💰 Paid Plan Feature: SIP Trunk integration is available on paid plans only. Free tier users can integrate Twilio SIP Trunks as an alternative.
Overview
SIP (Session Initiation Protocol) Trunks allow you to connect your existing telephony infrastructure with Vodex.ai for seamless call routing and management. This guide outlines the technical requirements and information needed for successful SIP Trunk integration.Required Information
To integrate a SIP Trunk with Vodex.ai, you’ll need to provide the following technical details:1. Connection Details
🌐 IP Address or FQDN
IP Address: Static IP address of your SIP Trunk providerFQDN: Fully Qualified Domain Name (e.g., sip.yourprovider.com)Example:
203.0.113.10
or trunk.siprovider.com
🔌 Port Configuration
SIP Port: Usually 5060 (UDP) or 5061 (TLS)RTP Port Range: For media transmissionExample:
5060
or 5061
2. Authentication (If Required)
Authentication is only required if your SIP Trunk is not configured for peer-to-peer connection.
Field | Description | Example |
---|---|---|
Username | SIP authentication username | vodex_user_123 |
Password | SIP authentication password | SecurePass123! |
Realm | Authentication realm (optional) | sip.provider.com |
3. Caller Line Identification (CLI)
CLI management allows Vodex to handle caller ID presentation and routing.
- Format: E.164 format preferred (+1234567890)
- Validation: Must be authorized by your SIP provider
- Presentation: Configure how caller ID appears to recipients
4. DID Configuration (Optional)
DID Range
Specify the range of Direct Inward Dialing numbers available:DID Prefix
Configure number formatting and routing:SIP Trunk Types
Peer-to-Peer Connection
- Authentication: Not required
- Security: IP-based authentication
- Configuration: Simpler setup
- Use Case: Direct carrier connections
Registered Connection
- Authentication: Username/Password required
- Security: Credential-based authentication
- Configuration: More secure
- Use Case: Third-party SIP providers
Technical Requirements
Network Configuration
1
Firewall Configuration
Open required ports for SIP signaling and RTP media:
- SIP Signaling: 5060 (UDP) or 5061 (TLS)
- RTP Media: 10000-20000 (UDP range)
2
NAT Considerations
Configure NAT traversal if your SIP Trunk is behind NAT:
- Enable STUN/TURN servers if needed
- Configure proper port forwarding
3
Quality of Service
Implement QoS policies for optimal call quality:
- Prioritize SIP and RTP traffic
- Ensure sufficient bandwidth allocation
Codec Support
Vodex.ai supports the following audio codecs:Codec | Bitrate | Quality | Bandwidth Usage |
---|---|---|---|
G.711 (PCMU/PCMA) | 64 kbps | High | ~87 kbps |
G.729 | 8 kbps | Good | ~31 kbps |
G.722 | 64 kbps | HD Audio | ~87 kbps |
Integration Checklist
Before submitting your SIP Trunk integration request, ensure you have:-
Connection Details
- IP Address or FQDN
- Port configuration
- Protocol specification (UDP/TCP/TLS)
-
Authentication Information (if required)
- Username and password
- Authentication realm
-
CLI Configuration
- Authorized caller ID numbers
- CLI presentation preferences
-
DID Information (if applicable)
- DID range specification
- Number formatting preferences
- Routing requirements
-
Network Preparation
- Firewall rules configured
- NAT traversal setup (if needed)
- QoS policies implemented
Common Configuration Examples
Example 1: Basic Peer-to-Peer Setup
Example 2: Registered SIP Trunk
Example 3: Enterprise Setup
Free Tier Alternative: Twilio Integration
Free tier users can integrate Twilio SIP Trunks for telephony functionality.
- Create Twilio Account: Sign up at twilio.com
- Configure SIP Trunk: Set up SIP trunk in Twilio console
- Get Credentials: Obtain SIP domain and authentication details
- Integrate with Vodex: Use Twilio SIP credentials in Vodex configuration
Support and Next Steps
Getting Help
If you need assistance with SIP Trunk integration:- Technical Support: support@vodex.ai
- Documentation: Review our Call Settings Guide
- Live Chat: Available in your Vodex dashboard
Integration Process
- Gather Requirements: Complete the checklist above
- Submit Request: Contact support with your SIP Trunk details
- Testing Phase: We’ll configure and test the connection
- Go Live: Activate your SIP Trunk integration
Pro Tip: Have your network administrator available during the integration process to assist with any firewall or network configuration changes that may be needed.