start --mcp

Or configure Claude Code with local installation:

{ "mcpServers": { "callcenter.js": { "command": "node", "args": ["dist/cli.js", "--mcp"], "cwd": "/path/to/voip-agent" } } }

Available MCP tools:

simple_call - Make calls with automatic instruction generation
advanced_call - Make calls with granular parameter control

Example usage in MCP Client:

You: "Can you call Bocca di Bacco restaurant and book a table for 2 people tonight at 7:30pm? My name is John Doe." MCP Client: I'll call Bocca di Bacco restaurant to book a table for 2 people tonight at 7:30pm. 🔧 mcp__ai-voice-agent__simple_call( phone_number: "+1234567890", brief: "Call Bocca di Bacco restaurant and book a table for 2 people tonight at 7:30pm", caller_name: "John Doe" ) ✅ Call completed successfully! 📞 Duration: 2 minutes 15 seconds 📝 Reservation confirmed for 2 people at 7:30pm tonight

More examples:

You: "My internet is down. Can you call my ISP and get a status update? I'm Sarah Johnson, account #12345." MCP Client: I'll call your internet service provider to check on the outage status. 🔧 mcp__ai-voice-agent__simple_call( phone_number: "+18005551234", brief: "Call ISP about internet outage, customer Sarah Johnson account #12345", caller_name: "Sarah Johnson" )

You: "Call Dr. Smith's office to reschedule my 3pm appointment to next week. I'm Mike Chen." MCP Client: I'll call Dr. Smith's office to reschedule your appointment. 🔧 mcp__ai-voice-agent__simple_call( phone_number: "+15551234567", brief: "Call Dr. Smith's office to reschedule Mike Chen's 3pm appointment to next week", caller_name: "Mike Chen" )

The MCP Client automatically handles the entire conversation using the AI Voice Agent!

2. Command Line Interface

Perfect for when you need to 😄

💡 Use

The --brief option uses OpenAI's o3-mini model to generate sophisticated instructions from your simple description, while --instructions sends your text directly to the Real-Time Voice API. Since the Real-Time Voice API is optimized for speed (not sophistication), --brief typically produces much better call outcomes.

# ✅ RECOMMENDED: Use brief for natural language goals npm start call "+1234567890" --brief "Call the restaurant and book a table for 2 at 7pm tonight" --user-name "John Doe" # ✅ RECOMMENDED: Brief works for any call type npm start call "+1234567890" --brief "Call to check appointment availability for John Doe" # ⚠️ ONLY use instructions for very specific, custom behavior npm start call "+1234567890" --instructions "You must follow this exact script: Say hello, ask for manager, then hang up" # Other examples with brief npm start call "+1234567890" --record "meeting.wav" --duration 300 --brief "Conference call to discuss project status" npm start call "+1234567890" --log-level verbose --brief "Test call to verify connectivity"

CLI Options

npm start call <number> [options] Options: -c, --config <path> Configuration file path (default: config.json) -d, --duration <seconds> Maximum call duration in seconds (default: 600) -v, --verbose Verbose mode - show all debug information -q, --quiet Quiet mode - show only transcripts, errors, and warnings --log-level <level> Set log level (quiet|error|warn|info|debug|verbose) (default: info) --no-colors Disable colored output --no-timestamp Disable timestamps in logs --record [filename] Enable stereo call recording (optional filename) --brief <text> Call brief to generate instructions from (RECOMMENDED) --instructions <text> Direct AI instructions (use only for specific custom behavior) --user-name <name> Your name for the AI to use when calling --voice <name> Voice to use (default: auto) - see Voice Selection section --help Display help information

3. Programmatic API

import { makeCall, createAgent } from 'callcenter.js'; // Simple call with brief const result = await makeCall({ number: '+1234567890', brief: 'Call Bocca di Bacco and book a table for 2 at 19:30 for Torben', userName: 'Torben', config: 'config.json' }); console.log(`Call duration: ${result.duration}s`); console.log(`Transcript: ${result.transcript}`); // Advanced usage with agent instance const agent = await createAgent('config.json'); agent.on('callEnded', () => { console.log('Call finished!'); }); await agent.makeCall({ targetNumber: '+1234567890', duration: 300 });

📚 API Reference

`makeCall(options: CallOptions): Promise<CallResult>`

Make a phone call with the AI agent.

