Best AI Voice Models for Startups
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Compare the Top AI Voice Models for Startups as of October 2026
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What are AI Voice Models for Startups?
AI voice models are artificial intelligence models designed to generate, understand, recognize, and transform human speech using machine learning and advanced audio processing technologies. These models power capabilities such as text-to-speech (TTS), speech-to-text (STT), voice cloning, speech synthesis, voice conversion, real-time voice conversations, and multilingual speech generation. AI voice models often use neural networks, transformers, diffusion models, and multimodal architectures to produce natural-sounding speech with realistic intonation, emotion, pacing, and speaker characteristics. Many AI voice models are available through APIs, SDKs, open-source frameworks, and cloud platforms, enabling developers to integrate voice capabilities into AI assistants, conversational agents, contact centers, accessibility tools, media production workflows, and enterprise applications. By enabling natural, low-latency voice interactions and automating speech-related tasks, AI voice models help organizations create more accessible, engaging, and intelligent voice-powered experiences. Compare and read user reviews of the best AI Voice Models for Startups currently available using the table below. This list is updated regularly.
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1
Zyphra Zonos
Zyphra
Zyphra is excited to announce the release of Zonos-v0.1 beta, featuring two expressive and real-time text-to-speech models with high-fidelity voice cloning. We are releasing our 1.6B transformer and 1.6B hybrid under an Apache 2.0 license. It is difficult to quantitatively measure quality in the audio domain; we find that Zonos’ generation quality matches or exceeds that of leading proprietary TTS model providers. Further, we believe that openly releasing models of this caliber will significantly advance TTS research. Zonos model weights are available on Huggingface, and sample inference code for the models is available on our GitHub. You can also access Zonos through our model playground and API with simple and competitive flat-rate pricing. We have found that quantitative evaluations struggle to measure the quality of outputs in the audio domain, so for demonstration, we present a number of samples of Zonos vs both proprietary models.Starting Price: $0.02 per minute -
2
Octave TTS
Hume AI
Hume AI has introduced Octave (Omni-capable Text and Voice Engine), a groundbreaking text-to-speech system that leverages large language model technology to understand and interpret the context of words, enabling it to generate speech with appropriate emotions, rhythm, and cadence, unlike traditional TTS models that merely read text, Octave acts akin to a human actor, delivering lines with nuanced expression based on the content. Users can create diverse AI voices by providing descriptive prompts, such as "a sarcastic medieval peasant," allowing for tailored voice generation that aligns with specific character traits or scenarios. Additionally, Octave offers the flexibility to modify the emotional delivery and speaking style through natural language instructions, enabling commands like "sound more enthusiastic" or "whisper fearfully" to fine-tune the output.Starting Price: $3 per month -
3
Piper TTS
Rhasspy
Piper is a fast, local neural text-to-speech (TTS) system optimized for devices like the Raspberry Pi 4, designed to deliver high-quality speech synthesis without relying on cloud services. It utilizes neural network models trained with VITS and exported to ONNX Runtime, enabling efficient and natural-sounding speech generation. Piper supports a wide range of languages, including English (US and UK), Spanish (Spain and Mexico), French, German, and many others, with voices available for download. Users can run Piper via the command line or integrate it into Python applications using the piper-tts package. The system allows for real-time audio streaming, JSON input for batch processing, and supports multi-speaker models. Piper relies on espeak-ng for phoneme generation, converting text into phonemes before synthesizing speech. It is employed in various projects such as Home Assistant, Rhasspy 3, NVDA, and others.Starting Price: Free -
4
Qwen3-TTS
Alibaba
Qwen3-TTS is an open source series of advanced text-to-speech models developed by the Qwen team at Alibaba Cloud under the Apache-2.0 license, offering stable, expressive, and real-time speech generation with features such as voice cloning, voice design, and fine-grained control of prosody and acoustic attributes. The models support 10 major languages, including Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, and Italian, and multiple dialectal voice profiles with adaptive control over tone, speaking rate, and emotional expression based on text semantics and instructions. Qwen3-TTS uses efficient tokenization and a dual-track architecture that enables ultra-low-latency streaming synthesis (first audio packet in ~97 ms), making it suitable for interactive and real-time use cases, and includes a range of models with different capabilities (e.g., rapid 3-second voice cloning, custom voice timbres, and instruction-based voice design).Starting Price: Free -
