- Introduce new fields in AssistantConfig, schemas, and database models to support vision capabilities, including `vision_enabled` and `vision_model_resource_id`.
- Enhance validation logic in routes to ensure proper handling of vision models and their requirements.
- Update the AssistantPage and related frontend components to include options for enabling vision understanding and selecting appropriate vision models.
- Modify database seed scripts to include vision-related data for assistants, ensuring consistent setup.
- Refactor related functions to integrate vision model handling in the audio-visual processing pipeline.
- Introduce a new `support_image_input` field in model resources, allowing models to indicate support for image input.
- Update the backend models, schemas, and database seed scripts to accommodate the new field.
- Enhance the AssistantConfig and related routes to handle image input capabilities, ensuring proper validation and error handling.
- Modify the frontend components to include toggles for enabling visual understanding and filtering models based on image input support.
- Implement necessary adjustments in the voice preview and pipeline to integrate video stream handling alongside audio functionalities.
- Introduce a new `.env.example` file for environment variable setup, including `PUBLIC_IP`, `TURN_SECRET`, and `TURN_URLS` for WebRTC TURN server configuration.
- Update `docker-compose.yaml` to support TURN server deployment with necessary environment variables and commands.
- Enhance backend configuration and routes to include WebRTC ICE server settings, allowing for STUN/TURN server integration.
- Implement a new service for managing WebRTC ICE server configurations, providing credentials for TURN when configured.
- Modify frontend API to fetch ICE server configurations dynamically, improving support for cross-network voice preview.
- Add new fields to AssistantConfig for realtime interface configuration, including types, values, and secrets.
- Introduce StepFunRealtimeService to handle speech-to-speech processing via WebSocket, integrating STT, LLM, and TTS functionalities.
- Refactor pipeline execution to support a new realtime mode, allowing direct text input processing and immediate responses.
- Update model resource testing to include validation for StepFun Realtime connections.
- Enhance service factory to create realtime services based on configuration settings.
- Modify README documentation to reflect new realtime capabilities and usage instructions.
- Update README to reflect the integration of the DebugVoicePanel with WebSocket support for voice interactions.
- Refactor voice_webrtc.py to improve error handling during WebRTC signaling and include assistant_id in the offer payload.
- Add useVoicePreview hook to manage microphone access and WebRTC connections for real-time voice previews.
- Modify AssistantPage to incorporate new visualizer options and pass assistantId to DebugVoicePanel, enhancing user experience during audio interactions.
- Update API model to include new fields for voice, speed, and language, supporting TTS and ASR configurations.