CallOptions

interface CallOptions { number: string; // Phone number to call duration?: number; // Call duration in seconds config?: string | Config; // Configuration file path or object instructions?: string; // Direct AI instructions (highest priority) brief?: string; // Call brief to generate instructions from userName?: string; // Your name for the AI to use recording?: boolean | string; // Enable recording with optional filename logLevel?: 'quiet' | 'error' | 'warn' | 'info' | 'debug' | 'verbose'; colors?: boolean; // Enable colored output timestamps?: boolean; // Enable timestamps in logs }

CallResult

interface CallResult { callId?: string; // Call ID if successful duration: number; // Call duration in seconds transcript?: string; // Full conversation transcript success: boolean; // Whether call was successful error?: string; // Error message if failed }

`createAgent(config, options?): Promise<VoiceAgent>`

Create a VoiceAgent instance for advanced use cases.

Configuration Structure

interface Config { sip: { username: string; password: string; serverIp: string; serverPort?: number; provider?: string; stunServers?: string[]; turnServers?: TurnServer[]; }; ai: { openaiApiKey: string; voice?: 'auto' | 'alloy' | 'ash' | 'ballad' | 'cedar' | 'coral' | 'echo' | 'marin' | 'sage' | 'shimmer' | 'verse'; instructions?: string; brief?: string; userName?: string; }; logging?: { level?: string; }; }

Environment Variables

All configuration options can be set via environment variables (useful for npx usage):

Required Variables:

SIP_USERNAME=your_extension SIP_PASSWORD=your_password SIP_SERVER_IP=192.168.1.1 OPENAI_API_KEY=sk-your-key-here USER_NAME="Your Name" # Required when using --brief

Optional Variables:

# SIP Configuration SIP_SERVER_PORT=5060 SIP_LOCAL_PORT=5060 SIP_PROVIDER=fritz-box # fritz-box, asterisk, cisco, 3cx, generic STUN_SERVERS="stun:stun.l.google.com:19302,stun:stun2.l.google.com:19302" SIP_TRANSPORTS="udp,tcp" # OpenAI Configuration OPENAI_VOICE=auto # auto (recommended), marin, cedar, alloy, echo, shimmer, coral, sage, ash, ballad, verse OPENAI_INSTRUCTIONS="Your custom AI instructions" # Advanced SIP Features SESSION_TIMERS_ENABLED=true SESSION_EXPIRES=1800 SESSION_MIN_SE=90 SESSION_REFRESHER=uac

Priority order: CLI flags > Config file > Environment variables

✅ Quick Success Check

Before making real calls, validate your setup with these safe tests:

1. Configuration Validation

# Basic validation - checks syntax and required fields npm run validate config.json # Detailed validation with network connectivity tests npm run validate:detailed # Get specific fix suggestions for issues npm run validate:fix

2. Test Call to Yourself (Fritz!Box users)

# Call your own extension to verify audio quality (safe test) npm start call "**620" --brief "Test call to check audio quality" --user-name "Your Name" --duration 30 # Or use your mobile number for end-to-end test npm start call "+49123456789" --brief "Quick test call" --user-name "Your Name" --duration 15

3. What to Expect

✅ Working setup: Clear audio, proper AI responses, clean call termination
⚠️ Network issues: "Connection failed" errors → check firewall/STUN settings
⚠️ Auth problems: "401 Unauthorized" → verify SIP credentials
⚠️ Codec issues: Poor audio quality → G.722 compilation may have failed

Pro tip: Start with --duration 30 for test calls to avoid long waits if something goes wrong.

