n8n SIP Client Node

A powerful n8n community node that enables SIP (Session Initiation Protocol) functionality within your n8n workflows. Make and receive calls directly from your automation workflows with real WebRTC audio streaming.

🚀 Features

Core SIP Functionality

• Make Outgoing Calls: Initiate calls to any SIP URI or phone number
• Receive Incoming Calls: Handle incoming calls with automatic detection
• Call Management: Hang up calls, get call status, and monitor call states
• Real-time Audio: WebRTC audio streaming for high-quality voice communication

SIP Server Support

• Asterisk: Full compatibility with Asterisk PBX systems
• FreeSWITCH: Native support for FreeSWITCH servers
• 3CX: Works seamlessly with 3CX phone systems
• Custom SIP Servers: Connect to any standard SIP server

WebRTC Integration

• ICE Servers: Configurable STUN/TURN servers for NAT traversal
• Audio Codecs: Support for Opus, PCMU, PCMA, and G722 codecs
• Real-time Streaming: Low-latency audio communication
• Cross-platform: Works on any platform that supports WebRTC

n8n Integration

• Workflow Automation: Integrate calling into your automation workflows
• Error Handling: Robust error handling with detailed error messages
• Status Monitoring: Real-time call status and monitoring
• Event-driven: Trigger workflows based on call events

📦 Installation

npm Installation (Recommended)

# Install the n8n SIP client node
npm install @paulstinchcombe/n8n-nodes-sip-client

# Or using yarn
yarn add @paulstinchcombe/n8n-nodes-sip-client

# Or using pnpm
pnpm add @paulstinchcombe/n8n-nodes-sip-client

🔧 Configuration

SIP Server Setup

The node supports various SIP servers. Here are example configurations:

Asterisk Configuration

{
  "sipServer": "asterisk.example.com",
  "sipPort": 5060,
  "username": "your_username",
  "password": "your_password",
  "displayName": "n8n SIP Client"
}

FreeSWITCH Configuration

{
  "sipServer": "freeswitch.example.com",
  "sipPort": 5060,
  "username": "your_username",
  "password": "your_password",
  "displayName": "n8n SIP Client"
}

3CX Configuration

{
  "sipServer": "3cx.example.com",
  "sipPort": 5060,
  "username": "your_extension",
  "password": "your_password",
  "displayName": "n8n SIP Client"
}

WebRTC Configuration

Configure WebRTC settings for optimal performance:

{
  "webrtcConfig": {
    "iceServers": "stun:stun.l.google.com:19302,stun:stun1.l.google.com:19302",
    "audioCodec": "Opus"
  }
}

📋 Usage

Making Calls

1. Add the SIP Client node to your workflow
2. Select "Make Call" operation
3. Configure SIP settings:
   • SIP Server: Your SIP server address
   • Username: Your SIP username
   • Password: Your SIP password
   • Destination: The number or SIP URI to call
4. Configure call parameters:
   • Duration: Maximum call duration (0 for unlimited)
5. Run the workflow

Receiving Calls

1. Add the SIP Client node to your workflow
2. Select "Receive Call" operation
3. Configure SIP settings (same as above)
4. The node will wait for incoming calls
5. Handle the call data in subsequent nodes

Call Management

• Hang Up: End the current active call
• Get Status: Retrieve current call status and information

🔗 Workflow Examples

Example 1: Automated Customer Support

graph LR
    A[Webhook Trigger] --> B[SIP Client - Make Call]
    B --> C[Wait for Answer]
    C --> D[Play IVR Message]
    D --> E[SIP Client - Hang Up]

Example 2: Call Monitoring

graph LR
    A[SIP Client - Receive Call] --> B[Log Call Details]
    B --> C[Send Notification]
    C --> D[SIP Client - Answer Call]

Example 3: Integration with CRM

graph LR
    A[CRM Trigger] --> B[Get Customer Info]
    B --> C[SIP Client - Make Call]
    C --> D[Update CRM with Call Result]

🛠️ Operations

Make Call

• Purpose: Initiate an outgoing call to a destination
• Parameters: SIP configuration, destination, duration
• Returns: Call ID, status, timestamp

Receive Call

• Purpose: Wait for and handle incoming calls
• Parameters: SIP configuration
• Returns: Call ID, caller information, status

Hang Up

• Purpose: End the current active call
• Parameters: None (uses current call)
• Returns: Status confirmation

Get Status

• Purpose: Get current call status and information
• Parameters: None
• Returns: Current status, active call details

🔧 Parameters

SIP Configuration

• SIP Server: Your SIP server address (required)
• SIP Port: Server port (default: 5060)
• Username: SIP username for authentication (required)
• Password: SIP password for authentication (required)
• Display Name: Optional display name for the client

WebRTC Configuration

• ICE Servers: Comma-separated list of STUN/TURN servers
• Audio Codec: Preferred audio codec (Opus, PCMU, PCMA, G722)

Call Parameters

• Destination: SIP URI or phone number to call
• Duration: Maximum call duration in seconds (0 for unlimited)

🚨 Error Handling

The node provides comprehensive error handling:

• Connection Errors: Clear messages for server connectivity issues
• Authentication Errors: Specific messages for credential problems
• Call Errors: Detailed information about call failures
• Configuration Errors: Validation of required parameters

All errors include:

• Item index for debugging
• Detailed error description
• Suggested solutions
• Operation context

📞 Supported SIP Servers

Asterisk

• ✅ Full compatibility
• ✅ WebRTC support
• ✅ Audio codec support
• ✅ Call management

FreeSWITCH

• ✅ Native support
• ✅ WebRTC integration
• ✅ Multi-codec support
• ✅ Advanced call features

3CX

• ✅ Seamless integration
• ✅ Extension support
• ✅ Call routing
• ✅ Status monitoring

Custom SIP Servers

• ✅ Standard SIP protocol
• ✅ WebRTC compatibility
• ✅ Configurable settings
• ✅ Flexible deployment

🔒 Security

• Password Protection: Secure credential handling
• WebRTC Security: Encrypted audio streams
• SIP Security: Standard SIP security protocols
• No Data Storage: No sensitive data is stored

📊 Performance

• Low Latency: Real-time audio communication
• High Quality: WebRTC audio streaming
• Scalable: Support for multiple concurrent calls
• Efficient: Optimized resource usage

🤝 Support

Getting Help

• Documentation: Comprehensive guides and examples
• Issues: Report bugs on GitHub
• Community: Join n8n community discussions
• Examples: See workflow examples in documentation

Troubleshooting

• Connection Issues: Check SIP server configuration
• Authentication: Verify username and password
• Audio Problems: Check WebRTC configuration
• Call Failures: Review error messages for guidance

📄 License

This project is licensed under the MIT License - see the LICENSE file for details.

🙏 Acknowledgments

• n8n Team: For the excellent n8n platform
• SIP.js: For the SIP protocol implementation
• WebRTC: For real-time communication capabilities
• Community: For feedback and contributions

Ready to integrate SIP calling into your n8n workflows? 🚀

Install the node and start building powerful voice-enabled automations today!