Has anyone figured out how to keep Google’s Live Voice Agent from jumping in too fast? I tried it during a practice support call with a 7-second pause built into my script. It handled interruptions so naturally that I kept forgetting I was testing software, but twice it responded before I had finished the actual question.
I checked the available voice settings and repeated the same script 13 times, speaking slower on the last 5 runs. The timing still felt inconsistent. Is there a setting for longer turn detection, or is the better move to rewrite prompts so it waits for a clear verbal cue before answering?
Prompts won’t reliably fix this. Slower speech can actually make turn detection worse because it creates more silence gaps. If you’re using the Live API, increase the silence duration and use low end-of-speech sensitivity. For pauses that long, disable automatic detection and send the end-of-turn signal yourself. A verbal cue like “done” is a decent fallback, but manual turn control is the dependable fix.
Don’t stretch the silence timeout for the whole call. That usually fixes the scripted pause and makes every normal exchange feel sluggish.
@owl1369 is right that prompt wording won’t make turn detection dependable, but full manual control can feel clunky too. Since your long pause is known in advance, handle it as a temporary “listening only” state in the app. Let the audio continue streaming, but block response generation until your script reaches a cue or your UI releases the agent.
That makes sure interrupts are quick everywhere else without relying on the model to guess if a long silence was intentional, and avoid line noise or breathing resetting a silence timer at the worst possible moment.
You probably won’t get perfectly human timing from silence detection alone. A global timeout is the wrong lever, as @owl1369 said, because callers pause unpredictably and background noise can keep restarting the clock.
The temporary listening state makes sense, but watch for a queued reply when you release it. If the agent keeps processing everything during the scripted pause, it may respond to an older fragment the moment generation is enabled again. I’d have the app explicitly mark that section as non-response audio, then commit or clear the pending turn when the cue arrives.
For a practice script, an explicit cue is much more dependable than asking the model to infer intent. Something subtle like a button release, a hidden control event, or a specific end phrase should preserve the fast interruption handling during the rest of the call.
If the silence falls after a sentence that sounds complete, the agent might be tempted to assume that you have finished your turn, so try placing a break after a statement that sets up your point. “The detail I need to check is…”, you could say, followed by a brief pause before stating what you need to say next. The turn detection feature picks up more than simply pauses in the conversation, as it can pick up on the change in tone of voice, so it can be worth leaving sentences unfinished if you are looking to add more breathing space without extending every exchange.
That is not dependable enough for a real customer workflow, but for a fixed practice script it is the quickest workaround. You can rewrite the line before the pause, lower your pitch less at the break, and avoid phrases such as “okay,” “that’s all,” or anything else that sounds like a handoff. Speaking slowly throughout is probably worse, as @owl1369 noted, because it creates lots of smaller openings for the agent to grab.
I’d be careful with completely blocking generation if the point of the exercise is to evaluate how the agent behaves on an actual call. At that point you may be testing your control layer more than the voice agent. @digital_circuit’s queued-response warning matters too. An agent that suddenly answers something from several seconds ago feels more broken than one that interrupts early.
For production, app-controlled turn boundaries are the way to go. For rehearsal, changing where the breaks come in the script is less hassle and leaves the natural interruption behavior intact. If the break has to be exactly where it is and the preceding line has to sound complete, then the early response provides useful test data: the agent can’t reliably distinguish between dramatic silence and a completed turn, the call design needs an explicit state change instead of better prompt wording.
If your client is gating silent audio before it reaches the service, changing the agent’s turn settings may solve the wrong problem. Some microphone, browser, or telephony layers stop sending normal audio frames during a pause, so the server sees an abrupt cutoff rather than someone quietly holding the floor.
Log the timestamps for outgoing audio chunks and the server’s speech-end event. Record the microphone locally at the same time. If the local recording contains the full pause but the outbound stream does not, check client-side voice activity detection, noise suppression, audio buffering, and any telephony silence removal. A headset test can help rule out echo cancellation doing something strange.
I would verify that before rewriting the script as @0xcoder2 suggests. Changing the phrasing is useful for rehearsal, but it can hide a capture problem that will return with real callers.
Once you know the complete pause is reaching the API, then the temporary listening state is the clean fix. Keep transcription active, suppress the response, and explicitly discard or commit the pending turn when the cue arrives. That last action matters because simply unmuting generation can release a stale answer.
If you can send session updates mid-call, you don’t have to pick between a global timeout and full manual control. The Live API lets you push new turn-detection settings during an active session, so you can run tight barge-in sensitivity for the whole conversation and only loosen the end-of-speech threshold right before the scripted pause, then snap it back after. That sidesteps the sluggish-everywhere problem @rocketlogic flagged without handing every turn boundary to your app.
The catch, which few people seem to mention: configuration changes may not always take effect precisely when intended. There can be a small window at the beginning of the new frame where the old configuration still takes effect until the new one fully propagates, and you can always schedule your pause to start precisely at that window. As such, change the configuration a moment before the line that precedes the pause, not on the same line.
@rapidhub2941’s capture point is the one I’d chase first though. I’ve seen enough cases where the client or telephony layer strips silent frames that I wouldn’t touch any turn settings until I confirmed the quiet audio actually reaches the server. If VAD or noise suppression is eating the pause locally, every config tweak you make is guessing at a problem that isn’t where you think it is.
On the queued-reply thing @digital_circuit raised, it’s real and it’s worse than an early interrupt because it feels like the agent lost track of the conversation. If you go the suppression route, clear the pending turn explicitly on your cue. Don’t just re-enable generation and hope the buffer is empty.
For a rehearsal script specifically, I’d lean toward the dynamic config over rewriting the line. Moving the pause like @0xcoder2 suggested works, but then you’re rehearsing against a script you edited to please the model instead of the one you’ll actually use. If the goal is practice, keep the real timing and make the plumbing handle it.
Randomize the pause length during testing and add a little room noise. If your fix only works when the break lands at an exact moment, you’ve built a demo rather than a reliable call flow.
Dynamic settings can still race with the audio stream, so validate the behavior across shorter and longer gaps. The acceptable outcome is a controlled fallback, not perfect human intuition from silence detection.
Nobody has asked what you are doing during those seven seconds. A realistic support pause is rarely completely dead. You type, turn a page, mutter something about pulling the item up. Pure silence is the worst possible test case because it is the most readily interpreted by the model as completion of the task. Decide before touching config whether your rehearsal pause should be silent.
That’s where I’d twist @securelogic5094pro’s room-noise point. Adding ambient sound isn’t just about robustness, it’s closer to how a genuine pause sounds, and it gives the endpointing something to hold onto. If your script has the agent go completely mute and motionless for that stretch, you’re rehearsing an unrealistic version of the call anyway.
The dynamic config idea from @byteguru is the cleanest of the bunch, but I’d rank the capture check ahead of it too. No point loosening a threshold if the quiet frames never leave your machine. Confirm the audio arrives, add a bit of real-world noise to the pause, then adjust settings only if it still trips.