📋 Configuration Validation

The built-in validation system provides comprehensive analysis:

# Basic validation npm run validate config.json # Detailed validation with network connectivity tests npm run validate:detailed # Get specific fix suggestions for issues npm run validate:fix # Test example configurations for different providers npm run validate:fritz-box # AVM Fritz!Box npm run validate:asterisk # Asterisk PBX npm run validate:cisco # Cisco CUCM npm run validate:3cx # 3CX Phone System npm run validate:generic # Generic SIP provider

The validator will check:

✅ Configuration syntax and required fields
✅ Provider-specific requirements
✅ Network connectivity to SIP server
✅ STUN server reachability
✅ Codec availability (G.722/G.711)
✅ Provider compatibility score

🌐 SIP Provider Compatibility

✅ Actually Tested

AVM Fritz!Box - German router brand with built-in VoIP/SIP phone system ✅ WORKS (only one actually tested)

🤷 Vibe-coded Configs (Educated Guesses)

Asterisk PBX - Open source PBX (FreePBX, Elastix, etc.) 🤷 UNTESTED
Cisco CUCM - Enterprise Unified Communications 🤷 UNTESTED
3CX Phone System - Popular business PBX 🤷 UNTESTED
Generic SIP Providers - Standards-compliant SIP trunks 🤷 UNTESTED

🔧 Provider-Specific Features

The provider profiles are based on research and documentation, not actual testing:

Provider	Transport	NAT Traversal	Session Timers	PRACK	Keepalive
Fritz Box	UDP	Not needed	Optional	Disabled	Re-register
Asterisk	UDP/TCP	STUN	Supported	Optional	OPTIONS ping
Cisco CUCM	TCP preferred	STUN required	Required	Required	OPTIONS ping
3CX	TCP/UDP	STUN	Supported	Optional	Re-register

Configuration Decision Tree

flowchart TD A[Choose Your SIP Provider] --> B{Fritz!Box Router?} B -->|Yes| C[✅ Use fritz-box profile UDP transport No STUN needed] B -->|No| D{Enterprise System?} D -->|Cisco CUCM| E[⚠️ Use cisco profile TCP transport STUN required Session timers + PRACK] D -->|3CX| F[⚠️ Use 3cx profile TCP/UDP transport STUN recommended] D -->|Asterisk/FreePBX| G[⚠️ Use asterisk profile UDP/TCP transport STUN for NAT] D -->|Other| H[⚠️ Use generic profile Start with UDP Add STUN if needed] C --> I[Configure Basic Settings] E --> J[Configure Enterprise Settings] F --> J G --> J H --> J I --> K[Set SIP credentials serverIp = router IP typically 192.168.1.1] J --> L[Set SIP credentials serverIp = server IP Add STUN servers] K --> M{Network Location?} L --> M M -->|Local Network| N[✅ Basic setup complete Should work reliably] M -->|Cloud/Remote| O[❓ May need additional STUN/TURN configuration] style C fill:#c8e6c9 style E fill:#ffecb3 style F fill:#ffecb3 style G fill:#ffecb3 style H fill:#ffecb3 style N fill:#c8e6c9 style O fill:#ffe0b2

📝 Configuration Examples

The project includes ready-to-use configurations for all major providers:

config.example.json - AVM Fritz!Box (home/SMB default)
config.asterisk.example.json - Asterisk PBX with advanced features
config.cisco.example.json - Cisco CUCM enterprise setup
config.3cx.example.json - 3CX Phone System configuration
config.generic.example.json - Generic SIP provider template

🎵 Audio Quality & Codecs

Codec Priority & Negotiation

G.722 (Preferred) - 16kHz wideband, superior voice quality
G.711 μ-law (Fallback) - 8kHz narrowband, universal compatibility
G.711 A-law (Fallback) - 8kHz narrowband, European standard

