Hola — gracias for the clear description. You’re asking why with Groq STT the agent keeps talking until transcription arrives, whereas Pipecat’s agent stops as soon as the user speaks, and whether you can fix that. Quick answer: LiveKit can stop the agent immediately, but it requires audio-level detection (VAD) and adaptive interruption handling — not waiting for full transcripts. What to check and try 1. Ensure Voice Activity Detection (VAD) is enabled for the session. VAD detects speech onset in real time and triggers interruption logic before any transcript arrives. 2. Confirm Adaptive Interruption Handling is enabled (it’s the model that uses acoustic cues to decide to yield the turn). When VAD + adaptive mode are active, the agent can stop within a few hundred ms of user speech. 3. Use Groq in streaming mode (Groq STT supports streaming). Streaming + VAD + adaptive handling gives the lowest-latency interruption behavior. 4. If you still see delays, add a lightweight VAD (e.g., Silero VAD) alongside Groq so the interruption pipeline gets an immediate audio cue while Groq produces partial transcripts. If you want, share: • whether you’re running on LiveKit Cloud or self-hosted, • whether VAD/adaptive interruption are currently enabled, • which Groq model and streaming settings you’re using. I’ll then give exact config pointers or next steps to get the same immediate-stop UX you have in Pipecat. Sources: Adaptive interruption handling | LiveKit Documentation | Groq STT plugin guide | LiveKit Documentation | Solving unwanted interruptions with Adaptive Interruption Handling | LiveKi... | Turns overview | LiveKit Documentation, with interactive elements