I noticed that performance wise when we were testing with Telnyx that sometimes the user speaks but the utterance doesn't get sent to the agent (and WebRTC doesn't seem to log a transcript either) - we speak again and then it works - is this an issue with tuning on STT?

Thanks for reaching out, Arthur. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.

We are using LiveKits default pipeline

Thanks — I’m responding to your update that you’re using LiveKit’s default pipeline. Because you’re on the default pipeline, the most likely causes remain audio preprocessing (noise cancellation), VAD/end-of-utterance segmentation, or very short utterances being dropped before they reach STT. Here are targeted, practical steps you can take and a few quick questions so I can help narrow this down further. What to check and try now 1. Disable client-side noise cancellation for a short test (if enabled) and run the same scenario to see if missing utterances stop. Aggressive cancellation can suppress valid speech. 2. Run tests with slightly longer utterances (add one extra word) to see if capture improves — short 1–3 word replies are commonly missed. 3. Capture client-side logs during a failure and look for any warnings like “silence detected on local audio track” — that shows the audio never made it to the pipeline. 4. Record a short audio sample when the issue happens and feed it directly to your STT provider (or replay it locally) to confirm whether the audio itself is intelligible. 5. If you can reproduce consistently, note the timestamps and I can help interpret logs or guide you on what to extract next. Why these steps help • If disabling noise cancellation fixes it, the preprocessing was dropping speech. • If longer utterances reliably succeed, VAD/segmentation sensitivity is likely the issue. • Client warnings confirm whether the problem is local (mic/routing) vs. pipeline/STT. A couple of clarifying questions • During tests, did you see any client-side warnings or errors in the browser console or SDK logs? • Can you reproduce this reliably (same room, same audio conditions) or is it intermittent across calls? • If possible, can you share a timestamped example of a failed attempt and any logs around that time (console logs, SDK debug output)? A short failed audio clip helps a lot. Next steps I can take once you share more • Recommend specific preprocessing/VAD adjustments to try with the default pipeline. • Help interpret SDK logs or RTCP/track-level indicators to determine if audio is being transmitted. • Walk you through capturing a failed audio sample and testing it with an STT endpoint. If you want, start with disabling noise cancellation for a focused test and tell me whether you see any client warnings — I’ll guide the next diagnostic step.

Its usually on small things like

yes

and

ok

Thanks — I’m responding to your note that the misses are usually very short utterances like “yes” and “ok.” That detail strongly points to VAD/segmentation or preprocessing rejecting very brief speech rather than an issue with the STT model itself. Why short words get missed • Very short utterances (1–2 syllables) are often filtered out by voice activity detection (VAD) or end-of-utterance logic because they look like noise or are shorter than the configured minimum speech segment. • Aggressive noise cancellation can cut the start or entirety of a short word, so the segment never reaches STT. • If the pipeline expects a minimum length or relies on silence windows to finalize segments, brief replies may not cross those thresholds and won’t be sent. Concrete, prioritized things to try 1. Temporarily disable client-side noise cancellation and re-run the test. If “yes”/“ok” start getting captured reliably, reduce the cancellation aggressiveness rather than leaving it off long-term. 2. Increase the VAD/segmentation sensitivity or the allowed minimum speech length in your pipeline: ◦ Relax end-of-utterance silence timeouts so the system waits a little longer before deciding a segment is finished. ◦ Lower the minimum energy/threshold for detecting speech if your pipeline exposes that setting. 1. Add a brief confirmation word or encouraging phrase in the UX (e.g., ask users to say “yes please” instead of “yes”) as a short-term mitigation while you tune detection. 2. Capture a failed short-utterance audio sample (record the client audio) and replay it through the STT to check whether the audio itself is intelligible. If it is, the issue is preprocessing/VAD; if it isn’t, it’s a mic/audio quality problem. 3. During tests, watch client logs for messages like “silence detected on local audio track” or VAD-related warnings — that tells us whether the audio ever left the client. Clarifying question to narrow the fix • Are you able to test with noise cancellation disabled and report whether the short utterances start arriving consistently? • If you can’t test that now, can you capture one or two failed audio samples (and any client console logs at the same time) and share timestamps? That will let me tell you exactly whether preprocessing or VAD is to blame and suggest the precise setting to change. What I can help with next • If you run the noise-cancellation test and tell me the result, I’ll provide exact parameters to adjust in the default pipeline (VAD thresholds and silence timeouts) and recommended aggressiveness for noise cancellation. • If you provide a short failed audio clip or logs, I’ll analyze them and give step-by-step tuning to fix the missed “yes”/“ok” cases. If you want, start by disabling noise cancellation for one test session and tell me whether short utterances are captured — I’ll guide the tuning from there.

It usually is at the start of the conversation only - like the first reply

Thanks — I’m responding to your note that the missed short utterances (like “yes”/“ok”) generally happen only at the very start of a conversation — the first reply. That detail points to timing and startup-related behavior rather than the STT model itself. Likely causes and why • Early-speech drop / pipeline warm-up: When a participant speaks immediately after joining, audio capture, routing, or the STT pipeline may not be fully initialized, so the very first short utterance can be lost. • VAD initialization / start thresholds: Voice activity detection or silence gates can take a moment to stabilize; very short initial words can fall below the detection threshold. • Client audio routing / mic warm-up: The browser or client may not be sending audio frames immediately on join (or may start with a low level), causing the first short word to not reach the server. • Noise cancellation at connection start: Aggressive preprocessing can trim the start of speech while it’s still adapting to the environment. Priority, practical steps to try now 1. Add a short delay before users speak (test): Have the caller wait ~1 second after joining before replying. If the first-word misses stop, it confirms a startup/timing issue. 2. Ensure the user has joined and audio tracks are active before prompting them to speak: ◦ Show a “Ready” indicator only after the local track is confirmed active. 1. Temporarily disable client-side noise cancellation for a quick test to see if the initial word is preserved. 2. Enable or increase any pre-buffering / initial-frame acceptance if available: let the client buffer a small amount of audio before the pipeline begins VAD/segmenting. 3. Relax VAD/start thresholds for the first few seconds after join (if configurable): allow lower energy or treat the first 500–1000 ms as potential speech. 4. Capture startup logs and one failed example: collect browser/SDK console logs and, if possible, a short audio recording of the failed first utterance for replay. Look specifically for messages like “local audio track started” or “silence detected on local audio track.” Short diagnostic checklist • Can you reproduce the issue by speaking immediately on join and see it stop when you wait ~1s? • Do client logs show the local audio track becoming active before the utterance? • Does disabling noise cancellation for the initial test change the behavior? What I can do next for you • If you confirm a timing issue (it reproduces when speaking immediately and goes away with a 1s delay), I’ll give exact client-side checks and code snippets to ensure the track is active before prompting users. • If you can share a short failed audio sample or the SDK console logs around a failure, I’ll analyze and recommend which VAD or startup parameters to tune in the default pipeline. • If you’d rather a quick mitigation, I can suggest UI/UX changes (auto-play a short prompt or “beep” on join) so users naturally wait a moment before replying. If you can try the 1-second wait test and tell me whether that fixes it — and whether you saw any “local audio track” log messages — I’ll provide the next concrete steps to make the fix permanent.