G.722 Implementation

Prebuilt WebAssembly codec bundled with the package so every install gets wideband audio out of the box
Native C++ addon still available for optimal performance when you opt-in with npm run build:native
Based on reference implementations from CMU and Sippy Software
Automatic fallback to G.711 if codec loading fails for any reason
Real-time encoding/decoding with low latency

Optional Call Recording

Stereo WAV format with caller on left channel, AI on right channel
Optional filename specification
Synchronized audio streams for perfect alignment
High-quality PCM recording at native sample rates

Testing Audio Quality

# Test codec availability npm run test:codecs # Rebuild all codec artifacts (native + WASM + TS) if you changed the C sources npm run build:all # Disable G.722 entirely if you only want the G.711 fallback npm run build:no-g722

🤖 AI Call Brief Processing

Why This Matters: Real-Time Voice API Needs Better Instructions

OpenAI's Real-Time Voice API is optimized for speed, not sophistication. It's great at natural conversation but struggles with complex, goal-oriented tasks without very specific instructions. Here's the problem:

❌ What doesn't work well:

# Vague brief - Real-Time Voice API will be confused and unfocused npm start call "+1234567890" --brief "Call the restaurant and book a table"

❌ What's tedious and error-prone:

# Writing detailed instructions manually every time npm start call "+1234567890" --instructions "You are calling on behalf of John Doe to make a restaurant reservation for 2 people at Bocca di Bacco for tonight at 7pm. You should start by greeting them professionally, then clearly state your purpose. Ask about availability for 7pm, and if not available, ask for alternative times between 6-8pm. Confirm the booking details including date, time, party size, and get a confirmation number if possible. If you reach voicemail, leave a professional message with callback information..."

✅ What works brilliantly:

# Simple brief - o3 model generates sophisticated instructions npm start call "+1234567890" --brief "Call Bocca di Bacco and book a table for 2 at 7pm tonight" --user-name "John Doe"

How It Works

The system uses OpenAI's o3-mini reasoning model (their latest small reasoning model - smart but fast) to automatically generate detailed, sophisticated instructions from your simple brief. The o3-mini model:

Analyzes your brief and understands the goal
Creates conversation states and flow logic
Generates specific instructions for each phase of the call
Handles edge cases like voicemail, objections, and alternatives
Adapts language and tone based on context
Provides fallback strategies when things don't go as planned

Call Flow Sequence

sequenceDiagram participant U as User/Claude participant V as VoiceAgent participant B as Brief Processor (o3-mini) participant S as SIP Client participant A as Audio Bridge participant O as OpenAI Realtime (gpt-realtime) participant P as Phone/VoIP U->>V: makeCall({brief, number, userName}) V->>B: Process brief with o3-mini B->>B: Generate detailed instructions & conversation states B->>V: Sophisticated call instructions V->>S: Connect to SIP server S->>P: INVITE (start call) P->>S: 200 OK (call answered) S->>V: Call established V->>A: Initialize audio bridge V->>O: Connect to OpenAI Realtime O->>V: WebSocket connected V->>O: Send generated instructions loop During Call P->>A: RTP audio packets A->>A: Decode G.722/G.711 → PCM A->>O: Stream PCM audio O->>O: Process speech → text O->>O: Generate AI response O->>A: Stream AI audio (PCM) A->>A: Encode PCM → G.722/G.711 A->>P: RTP audio packets Note over V: Monitor call progress & transcript logging end alt Call completed naturally O->>V: Call completion signal V->>S: Send BYE else Duration limit reached V->>V: Safety timeout triggered V->>S: Send BYE end S->>P: BYE (end call) P->>S: 200 OK V->>U: CallResult{transcript, duration, success}

Before/After Example

Your simple input:

"Call Bocca di Bacco and book a table for 2 at 7pm tonight"

What o3-mini generates (excerpt):

