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Phone networks (PSTN, SIP, Twilio Media Streams) carry 8 kHz G.711 audio, μ-law in North America and Japan, A-law across most of EMEA/APAC. This page covers how to produce and consume it cleanly: ask Speak for native telephony codecs, decode μ-law to PCM for the recognizer/agent leg, and, only if you truly must, resample with the correct integer ratios.
Never resample by dropping or duplicating samples. Naive decimation (keeping every Nth sample) and sample duplication / zero-stuffing without a low-pass filter alias high-frequency energy back into the speech band. It sounds “tinny” or “robotic” to humans and wrecks transcription accuracy, the recognizer sees energy that was never in the original speech. Always rate-convert with a real resampling filter (polyphase / sinc, or a maintained library).
POST /v1/audio/speech can emit phone-ready audio so you never resample on the client. Shipping now: request a G.711 codec, which is always 8 kHz mono:
The response is raw, headerless G.711 (Content-Type audio/basic), 8-bit companded bytes you can hand straight to the carrier.
For g711_ulaw / g711_alaw the rate is fixed at 8 kHz: omit sample_rate, or pass exactly 8000, any other value is rejected. Speak’s default response_format is wav and its native rate is 24 kHz, so for linear output pass the sample_rate you actually need rather than inheriting 24 kHz.
Because Speak renders at the codec/rate you ask for, requesting g711_ulaw (or pcm at 8000/16000) means no resampling in your code at all, the cleanest possible path.

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Native μ-law: it’s companding, not a sample rate

μ-law (and A-law) is an 8-bit logarithmic companding of 16-bit samples at 8 kHz, a codec, not a rate. Converting between μ-law and 16-bit PCM is a lossless table lookup, not resampling, so it never aliases.
  • Outbound (Speak → phone): g711_ulaw hands you μ-law bytes ready for the carrier, no encode step on your side.
  • Inbound (phone → Hear/Omni): the realtime surfaces take PCM16 (Hear streaming encoding=pcm16; Omni format=pcm16), so decode the carrier’s μ-law to PCM16 first. Keep the session at 8 kHz and the only conversion is companding, never a resample.
μ-law ↔ PCM16 (companding only, no resampling)
Running both legs at 8 kHz (Twilio ↔ μ-law decode ↔ 8 kHz session) is the Twilio guide’s entire codec path: companding only, zero resampling.

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If you must resample: exact integer ratios

Sometimes you can’t choose the rate at the source, e.g. you already hold a 24 kHz asset (Speak’s native rate) and need 16 kHz for Hear or 8 kHz for a phone leg. Rate-convert with these rational ratios, upsample by L, low-pass, then downsample by M: The low-pass filter is not optional, it’s the anti-aliasing (on downsample) and anti-imaging (on upsample) filter that keeps the warning above from biting you. Set its cutoff just below the lower of the two Nyquist frequencies (half the smaller sample rate).
The simplest fix is to not create 24 kHz in the first place: ask Speak for g711_ulaw (telephony), or pcm at sample_rate: 8000 / 16000, and keep capture and sessions on a single rate end to end. Resample only legacy audio you don’t control.

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Common telephony recipes

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Barge-in on noisy / mobile / PSTN lines

On Omni, barge-in is engine-side and automatic: when the caller speaks over the agent, the engine cuts the agent’s audio and emits a flush event; your bridge stops playback (uuid_break, Twilio clear). See the Twilio guide for the wiring. Indian mobile and PSTN lines can still produce false barge-ins when noise looks like deliberate speech. Preserve the telephony bridge’s echo cancellation, resampling, and flush handling; do not discard or delay caller audio locally.
barge_sensitivity is a deprecated compatibility field and is not applied by the current Omni runtime. Do not build tuning logic around it. PyAI will publish a new documented control only after its values are wired to the served VAD policy and covered by telephony regression tests.

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See also

Phone agent with Twilio

The μ-law ↔ PCM16 bridge at 8 kHz, end to end.

Stream speech-to-text

Feed live call audio into Hear streaming.

FreeSWITCH integration

Fork SIP/PSTN audio into Omni at 16 kHz.

Errors & limits

Close codes, rate limits, and concurrency.