5
Cartesia Ink-Whisper
Cartesia
Cartesia Ink is a family of real-time streaming speech-to-text (STT) models designed to power fast, natural conversations in voice AI applications, acting as the “voice input” layer that converts spoken language into accurate text instantly. Its flagship model, Ink-Whisper, is specifically engineered for conversational environments, delivering ultra-low latency transcription with a time-to-complete-transcript as fast as 66 milliseconds, enabling fluid, human-like interactions without noticeable delays. Unlike traditional transcription systems built for batch processing, Ink is optimized for live dialogue, handling fragmented, variable-length audio through dynamic chunking, which reduces errors and improves responsiveness during pauses, interruptions, or rapid exchanges.Starting Price: $4 per month -
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Realtime TTS-2
Inworld
Realtime TTS-2 from Inworld AI is a new generation of voice model built for real-time conversation: a voice model that feels as human as it sounds. It hears the full audio of an exchange, picks up the user’s tone, pacing, and emotional state, then takes voice direction in plain English, the way developers prompt an LLM. Instead of generating speech in isolation, it listens to prior turns of the exchange, so tone and pacing carry forward, and the same line can land differently after a joke than after bad news. Voice Direction lets developers steer delivery like a director would steer a voice actor, using natural-language descriptions rather than fixed emotion presets or sliders. Inline nonverbals like [sigh], [breathe], and [laugh] can be placed inside the text, and the model renders them as audio events. Realtime TTS-2 preserves one voice identity across more than 100 languages, including mid-utterance language switches.Starting Price: $25 per month -
7
aiOla
aiOla
aiOla is a deep tech Conversational, Voice, and Speech AI lab with an enterprise-level automatic speech recognition (ASR) foundation model, Text-to-speech (TTS) technology and Natural Language Understanding (NLU). It’s designed to help enterprises and developers adapt speech technologies to any process, whether through seamless API integration or an intuitive in-house app. aiOla is revolutionizing enterprise operations with enterprise level Conversational AI. We specialize in speech-to-text and text-to-speech AI that deliver unmatched accuracy (95%), specialized in specific jargon, in any language, accent, vertical, or acoustic environment. From empowering frontline workers with hands-free workflows to enabling voice AI agents with enterprise-grade ASR and TTS, aiOla seamlessly integrates into workflows, internal apps and products. -
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Replica
Replica
Replica Studios provides cutting edge text to speech, and speech to speech solutions in multiple languages for creative professionals, with fully licensed AI models safe for commercial use. Replica Studios offers two products: Replica Voice Director: Generate voice overs and dialogue instantly with text to speech OR speech to speech, while also managing the scripts for your project where it’s all tracked in one place. Access thousands of unique, natural-sounding, expressive AI voices tailored for specific projects or brands, such as content creators, audiobooks, corporate videos, educational content, games, and open-world games. Replica Voice Lab: Design unique human quality AI voices that can perform in multiple languages in seconds with Replica Studios Voice Lab. Blend up to 5 voice personas to create unique voices, with unique and interesting styles and accents. Multi Language Support: Localize and dub your content using our multi-lingual generative AI voice generator.Starting Price: $10 per month -
9
Hume AI
Hume AI
Our platform is developed in tandem with scientific innovations that reveal how people experience and express over 30 distinct emotions. Expressive understanding and communication is critical to the future of voice assistants, health tech, social networks, and much more. Applications of AI should be supported by collaborative, rigorous, and inclusive science. AI should be prevented from treating human emotion as a means to an end. The benefits of AI should be shared by people from diverse backgrounds. People affected by AI should have enough data to make decisions about its use. AI should be deployed only with the informed consent of the people whom it affects.Starting Price: $3/month -