## Personality and Tone Identity: I am an assistant calling on behalf of John Doe to make a restaurant reservation. Task: I am responsible for booking a table for 2 people at Bocca di Bacco today at 7:00 PM. Tone: Professional, warm, and respectful. ## Instructions 1. Open the conversation immediately: "Hello, this is an assistant calling on behalf of John Doe." 2. Read back critical data: Repeat times and details for confirmation. 3. Handle objections: Respond politely and offer alternatives between 6-8 PM. ... ## Conversation States [ { "id": "1_greeting", "description": "Greeting and introduction of call purpose", "instructions": ["Introduce yourself as an assistant", "Immediately mention the reservation request"], "examples": ["Hello, this is an assistant calling on behalf of John Doe. I'm calling to book a table for 2 people today at 7:00 PM."] } ]

Automatic Adaptations

The o3-mini brief processor automatically:

Detects language from your brief and generates instructions in that language
Creates conversation flow with logical states and transitions
Handles cultural context (German restaurants vs. American vs. Japanese)
Generates appropriate examples with real phrases (no placeholders)
Provides voicemail scripts for when nobody answers
Plans for objections and alternative solutions

When to Use Each Approach

Use for 95% of calls - it's easier and produces better results
Use only when you need very specific, custom behavior
Brief processing is perfect for: reservations, appointments, business calls, customer service
Direct instructions are better for: highly specialized scenarios, testing, or when you've already perfected your prompt

🎤 Voice Selection

The AI agent supports 10 different voices from OpenAI's Realtime API, each with unique characteristics. By default, the system uses auto mode where o3-mini intelligently selects the optimal voice based on your call's context.

Available Voices

Voice	Gender	Description	Best For
marin	Female	Clear, professional feminine voice	All-purpose: business calls, customer support, negotiations
cedar	Male	Natural masculine voice with warm undertones	All-purpose: professional calls, consultations, service interactions
alloy	Neutral	Professional voice with good adaptability	Technical discussions, business contexts, general inquiries
echo	Male	Conversational masculine voice	Casual to formal interactions, versatile tone
shimmer	Female	Warm, expressive feminine voice	Empathetic conversations, sales, professional contexts
coral	Female	Warm and friendly feminine voice	Customer interactions, consultations, support calls
sage	Neutral	Calm and thoughtful voice	Medical consultations, advisory roles, serious discussions
ash	Neutral	Clear and precise voice	Technical explanations, instructions, educational content
ballad	Female	Melodic and smooth feminine voice	Presentations, storytelling, engaging conversations
verse	Neutral	Versatile and expressive voice	Dynamic conversations, adaptable to any context

Auto Voice Selection (Recommended)

The auto mode (default) uses o3-mini to analyze your call context and select the most appropriate voice:

# Auto mode - AI selects the best voice npm start call "+1234567890" --brief "Call doctor's office to schedule appointment" --user-name "John" # Might select: sage (calm, professional for healthcare) # Auto mode adapts to context npm start call "+1234567890" --brief "Call pizza place to order delivery" --user-name "Sarah" # Might select: coral or echo (friendly, casual for food service)

Manual Voice Selection

You can override auto selection when you have specific requirements:

# Use a specific voice npm start call "+1234567890" --voice marin --brief "Call to book reservation" --user-name "Alex" # Professional contexts npm start call "+1234567890" --voice cedar --brief "Call bank about account" --user-name "Pat" # Friendly service calls npm start call "+1234567890" --voice coral --brief "Call flower shop for delivery" --user-name "Sam"

Configuration Options

Set default voice in your config file or environment:

// config.json { "ai": { "voice": "auto", // or specific voice like "marin", "cedar", etc. // ... } }

# Environment variable export OPENAI_VOICE=auto # or marin, cedar, alloy, etc.

Voice Selection Guidelines

The auto mode considers these factors:

Formality Level: High (cedar, marin, sage) → Medium (alloy, verse) → Low (echo, coral, shimmer)
Industry Context: Healthcare (sage, shimmer), Finance (cedar, sage), Retail (coral, echo), Tech (alloy, ash)
Goal Type: Authority needed (cedar, sage), Friendliness (coral, shimmer), Efficiency (marin, alloy)
Language: Voices adapt to detected language from your call brief

