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Realtime TTS API

The #1 realtime TTS API. Sub-100ms latency. Ship voice features in minutes, not weeks. From prototype to production on one API.

Realtime API response times

Audio chunks arrive before users notice a delay. REST, streaming, and WebSocket endpoints built for speed.
Latency (TTFB)
100 ms TTFB* for Realtime TTS-2 and 20 ms for Realtime TTS-2 Flash. Your slowest API calls still feel instant to end users. *P90 time to first audio byte, measured server-side — excludes network latency.
Time to first audio byte*
~0ms
Realtime TTS-2 Flash
~0ms
Realtime TTS-2
~0ms
Human
WebSocket streaming
Persistent bidirectional connections for realtime synthesis. Audio streams as it's generated — no buffering, no polling. Ideal for LLM-powered voice agents.

Top-ranked voice quality

Expressive, stable output that keeps users listening, validated by thousands of blind tests.
Top-ranked on public benchmarks. Truly expressive.
Top-ranked on independent speech benchmarks, scored by thousands of listeners in blind comparisons. Realtime TTS-2 pairs that quality with natural-language steering and reliable non-verbals.
Low error rate
40% fewer word errors than Realtime TTS 1. Fewer hallucinations, fewer cutoffs, fewer audio artifacts in production. Less post-processing, fewer edge cases to handle in your code.
Voice cloning via API
Clone any voice with a single API call. Pass 5 to 15 seconds of reference audio to get a unique voiceId, then use it in any TTS request. Professional fine-tuning available for maximum fidelity.

Built for massive scale

Top-ranked voice, a fraction of the cost: Realtime TTS-2 as low as $5 per 1M characters, falling further as you scale. Multilingual and on-prem ready.
Enhanced multilingual
English, Spanish, French, Korean, Chinese, Hindi, Japanese, German, and more. Native-quality output in every language. Deploy globally without separate pipelines.
A user interface demonstrating multilingual support, with a dropdown menu to select languages such as Korean, Spanish, and German.
On-prem deployment
Run high-quality text-to-speech models locally — without sending text or audio data to the cloud. Built for enterprises that require strict data control, low latency, and compliance with internal or regulatory standards.
Voice AI down to $0.005/min
As low as $5/1M characters for Realtime TTS-2 at scale, after price cuts of half or more for most developers. At a fraction of a cent per minute, realtime voice can be always-on, not rate-limited by cost.

Full breakdown

Best for
Realtime TTS-2
Most expressive applications
Realtime TTS-2 Flash
Latency-critical, high-volume applications
Pricing (enterprise scale)
Realtime TTS-2
Down to $5/million characters
Realtime TTS-2 Flash
Sub-$5/million characters
Latency (TTFB) *
Realtime TTS-2
100 ms
Realtime TTS-2 Flash
20 ms
Quality
Realtime TTS-2
Highest expressiveness, native steering
Realtime TTS-2 Flash
High expressiveness
Multilingual
Realtime TTS-2
200+ languages, cross-lingual (BCP-47)
Realtime TTS-2 Flash
200+ languages, cross-lingual (BCP-47)
Natural-language steering
Realtime TTS-2
Realtime TTS-2 Flash
Non-verbal cues
Realtime TTS-2
Realtime TTS-2 Flash
Professional voice cloning
Realtime TTS-2
Realtime TTS-2 Flash
Voice cloning
Realtime TTS-2
Realtime TTS-2 Flash
Character, word, viseme and phoneme timestamps
Realtime TTS-2
Realtime TTS-2 Flash
Custom pronunciation
Realtime TTS-2
Realtime TTS-2 Flash
On-premise
Realtime TTS-2
Realtime TTS-2 Flash
Zero data retention
Realtime TTS-2
Realtime TTS-2 Flash
*P90 time to first audio byte, measured server-side — excludes network latency.

Research

Integrations

Try Realtime TTS now

Get started with Realtime TTS-2, the flagship for expressiveness and steering. Pick Realtime TTS-2 Flash for the latency and cost floor.

FAQs

A TTS (text-to-speech) API converts written text into spoken audio via HTTP requests. You send text to an endpoint and receive audio back-as MP3, WAV, or streaming chunks. TTS APIs power voice assistants, audiobook generation, accessibility features, video narration, and AI voice agents. Modern APIs like Inworld offer realtime streaming, instant voice cloning, and support for 200+ languages with Realtime TTS-2.
Realtime TTS is top-ranked on independent speech benchmarks, including the Artificial Analysis TTS Arena, a public leaderboard based on blind listening tests by thousands of real users. It combines top-ranked voice quality with sub-100ms latency and pricing that scales down to as low as $5 per million characters. For most applications, start with Realtime TTS-2, the flagship for expressiveness and steering. Pick Realtime TTS-2 Flash for the latency and cost floor.
Realtime TTS is top-ranked on Artificial Analysis. Realtime TTS-2 Flash delivers 20 ms TTFB down to $5/1M characters, while Realtime TTS-2 pairs 100 ms TTFB with the best quality and natural-language steering. Inworld also provides WebSocket streaming for realtime voice agents, on-premise deployment, and instant voice cloning from 5 to 15 seconds of audio.
Realtime TTS-2 runs as low as $5 per million characters at scale, about $0.005 per minute of audio. Realtime TTS-2 Flash starts at $15 per million and falls to $7. Rates fall as your total spend grows, and spend on TTS counts toward one combined commit with the Router, speech-to-text, and compute. Exact tiers and a cost calculator are at inworld.ai/pricing.
Make a POST request to the TTS endpoint with your text, voice ID, and model. You'll receive audio back as base64-encoded data. For realtime apps, use WebSocket streaming to receive audio chunks as text is generated. Works with Python, Node.js, curl, or any HTTP client. Generate an API key and check the Developer Quickstart for code examples.
Inworld Realtime TTS-2 Flash delivers 20 ms TTFB*, and Realtime TTS-2 runs 100 ms while adding natural-language steering. Both support native WebSocket streaming where audio generates instantly with no buffering delay. This makes Inworld ideal for voice agents, interactive media, and any app where response time is critical.
*P90 time to first audio byte, measured server-side — excludes network latency.
Realtime TTS is top-ranked for voice quality on Artificial Analysis, based on blind tests by thousands of real users. Realtime TTS-2 is optimized to minimize hallucinations, word cutoffs, and audio artifacts, and holds one voice identity across every language it speaks. Try it in the Playground.
Yes. Inworld offers instant voice cloning: upload 5 to 15 seconds of audio and get a unique voice ID to use in TTS requests immediately. For maximum fidelity, professional voice cloning uses 30+ minutes of clean audio (minimum 5 minutes, 20+ minutes recommended). Contact sales for professional voice cloning.
Realtime TTS-2 and Realtime TTS-2 Flash support 200+ languages with one voice identity preserved across every language and mid-utterance switching inside a single generation. Use the List Voices endpoint to filter available voices by language.
REST returns complete audio after all text is processed-best for batch jobs and short text. HTTP streaming returns audio chunks progressively for faster playback start. WebSocket maintains a persistent bidirectional connection, ideal for voice agents where text is generated incrementally by an LLM. Inworld supports all three from a single API.
Our published latency figures are on-server inference time, which measures how long our models take to generate the first audio chunk once the request reaches our servers:
  • Realtime TTS-2: 100 ms TTFB
  • Realtime TTS-2 Flash: 20 ms TTFB
Your end-to-end latency also includes network round-trip time between your application and our servers.
  • Use WebSocket streaming to avoid repeated connection overhead (Python, JavaScript)
  • Choose the server region closest to your users
  • Consider on-premise deployment for latency-critical applications
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