There is no single best sample rate, codec, or volume setting for every voice AI app. Start with the audio path: browser-based real-time voice, an application-managed WebSocket stream, telephone audio, a transcription file, or a custom voice sample. Use the format documented for that specific path; changing a setting alone does not convert audio bytes.
Choose settings for the way your voice AI session runs
Voice AI covers several different workflows, and their audio requirements are not interchangeable. OpenAI’s audio guide separates speech-to-speech agents, voice interfaces for existing agents, file transcription, live captions, translation, speech generation, and audio in an existing app. It points developers toward WebRTC for browser audio and WebSockets for server-managed audio pipelines.
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| Workflow | Who manages audio | Practical starting point |
|---|---|---|
| Browser, real-time conversation | WebRTC negotiates media capture and playback between the browser and service. | Use the documented WebRTC path; do not add a separate audio-format setting when the session negotiates it through SDP. |
| Application-managed real-time stream | Your app handles capture, buffering, playback, and any needed resampling. | Use the WebSocket guide’s supported session format and send audio in the expected encoding. |
| Telephone audio | Depends on the telephony integration. | Follow that integration’s documentation; telephony can use G.711 formats, but do not assume browser settings apply. |
| File transcription or another audio task | Depends on the endpoint and application. | Follow the selected endpoint’s format requirements rather than applying a real-time setting by default. |
These are workflow distinctions, not a ranking of audio quality. A sample rate is meaningful only alongside the API path and encoding that use it.
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For browser real-time voice with OpenAI, use WebRTC’s negotiated media tracks. The WebRTC guide says microphone input and generated speech travel over those tracks, while a data channel carries events such as transcripts and session updates. Because SDP negotiates the media format, omit an explicit audio format in session configuration for this path.
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For the documented GPT-Live WebSocket workflow, choose one supported session format at startup. The session uses one format for both input and output:
| Format | Documented audio | When to use it |
|---|---|---|
audio/pcm, 24,000 Hz |
Mono, signed 16-bit little-endian PCM; default for this WebSocket path. | Use when it matches your configured session and the audio you send or produce. |
audio/pcm, 16,000 Hz |
Mono, signed 16-bit little-endian PCM. | Use when you configure the session for this supported rate. |
audio/pcmu, 8,000 Hz |
G.711 μ-law. | Use when this telephony-style encoding is appropriate to the integration. |
audio/pcma, 8,000 Hz |
G.711 A-law. | Use when this telephony-style encoding is appropriate to the integration. |
These are the options in OpenAI’s WebSocket documentation, not universal settings for other providers or endpoints. For a different service, use its own current documentation.
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What sample rate and channel count should you use?
For the documented GPT-Live WebSocket formats above, PCM is mono at either 24 kHz or 16 kHz; the G.711 options are at 8 kHz. Use the rate that matches the session’s configured format. A mono requirement for this path is not a rule that every voice AI product accepts only mono.
If your microphone or source audio runs at a different rate from the configured session, resample the audio before sending it. Selecting a rate in configuration does not resample the source or convert its encoding. For PCM16, send complete 16-bit samples so chunks do not split a sample.
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How to send and play WebSocket audio correctly
- Set the session format at startup. Select one supported encoding and rate for both input and output.
- Prepare the source to match. Resample if its rate differs, and encode it in the selected format. Base64-encode raw audio bytes; do not include a WAV or other container header.
- Stream microphone audio as live input. Send continuously at the rate the microphone audio was recorded. Sending an entire file at once is not equivalent to simulating live microphone capture.
- Keep PCM chunks sample-aligned. A PCM16 chunk should contain complete 16-bit samples.
- Play output deltas in sequence. Decode each returned audio chunk and queue it in order rather than playing chunks as unrelated clips.
In this arrangement, the application is responsible for audio capture, buffering, playback, and any necessary resampling. The operational details are documented in OpenAI’s WebSocket guide.
How to record a custom voice sample
For an OpenAI custom voice sample, recording conditions matter more than chasing a universal numerical setting. OpenAI recommends recording in a quiet space with minimal echo, using a professional XLR microphone, placing a pop filter between the speaker and microphone, and maintaining a consistent distance of about 7–8 inches. These are recommendations for custom voice creation, not minimum equipment requirements for ordinary browser voice chats.
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Custom voice creation also requires a separate consent recording, and the voice in the sample must match the consent recording. OpenAI’s custom voices guide does not give a measured improvement percentage or compare microphone models, so it does not establish a best brand or a guaranteed quality gain from buying particular gear.
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Why does my AI voice recording sound noisy or cut out?
If recognition is affected by noise or interruptions
Background noise, overlapping speakers, long pauses, network conditions, and microphone settings can affect ChatGPT Voice interactions, according to the ChatGPT Voice support guidance. Try moving to a quieter place, using headphones, or adjusting the device volume when interruptions occur. These are troubleshooting suggestions, not a universal headphone requirement or recommended output-volume target.
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If a WebSocket stream is malformed or plays unevenly
- Check that the configured session format matches the bytes being sent.
- Resample when the source rate and configured rate differ; changing the setting alone does not convert the audio.
- For PCM16, make sure each chunk contains complete samples.
- Send raw audio bytes in the expected encoding rather than a WAV file header plus audio.
- Decode output chunks and play them in order.
If a browser session has unexpected format settings
For the documented WebRTC route, leave explicit audio-format configuration out of the session configuration: the format is negotiated through SDP. Use the appropriate API documentation if the session is not actually using that browser WebRTC path.
Do you need a microphone for voice AI?
For a custom voice sample, OpenAI recommends an XLR microphone and pop filter. For other voice AI workflows, the sources do not establish a universal microphone requirement or identify a best consumer microphone. The choice depends on the product and device; first improve the conditions you can control, especially room noise, echo, and consistent mic distance when recording a sample.
Quick Recap
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