MCP Integration

The MCP tools strongly recommend auto mode but support manual override:

// Simple call - auto voice selection mcp__callcenter_js__simple_call({ phone_number: "+1234567890", brief: "Call restaurant for reservation", caller_name: "John", voice: "auto" // Optional, defaults to auto }) // Advanced call - manual voice selection mcp__callcenter_js__advanced_call({ phone_number: "+1234567890", goal: "Schedule medical appointment", user_name: "Jane", voice: "sage" // Override for specific voice })

🔄 Advanced Features

Smart Connection Management

Automatic Reconnection: Exponential backoff with provider-specific error handling
Transport Fallback: UDP → TCP → TLS based on what works
Provider-Aware Error Recovery: Different strategies for Fritz Box vs. Asterisk vs. Cisco
Network Change Handling: Adapts to network connectivity changes

Enhanced SIP Protocol Support

STUN/TURN Integration: NAT traversal for cloud and enterprise deployments
Session Timers (RFC 4028): Connection stability for long calls
PRACK Support (RFC 3262): Reliable provisional responses for enterprise systems
Multiple Transports: UDP, TCP, TLS with intelligent fallback

Configuration Intelligence

Provider Auto-Detection: Identifies provider from SIP domain/IP
Requirements Validation: Ensures all provider-specific needs are met
Network Testing: Real connectivity tests to SIP servers and STUN servers
Optimization Suggestions: Actionable recommendations for better performance

🛠️ Development & Testing

Build Commands

# Default build (WASM refresh + TypeScript, skips if artifacts already exist) npm run build # Build components separately (useful for maintainers) npm run build:wasm # Regenerate the G.722 WebAssembly codec npm run build:native # Rebuild the native addon (requires toolchain) npm run build:all # Run native + WASM + TypeScript in one go npm run build:ts # TypeScript compilation only # Development with hot reload npm run dev # Clean all build artifacts npm run clean

Configuration Testing

# Validate any config file npm run validate path/to/config.json # Test with different providers npm run validate -- --provider asterisk config.json # Get detailed network diagnostics npm run validate -- --detailed --network config.json # Show fix suggestions for issues npm run validate -- --fix-suggestions config.json

Project Structure

src/ ├── voice-agent.ts # Main orchestration with ConnectionManager ├── connection-manager.ts # Smart connection handling & reconnection ├── sip-client.ts # Enhanced SIP protocol with provider support ├── audio-bridge.ts # RTP streaming and codec management ├── openai-client.ts # OpenAI Real-Time Voice API integration ├── call-brief-processor.ts # o3-mini model call brief processing ├── mcp-server.ts # MCP (Model Context Protocol) server ├── validation.ts # Configuration validation engine ├── config.ts # Enhanced config loading with provider profiles ├── logger.ts # Comprehensive logging with transcript capture ├── index.ts # Main programmatic API exports ├── providers/ │ └── profiles.ts # Provider-specific configuration database ├── testing/ │ └── network-tester.ts # Real network connectivity testing ├── codecs/ # Codec abstraction layer │ ├── g722.ts # G.722 wideband implementation │ └── g711.ts # G.711 fallback codecs └── cli.ts # Command-line interface scripts/ └── validate-config.js # Comprehensive validation CLI tool config.*.example.json # Provider-specific example configurations

📊 Validation & Diagnostics

The built-in validation system provides comprehensive analysis:

Configuration Report Example

🔍 CallCenter.js Configuration Validator 📋 Provider: AVM Fritz!Box (auto-detected) 🎯 Provider Compatibility Score: 100% ✅ Configuration is valid and ready for use! 🌐 Network Connectivity: ✅ SIP Server: Reachable (12ms latency) ✅ G.722 codec: Available for high-quality audio 💡 Optimization Suggestions: 💡 G.722 wideband codec available (already enabled) 💡 Excellent latency - local network performance optimal 🚀 Next steps: npm start call "<number>"

