AI Voices for IVR & Telephony
IVR voices need to survive a compressed phone line, not just sound good in a browser. These output ulaw and alaw directly, generate in roughly 200 milliseconds, and handle Hindi and English in the same prompt.
VOICES FOR THIS USE CASE
Why IVR is a different synthesis problem
Phone audio is narrowband and heard once, with no chance to replay. A voice that sounds warm on studio monitors can lose its consonants after ulaw encoding. What matters here is intelligibility under codec loss and stable pacing across renders, so a prompt sounds the same on the ten thousandth call as on the first.
Latency and what to pre-render
Time to first audio averages around 200 milliseconds on the streaming endpoints, which sits inside the window a caller reads as immediate. Stream over WebSocket for interactive turns. Anything you play more than once, such as a welcome message or a menu, should be generated ahead of time and stored rather than synthesised on every call.
Code-mixed Hindi and English
Indian phone prompts rarely stay in one language. Words like appointment, confirm and OTP arrive inside Devanagari sentences and need to be pronounced as English rather than transliterated. Voices in the Indic family handle inline Latin tokens without switching register mid-sentence, which is why they suit Indian call flows better than an English-only voice.
Output formats for telephony stacks
Synthesis runs at 44.1 kHz natively. Request ulaw or alaw when the audio is going straight into a phone path, since those are the codecs telephony platforms expect and no conversion step is needed. PCM, MP3 and WAV remain available for reviewing or archiving a prompt outside the call flow.
SPECIFICATION
Sample rate
44.1 kHz native, resampled to 8 kHz for telephony
Latency
~200 ms to first byte (p50, warm region)
Output formats
ulaw, alaw, PCM 16-bit, WAV, MP3
Streaming transports
WebSocket, HTTP chunked transfer
Speed range
0.5× – 2.0×, set per request
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AI Voices for IVR & Telephony
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