This page focuses on where Generic SIP Trunk fits in a production voice stack. For full setup steps, credentials, and API details, use the documentation link above.
Overview
Session Initiation Protocol (SIP) is the industry standard protocol for voice over IP communications. With Bolna's generic SIP integration, you can connect any SIP-compatible trunk or PBX to power your voice agents giving you complete control over your telephony infrastructure and carrier relationships.
This integration works with any Internet Telephony Service Provider (ITSP), on-premises PBX, or SIP-enabled infrastructure that follows RFC 3261 standards. Perfect for enterprises with existing telephony investments or specific carrier requirements.
Features & Use Cases
Total Freedom (Carrier Independence)
Work with any SIP phone provider you want. This means you are never locked into one company and can always shop around for the best rates and terms.
Bring Your Own Provider
Already have a telecom provider you love with great rates? Perfect! You can easily plug them directly into Bolna's AI agents.
Connect to Your Existing Phone System
Easily link Bolna's AI to your current office phone system (like Cisco, Avaya, or Asterisk) to upgrade your standard phone menus with smart conversational AI.
Local Presence Anywhere
Use different local phone providers in different countries to ensure clear, fast calls and a true local feel for your global customers.
Keep Call Costs Low
Pick and choose the most affordable phone carriers for your massive call volumes while still using all of Bolna's advanced AI features.
Use Case: Upgrading Office Phone Systems
Great for big companies that want to make their existing phone systems smarter by replacing old menu systems with actual AI conversations.
Use Case: Guaranteed Reliability
Set up multiple phone providers at the same time so if one carrier goes down, your calls automatically switch to another without dropping.
Use Case: Staying Compliant Locally
Perfect for connecting with specific, strictly regulated local phone companies to make sure you follow all regional telecom and data privacy rules correctly.
Technical Details
Supported Codecs
- G.711 (PCMU): 64 kbps, standard quality, minimal latency
- G.711 (PCMA): 64 kbps, European variant
- G.729: 8 kbps, lower bandwidth (if supported by trunk)
Authentication Methods
- Digest Authentication: Username/password-based (RFC 2617)
- IP Authentication: Whitelist-based, no credentials needed
Transport Protocols
- UDP: Lightweight, standard for SIP (port 5060)
- TCP: Reliable, for stable connections (port 5060)
- TLS: Encrypted, secure SIP (port 5061)
Carrier-Specific Guides
For streamlined setup with specific carriers, see our dedicated guides: