Low-latency speech-to-text library with voice activity detection and wake words.
Build voice assistants, dictation tools, and streaming servers with minimal code.
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.
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.
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.
Detects speech automatically using WebRTC VAD and Silero VAD.
Supports optional wake word triggering via Porcupine or OpenWakeWord.
Runs on faster-whisper, sherpa-onnx, Moonshine, and other selectable backends.
Includes an optional remote server with versioned HTTP and WebSocket endpoints.
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.
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.
Initialize the recorder with `use_microphone=False` and feed 16-bit mono PCM chunks using `feed_audio()`.
Yes, the README recommends pairing sherpa-onnx-nemotron for live hypotheses with sherpa-onnx-nemo-parakeet for final transcripts on CPU.
See Zekai first in Google