active-call 0.3.18

A SIP/WebRTC voice agent
Documentation

Active Call

active-call is a standalone Rust crate designed for building AI Voice Agents. It was originally decoupled from rustpbx to provide a dedicated, high-performance library for voice agent integration.

Overview

This crate handles the complex low-level details of voice communication, making it easy to connect LLM/NLP engines with telephony platforms. It manages:

  • ASR (Automatic Speech Recognition): Handling speech-to-text transitions and events.
  • TTS (Text-to-Speech): Managing real-time audio synthesis and playback.
  • SIP & WebRTC: Bridging traditional telephony and modern web communication protocols.
  • Voice Details: Managing audio buffers, codecs (like Opus), and real-time streaming.

By abstracting these technical hurdles, active-call allows developers to focus on building intelligent dialogue systems rather than worrying about the underlying voice infrastructure.

Key Features

  • Standalone Crate: Decoupled from the main PBX logic for better modularity.
  • LLM/NLP Friendly: Designed to easily feed ASR results into LLMs and stream TTS responses back to the caller.
  • Multi-Protocol Support: Supports SIP, WebRTC, and raw WebSocket audio streams.
  • Real-time Performance: Built with Rust for low-latency audio processing.

API Documentation

For detailed information on REST endpoints and WebSocket protocols, please refer to the API Documentation.

License

This project is licensed under the MIT License.