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RealtimeSTT

Low-latency speech-to-text library with voice activity detection and wake words.

Realtime Speech-to-Text for Python

Build voice assistants, dictation tools, and streaming servers with minimal code.

Free forever · open source · MIT
✓ Light setup✓ Browser-based✓ Real-time collaboration
8.3
ZEKAI
Low-latency local STT for Python
Free & open source · Gaming
Ease of setup6
Usefulness9
Maintenance9.5
Community8.5
★ 10,237
GITHUB
Oct 2026
UPDATED
MIT
LICENSE
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⚡ Quick answer

RealtimeSTT is a Python library providing voice activity detection, fast transcription, and wake word activation. It suits developers building interactive voice applications or custom streaming servers.

Runs on
Cross-platform
Hosting
Self-host
Languages
Python
License
MIT

What it is

The library captures audio streams, detects speech using WebRTC or Silero VAD, and transcribes audio via engines like faster-whisper or sherpa-onnx. It supports optional wake word detection through Porcupine or OpenWakeWord and includes a packaged production FastAPI server.

Who it's for

It suits Python developers building local voice interfaces, dictation software, or browser streaming backends. It does not suit teams seeking non-Python stacks or zero-dependency setups, as it requires audio drivers and heavy machine learning models.

Key features

Voice Activity Detection

Detects speech automatically using WebRTC VAD and Silero VAD.

Wake Word Activation

Supports optional wake word triggering via Porcupine or OpenWakeWord.

Multiple Engine Support

Runs on faster-whisper, sherpa-onnx, Moonshine, and other selectable backends.

Production FastAPI Server

Includes an optional remote server with versioned HTTP and WebSocket endpoints.

How to start

Up and running in 3 steps

1
Install PortAudio development headers on your operating system.
2
Run `pip install "RealtimeSTT[faster-whisper]"` to install the package and default engine.
3
Import AudioToTextRecorder in your Python script to start capturing and transcribing speech.
Real talk

Our honest verdict

What's great

  • Provides direct microphone input or application-fed external audio chunks.
  • Includes a packaged production FastAPI server with session isolation and authentication.
  • Supports CPU-optimized streaming pipelines using sherpa-onnx models.
  • Offers event callbacks for recording, VAD, realtime text, and wake word states.

Where it falls short

  • Requires platform-specific system dependencies like PortAudio headers on Linux and macOS.
  • Heavy machine learning models demand significant local compute resources or GPU hardware.
  • Python 3.13 and newer are not yet supported release targets.
Cost

Pricing & license

Free under its MIT licence. There are no paid hosted options mentioned in the repository, so the only cost is running the hardware and models yourself.

Paid work

Open remote AI-training roles

876 open remote AI-training roles, updated November 15, 2026; most pay $20–$170 an hour. Zekai lists them free; the application goes to the employer’s platform.

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Alternatives

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Questions

FAQ

How do I feed audio from an external file or stream instead of a microphone?

Initialize the recorder with `use_microphone=False` and feed 16-bit mono PCM chunks using `feed_audio()`.

Can I run RealtimeSTT on a CPU-only production server?

Yes, the README recommends pairing sherpa-onnx-nemotron for live hypotheses with sherpa-onnx-nemo-parakeet for final transcripts on CPU.

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