Skip to main content
In real-world scenarios, phone calls often face audio quality challenges such as background noise, echo, or other unwanted sounds. We have a couple of different features that are designed to help you handle these challenges.

​
Set denoising mode

Realtime transcription settings showing denoising and transcription controls.

Find Denoising Mode under Realtime transcription settings.

  • No denoising: disables all audio preprocessing and passes the raw audio signal directly to the ASR model. The ASR model itself can handle a minimal level of ambient noise without preprocessing. Choose this mode if you experience issues with missing short responses (e.g. “sure”, “yes”) or degraded accuracy with non-English transcription, particularly when background noise is not significant.
  • Remove noise (default): removes background noise with nearly no distortion to the waveform, so it has no meaningful impact on speech-to-text accuracy. It will not remove loud background speech.
  • Remove noise + background speech: a more aggressive mode that removes both background noise and background speech. This may distort the waveform and can reduce speech-to-text accuracy in some cases. This option incurs a $0.005/min surcharge due to the additional processing required.
If call screening is enabled, do not select Remove noise + background speech. It locks onto the first voice it hears. On screened calls, the first voice is the pre-recorded screening voice, therefore Remove noise + background speech treats the caller’s voice as background speech and filters it out. Use Remove noise or No denoising instead with call screening.
This mode applies background voice cancellation (BVC) to reduce background speech and noise in the caller’s audio. It may also suppress or distort the caller’s speech, so test it with audio representative of your use case before enabling it broadly. Call recordings preserve the raw caller audio from before BVC, so background noise and speech may still be audible in recordings even when filtered out of the audio processed by the agent.
Because this model keeps only the dominant near-mic voice, the speaker you care about must be the clearly dominant voice on the line. If your user is far from the mic, soft-spoken, or no louder than the people around them, the model can mistake them for background speech and filter them out.
When the results can get weird. Because the model aggressively isolates one voice and reshapes the waveform, these situations can produce dropped, clipped, or distorted transcription — test before relying on this mode:
  • No single dominant speaker — speakerphone in a room, two people equally close to the mic, or a handoff between people. The model may pick the wrong voice, switch between voices, or drop one of them.
  • The intended speaker is quiet or distant while background voices are loud (e.g. a far-field mic). Your user can be treated as background and suppressed.
  • Already-clean, single-speaker audio. The extra processing distorts the waveform and can lower accuracy or drop short utterances (e.g. “yes”, “sure”) that Remove noise would keep — with no benefit and an added surcharge.
  • Non-speech or machine audio that you still need transcribed — IVR prompts, voicemail greetings, hold music, or announcements. Aggressive voice isolation can mangle or drop these, since the model is tuned to keep a live human primary speaker.
If your use case typically involves loud background speech — such as a TV or a construction site, with a single caller speaking close to the mic — try the Remove noise + background speech mode, and verify accuracy on real calls before rolling it out. For most use cases, Remove noise is the recommended default.

​
Tuning interruption sensitivity

If background speech or noise still causes unwanted interruptions after denoising, lower Interruption Sensitivity. The range is 0–1. Higher values make the agent easier to interrupt; lower values require the caller to speak longer or say more words before interrupting. At 0, the caller cannot interrupt the agent. The API defaults to 1 when interruption_sensitivity is omitted. New blank voice agents created in the dashboard start at 0.9.
  1. Open Speech settings in the agent settings panel.
  2. Lower Interruption Sensitivity from its current value, for example from 0.9 to 0.8.
  3. Test with representative background noise and check that callers can still interrupt when needed.
Interruption Sensitivity slider set to 0.8 in Speech settings.

Interruption Sensitivity set to 0.8 in Speech settings.

For extremely noisy environments, try lower values. Lower sensitivity reduces false interruptions but also makes intentional caller interruptions harder.

​
Remove noise from the user’s side

As the audio quality is determined by the user’s side, you can also try the following:
  • User side noise reduction: use better microphone & client side noise reduction libraries if using web calls
  • Prompt the agent to ask your users to speak louder so it can be distinguished from background speech easier