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Kokoro TTS
Kokoro TTS
Kokoro TTS is an efficient text-to-speech tool with multilingual and customizable voice support. Its 182M parameter architecture delivers high-quality audio, supporting languages like American English, British English, French, Korean, Japanese, and Mandarin. It features lifelike voice options, automatic content segmentation, and OpenAI compatibility, facilitating content creation and application integration. With NVIDIA GPU acceleration, it ensures real-time audio generation, making it suitable for various projects.Starting Price: $0 -
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Orpheus TTS
Canopy Labs
Canopy Labs has introduced Orpheus, a family of state-of-the-art speech large language models (LLMs) designed for human-level speech generation. These models are built on the Llama-3 architecture and are trained on over 100,000 hours of English speech data, enabling them to produce natural intonation, emotion, and rhythm that surpasses current state-of-the-art closed source models. Orpheus supports zero-shot voice cloning, allowing users to replicate voices without prior fine-tuning, and offers guided emotion and intonation control through simple tags. The models achieve low latency, with approximately 200ms streaming latency for real-time applications, reducible to around 100ms with input streaming. Canopy Labs has released both pre-trained and fine-tuned 3B-parameter models under the permissive Apache 2.0 license, with plans to release smaller models of 1B, 400M, and 150M parameters for use on resource-constrained devices. -
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MARS6
CAMB.AI
CAMB.AI's MARS6 is a groundbreaking text-to-speech (TTS) model that has become the first speech model accessible on Amazon Web Services (AWS) Bedrock platform. This integration allows developers to incorporate advanced TTS capabilities into generative AI applications, facilitating the creation of enhanced voice assistants, engaging audiobooks, interactive media, and various audio-centric experiences. MARS6's advanced algorithms enable natural and expressive speech synthesis, setting a new standard for TTS conversion. Developers can access MARS6 directly through the Amazon Bedrock platform, ensuring seamless integration into applications and enhancing user engagement and accessibility. The inclusion of MARS6 in AWS Bedrock's diverse selection of foundation models underscores CAMB.AI's commitment to advancing machine learning and artificial intelligence, providing developers with vital tools to create rich audio experiences supported by AWS's reliable and scalable infrastructure. -
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VibeTTS
code01 studio LLC
VibeTTS offers unrivaled 7,000+ language support and phoneme-level control over pitch, energy, and duration. Clone voices from a single sample, edit with a visual editor, preview in real-time, and access multiple specialized TTS models. Ideal for creators, businesses, and developers needing high-quality, commercial-ready audio with API and offline capabilities.Starting Price: $10/month -
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Inworld TTS
Inworld
Inworld TTS is a state-of-the-art text-to-speech platform designed to deliver ultra-realistic, context-aware speech synthesis and precise voice-cloning capabilities at a radically accessible price. The flagship model, TTS-1, is optimized for real-time applications and supports low-latency streaming (first audio chunk in ≈200 ms) as well as multiple languages (including English, Spanish, French, Korean, Chinese, and more). Developers can use instant zero-shot voice cloning (5-15 seconds of audio) or professional fine-tuned cloning, add voice-tags for emotion, style, and non-verbal sounds, and switch languages while preserving voice identity. The larger TTS-1-Max model (in preview) offers even more expressive speech and multilingual strength. The platform supports both API and portal access, streaming or batch mode, and is designed for everything from interactive voice agents and gaming characters to branded audio experiences.Starting Price: $0.005 per minute -
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Voxtral TTS
Mistral AI
Voxtral TTS is a state-of-the-art, multilingual text-to-speech model designed to generate highly realistic and emotionally expressive speech from text, combining strong contextual understanding with advanced speaker modeling to produce natural, human-like audio output. Built as a lightweight model with around 4 billion parameters, it delivers efficient performance while maintaining high quality, enabling scalable deployment for enterprise voice applications. It supports nine major languages and diverse dialects, and can adapt to new voices using only a short reference audio sample, capturing not just tone but also rhythm, pauses, intonation, and emotional nuance. Its zero-shot voice cloning capabilities allow it to replicate a speaker’s style without additional training, and it can even perform cross-lingual voice adaptation, generating speech in one language while preserving the accent of another. -
