SIP Explained: How Phone Calls Reach AI Voice Agents (2026)

Connecting Phone Calls to AI Agents

When users interact with a voice AI agent via a phone call rather than a browser, the call must transition from the phone network to an AI running on a server. This process relies on SIP (Session Initiation Protocol), a plain-text language used to set up, manage, and terminate calls.

How SIP and SIP Trunks Work

To receive calls, an AI agent requires a phone number rented from a carrier or provider. Businesses typically connect to the phone network using a SIP trunk, which acts as a virtual bundle of phone lines over the internet to handle multiple concurrent calls. A standard SIP call setup follows a specific script:

• The caller sends an invite.
• The receiving side responds with "ringing" and then "OK."
• The caller confirms the connection.
• The invite and reply exchange details on supported audio formats and destination addresses.
• The call is terminated with a "bye" message from either side.

While SIP manages the call state, the actual voice audio travels separately in small packets.

Phone Audio Quality and DTMF

Phone audio is significantly limited compared to studio audio. A standard phone call restricts frequencies to between 300 and 3,400 Hz and samples audio only 8,000 times per second. This low resolution makes certain sounds, like "S" and "F," easy to confuse, meaning speech-to-text engines have less data to work with. Additionally, callers can interact using keypad presses, which transmit dual-tone multi-frequency (DTMF) signals to navigate menus or request a human operator.

Paths to the AI Agent

There are two primary pathways for a phone call to reach an AI server:

• Path 1 (WebSocket Stream): A phone provider answers the call and streams the audio bidirectionally to the AI server over a WebSocket.
• Path 2 (Direct SIP): The call travels over a SIP trunk directly to a system that natively speaks SIP. Because SIP and WebRTC are closely related, a bridge can translate between them to place the caller into a virtual room with the agent, or the call can connect directly to an API without a bridge.

Best Practices for Going Live

Before deploying a phone-based voice agent, developers should consider several practical steps:

• Test the agent using real phone calls rather than high-quality laptop microphones.
• Select a speech-to-text model optimized for thin phone audio, specifically testing its accuracy with names and numbers.
• Ensure the system can handle keypad (DTMF) inputs.
• Always provide an option to transfer the caller to a human.
• Review local consent and robocall regulations if the agent is making outbound calls.