Bridge between SIP server and OpenAI Realtime voice agent, controlled by Hermes via MCP
View on GitHub ↗Agent-SIP Ideal for Hermes and n8n workflows
Agent-SIP — a self-hosted SIP voice bridge that can be controlled with AI agents via MCP.
Bring an AI agent to an ordinary phone extension. Agent-SIP registers with your PBX, sends and receives G.711 audio over RTP, and gives you a web dashboard, automation webhooks, and MCP tools for controlling calls from an assistant.
OpenAI's official Realtime SIP integration is designed around Twilio's cloud telephony, but Agent-SIP works with any local SIP server—including Asterisk, FreePBX, 3CX, Kamailio, or a carrier trunk. It speaks SIP directly using UDP signaling and RTP audio, so it doesn't rely on a cloud telephony provider. In our own setup, it runs as extension 500 on a FreePBX box.
Select any screenshot to open the full-size image.
- Real phone calls through a SIP extension using UDP signaling and RTP audio (PCMU or PCMA). - OpenAI Realtime speech with natural multilingual conversations, French defaults, configurable language and the marin voice. - CALL BRIEF objectives that tell the agent what to accomplish on each outbound call. - MCP control to make calls, steer the agent, speak, hang up, retrieve transcripts, inspect status, and save messages. - Automation webhooks for call and transcript events—ideal for n8n, Hermes, or your own service. - Background office ambience with selectable bundled sounds and adjustable volume. - Authenticated web UI for calls, live transcripts, logs, configuration, and prompts. - Simple deployment with Docker and a published GHCR image. - Call safeguards including configurable ring limits, ring
From the project README.
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