AI Voices for Live Narration & Streaming

AI Voices for Live Narration & Streaming

Generation runs at 3.3 times real time, so audio renders faster than it plays and the buffer stays ahead. First audio arrives in about 200 milliseconds over SSE or WebSocket.

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realtime narration voice ai
PratikVerified
Indian · Male · Young
India flagHindi
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MishaVerified
Indian · Female · Young
India flagHindi
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PariVerified
Indian · Female · Young
India flagHindi
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GowthamVerified
Indian · Male · Young
India flagHindi
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LucaVerified
Italian · Male · Young
Italian
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JuneVerified
Korean · Female · Young
Korean
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PratikVerified
Indian · Male · Young
India flagHindi
Use voice
MishaVerified
Indian · Female · Young
India flagHindi
Use voice
PariVerified
Indian · Female · Young
India flagHindi
Use voice
GowthamVerified
Indian · Male · Young
India flagHindi
Use voice
LucaVerified
Italian · Male · Young
Italian
Use voice
JuneVerified
Korean · Female · Young
Korean
Use voice

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Narration is the one job where a voice has to hold up for hours rather than seconds. 163 of the 244 voices here carry the narration tag, the largest group in the catalog, across English, Hindi and nine other Indic languages.

1500+ voices to explore

Browse our full voice library and find the right tone, accent, and personality for your agent.

Model Specifications

Streaming transports

WebSocket, HTTP chunked transfer

WebSocket, HTTP chunked transfer

Output formats

.ulaw, .alaw, PCM 16-bit, .wav, .mp3

.ulaw, .alaw, PCM 16-bit, .wav, .mp3

Speed range

0.5× – 2.0×, set per request

0.5× – 2.0×, set per request

Sample rate

44.1 kHz native, resampled to 8 kHz for telephony

44.1 kHz native, resampled to 8 kHz for telephony

Latency

~200 ms to first byte (p50, warm region)

~200 ms to first byte (p50, warm region)

Generation outruns playback

Synthesis runs at 3.3 times real time, so audio renders faster than it plays and the buffer stays ahead of the listener. First audio arrives in around 200 milliseconds. For live commentary that margin is what makes continuous delivery possible rather than merely fast.

Planning for a dropped connection

A socket closing mid stream leaves playback truncated, and that decision belongs in your application. Retry, fall back to a cached holding clip, or fail silently are the options. For live narration a short pre-rendered holding line is usually better than dead air while a retry completes.

Transport decides the experience

The synchronous endpoint waits for a complete clip, which is fine for files and wrong for live work. Use SSE for one directional streaming, or WebSocket where text arrives incrementally and playback should begin mid sentence. On live material the pause before audio is the thing listeners notice.

Writing for a 250 character window

Requests cap at 250 characters, so continuous narration is a sequence of short calls rather than one long stream. That suits live commentary, which is naturally clause by clause, but it does mean your text pipeline needs to segment sensibly rather than cutting mid phrase.

Popular voices

Frequently asked questions

Roughly 200 milliseconds, with generation running at 3.3 times real time thereafter. Use the SSE or WebSocket transport rather than the synchronous endpoint, since those begin returning audio while the rest is still rendering. The synchronous endpoint waits for the full clip.