voxtra 0.3.0

Open voice infrastructure for AI agents — bridging telephony and AI

Voxtra

Open voice infrastructure for AI agents.

Built by Rexplore Research Labs

Voxtra is a Python framework that bridges telephony infrastructure (Asterisk, FreeSWITCH, LiveKit) with AI voice agents (STT, LLM, TTS). It lets developers build AI-powered call centers without needing to understand telecom internals.

Architecture

graph LR
    A[Cellular Provider] -->|SIP Trunk| B[Asterisk PBX]
    B -->|ARI + Media| C[Voxtra]
    C --> D[STT]
    C --> E[LLM]
    C --> F[TTS]

    D -->|transcript| E
    E -->|response| F
    F -->|audio| C

    style A fill:#4a90d9,stroke:#333,color:#fff
    style B fill:#e67e22,stroke:#333,color:#fff
    style C fill:#2ecc71,stroke:#333,color:#fff
    style D fill:#9b59b6,stroke:#333,color:#fff
    style E fill:#e74c3c,stroke:#333,color:#fff
    style F fill:#1abc9c,stroke:#333,color:#fff

Layer Design

Layer	Package	Responsibility
Core	voxtra.app, voxtra.router, voxtra.session	App lifecycle, routing, call sessions
Telephony	voxtra.telephony	Asterisk ARI, LiveKit, FreeSWITCH adapters
Media	voxtra.media	Audio frames, WebSocket/RTP transport, codecs
AI	voxtra.ai	STT, TTS, LLM, VAD provider abstractions
Pipeline	voxtra.core.pipeline	Real-time STT → LLM → TTS orchestration

Quick Start

Installation

From PyPI:

pip install voxtra

With provider extras:

pip install voxtra[asterisk,deepgram,openai,elevenlabs]

From GitHub (latest development version):

pip install git+https://github.com/rexplore-ai/voxtra.git

From source (for development):

git clone https://github.com/rexplore-ai/voxtra.git
cd voxtra
python3 -m venv .venv && source .venv/bin/activate
pip install -e ".[dev]"

Code-First Usage

from voxtra import VoxtraApp

app = VoxtraApp.from_yaml("voxtra.yaml")

@app.route(extension="1000")
async def support_call(session):
    await session.answer()
    await session.say("Hello, welcome to support. How can I help you?")
    text = await session.listen()
    reply = await session.agent.respond(text)
    await session.say(reply.text)
    await session.hangup()

app.run()

Config-First Usage

Create voxtra.yaml:

app_name: my-call-center

telephony:
  provider: asterisk
  asterisk:
    base_url: http://localhost:8088
    username: asterisk
    password: secret
    app_name: voxtra

media:
  transport: websocket
  codec: ulaw
  sample_rate: 8000

ai:
  stt:
    provider: deepgram
    api_key: ${DEEPGRAM_API_KEY}
    model: nova-2
  llm:
    provider: openai
    api_key: ${OPENAI_API_KEY}
    model: gpt-4o
    system_prompt: "You are a helpful voice assistant for a call center."
  tts:
    provider: elevenlabs
    api_key: ${ELEVENLABS_API_KEY}
    voice_id: your-voice-id

routes:
  - extension: "1000"
    agent: support_agent

Then run:

voxtra start

Asterisk Integration

Voxtra connects to Asterisk via ARI (Asterisk REST Interface). Add this to your Asterisk dialplan:

[voxtra-inbound]
exten => _X.,1,Stasis(voxtra)
 same => n,Hangup()

Supported Providers

Telephony

• Asterisk (ARI) — Production ready
• LiveKit (SIP) — Planned
• FreeSWITCH — Planned

Speech-to-Text

• Deepgram (streaming)
• More coming soon

LLM / Agents

• OpenAI (GPT-4o, streaming)
• LangGraph integration planned

Text-to-Speech

• ElevenLabs (streaming)
• More coming soon

Project Structure

src/voxtra/
├── app.py                  # VoxtraApp — main entry point
├── session.py              # CallSession — per-call handle
├── router.py               # Decorator-based call routing
├── events.py               # Event system
├── config.py               # Pydantic config models
├── middleware.py            # Event middleware
├── exceptions.py           # Custom exceptions
├── types.py                # Shared types
├── core/
│   └── pipeline.py         # STT → LLM → TTS pipeline
├── telephony/
│   ├── base.py             # TelephonyAdapter ABC
│   ├── asterisk/           # Asterisk ARI adapter
│   └── livekit/            # LiveKit adapter (stub)
├── media/
│   ├── audio.py            # AudioFrame, codec conversion
│   ├── base.py             # MediaTransport ABC
│   ├── websocket.py        # WebSocket transport
│   └── buffer.py           # Audio buffering
└── ai/
    ├── stt/                # Speech-to-Text providers
    ├── tts/                # Text-to-Speech providers
    ├── llm/                # LLM / Agent providers
    └── vad/                # Voice Activity Detection

Documentation

• Architecture — Deep-dive into every layer, component, data flow, and design decision
• Contributing — How to set up dev environment, add providers, submit PRs, and code standards

Development

git clone [e-mail]:rexplore-ai/voxtra.git
cd voxtra
python3 -m venv .venv && source .venv/bin/activate
pip install -e ".[dev]"
pytest

Roadmap

• Core abstractions (VoxtraApp, Router, CallSession, Events)
• Asterisk ARI adapter
• AI provider interfaces (STT, TTS, LLM, VAD)
• WebSocket media transport
• Voice pipeline (STT → LLM → TTS)
• End-to-end Asterisk + AI demo
• LiveKit adapter
• FreeSWITCH adapter
• LangGraph agent integration
• Multi-agent handoff
• Dashboard / Admin API
• Conversation analytics

Contributors

Thanks to everyone who has contributed to Voxtra!

Patrick Byamasu — Creator & Lead Maintainer

Want to contribute? Check out our Contributing Guide.

License

Apache 2.0 — See LICENSE

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Voxtra — The LangGraph of AI Telephony Built by Rexplore Research Labs