Understand the project
Practical overview
When it fits
Use Fastrtc when a Python application needs interactive audio or video streaming over WebRTC or WebSockets, particularly for conversational, voice-controlled or media-processing workflows.
Who it is for
- Python developers building browser-based real-time voice or video applications.
- Teams prototyping conversational AI, voice assistants or live customer-support interfaces.
- Developers streaming output from Python AI or media-processing functions to connected clients.
Documented uses
- Expose a Python function through WebRTC as a live audio or video client interface.
- Build a browser-based real-time voice assistant.
- Create live audio or video customer-support experiences.
- Add voice activity detection, pause detection and turn taking.
- Add documented speech-to-text or text-to-speech capabilities to a conversational application.
Before you choose
Requirements & limits
The project is identified as a public, MIT-licensed repository.
- Choose between WebRTC and WebSockets according to the application’s real-time communication design.
- The library is Python-focused, while the repository metadata identifies JavaScript as its primary language.
- The documented vad and tts extras support pause detection and text-to-speech, but the underlying speech models or providers are unspecified.
- The capability is documented, but a complete deployment example is not supplied.
Check current requirements in the official documentation. Source-based guidance does not mean OpenViST has run or independently tested this project.
Evidence and transparency
Sources & review
Source record details
- Source availability
- Active
- Sources checked
- 30 September 2026
- Original website title
- gradio-app/fastrtc · Official homepage title · retained in its source language