Network Diagnostics

Real SIP Server Testing: Actual UDP/TCP connectivity tests
STUN Server Validation: Tests NAT traversal capability
Latency Measurement: Network performance assessment
Provider-Specific Recommendations: Tailored advice based on detected issues

🔧 Troubleshooting

Configuration Issues

Run validation first:
npm run validate:detailed
Check provider compatibility:
npm run validate -- --provider fritz-box config.json
Get specific fix suggestions:
npm run validate:fix

Network Connectivity

Fritz Box: Usually works with UDP on local network
Cloud/Enterprise: May need STUN servers for NAT traversal
Firewall Issues: Ensure SIP port (5060) and RTP ports are open

Audio Quality

Verify G.722 is available:
npm run test:codecs
Check codec negotiation in logs:
✅ Selected codec: PT 9 (G722/8000)
Network issues: High latency/packet loss affects audio quality

Build Problems

Native compilation fails (only happens if you explicitly ran npm run build:native or npm run build:all): stick with the bundled WASM codec unless you specifically need native performance. To drop back to G.711 entirely, run:
npm run build:no-g722
Provider-specific issues: Check validation recommendations for your provider

MCP Integration Issues

Server won't start:
# Check for port conflicts or config issues npm start --mcp
Claude Code not connecting:
- Verify MCP server configuration in Claude Code settings
- Check that the working directory path is correct
- Ensure the server is running and accessible

📈 What I Built

This is a personal project that includes:

🌐 Fritz!Box Support: Actually tested and works
🤷 Other SIP Configs: Vibe-coded based on documentation reading
🔄 Connection Handling: Seems to work, has retry logic
✅ Config Validation: Catches obvious mistakes
📊 Network Testing: Basic connectivity checks
🎯 Provider Profiles: Research-based guesses about different systems
🔌 MCP Server: Works with Claude Code (tested)
📚 TypeScript API: Clean interfaces for programmatic use
📝 Call Brief Processing: Uses o3-mini to generate instructions (works well)
🎵 Optional Call Recording: Stereo WAV files with left/right channels
📋 Transcript Capture: Real-time conversation logs

⚠️ Important Disclaimer

This project is vibe-coded! 🚀

This means:

✅ Works on Fritz!Box - that's what I actually tested
🤷 Other providers - I tried to make it more useful but can't promise anything
🤷 Advanced features - seemed like good ideas based on research, but who knows
⚠️ YMMV - your setup is probably different than mine
⚠️ No warranties - use at your own risk

What This Means for You

Fritz Box users: Should work great! ✅
Other providers: The configuration profiles are educated guesses based on research - they might work, they might not
Enterprise users: I tried to add the features that seemed important, but I have no idea if they actually work correctly
Issues & PRs: I'll accept pull requests, but I can't promise to fix issues I can't reproduce or test

If You Want to Contribute

✅ Test it on your setup and let others know what works
✅ Share working configs if you get something else working
✅ Fix stuff that's broken and submit PRs
✅ Tell me if my assumptions were wrong about how providers work

The validation tools might help debug issues, but honestly, the real test is whether you can make actual calls.

📜 License

MIT License - see LICENSE for details.

Third-Party Components

G.722 Codec: Public domain and BSD licensed implementations
SIP Protocol: Based on sipjs-udp (MIT licensed)
Dependencies: Various open source licenses (see package.json)

🤝 Contributing

Fork the repository
Create a feature branch
Add/update validation for new providers
Test with npm run validate:detailed
Submit a pull request

📞 Support

Configuration Issues: Use npm run validate:detailed for diagnostics
Provider Support: Check compatibility matrix above
Build Problems: See troubleshooting section
Feature Requests: You can open GitHub issues, but they're unlikely to get attention anytime soon. Pull requests are much preferred!

Ready to get started? Copy an example config, run npm run validate:detailed, and start making AI-powered voice calls! 🚀