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KugelAudio
KugelAudio
KugelAudio is the most realistic speech AI platform, combining text-to-speech, speech-to-text, and voice-to-voice in one stack. With 39-50ms inference latency (lowest on the market), 30-second voice cloning, on-premises deployment, and industry-leading accuracy on email addresses, IBANs, and phone numbers, it's built for production voice applications where quality and compliance matter. It's a strong fit for voice bots and conversational agents that need to handle structured data without misreads, real-time applications requiring sub-50ms latency, and regulated industries like banking, insurance, healthcare, and the public sector that need on-premises or EU-sovereign deployment. Beyond enterprise voice automation, KugelAudio also powers branded voice experiences through natural cloning from 30 seconds of audio, multilingual products across over 30 languages German, English, French, and Italian, and media or content production needing the most realistic synthetic voices available.Starting Price: $1 -
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MiniMax Speech 2.8
MiniMax
MiniMax Speech 2.8 is a next-generation AI speech model built to make synthetic voice feel alive, expressive, and deeply human. It focuses on performance in real-world voice agent scenarios, combining ultra-fast response, richer emotional expression, cleaner audio, and stronger cross-lingual performance for products that need natural spoken interaction. Speech 2.8 is designed to reduce the distance between AI voice and real human communication, giving developers and creators more control over how a voice sounds, reacts, and carries meaning. It supports flexible emotion control, allowing users to shape delivery with moods, tone, and expressive direction instead of relying on flat or robotic speech. It can produce speech with more natural pauses, cadence, emphasis, and emotional texture, helping AI characters, assistants, narrators, and interactive agents sound more believable across longer conversations. -
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Grok Text to Speech (TTS)
SpaceXAI
Grok Text to Speech (TTS) is a standalone audio API built to help developers generate fast, natural, and expressive speech from text. Built on the same stack that powers Grok Voice, Tesla vehicles, and Starlink customer support, the API makes it straightforward to integrate high-quality voice generation into applications such as voice agents, accessibility tools, podcasts, assistants, customer experiences, and interactive audio products. Grok TTS can turn long-form text into speech through a REST API or generate speech in real time through a WebSocket API, giving developers flexibility for both batch audio generation and live conversational experiences. It is designed around expressive delivery, not just flat narration, with fine-grained control through simple inline and wrapping speech tags. Developers can add natural prosody and emotion using tags, allowing lifelike delivery without complex markup. -
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Grok Speech to Text (STT)
SpaceXAI
Grok Speech to Text is a standalone audio API built to help developers integrate fast, accurate transcription into any application. Built on the same stack that powers Grok Voice, Tesla vehicles, and Starlink customer support, the API is designed for use cases such as voice agents, real-time transcription tools, accessibility solutions, podcasts, meeting capture, telephony, and interactive audio experiences. Grok STT can generate transcripts from large audio files through a REST API or transcribe speech in real time through a low-latency WebSocket API. It includes word-level timestamps, speaker diarization, multichannel support, and intelligent Inverse Text Normalization that converts spoken language into properly formatted structured output for numbers, dates, currencies, and more. Grok Speech to Text is evaluated across phone calls, meetings, video and podcast content, and telephony, with strong performance in entity recognition and business use cases. -
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Qwen-Audio-3.0-TTS-Plus
Alibaba
Qwen-Audio-3.0-TTS-Plus is the high-quality variant of Qwen-Audio-3.0-TTS, optimized for naturalness and timbre fidelity when output quality matters more than speed. It supports 16 languages, plus improved fidelity for several Chinese dialects. The model delivers strong multilingual intelligibility and ranks first in speaker similarity across all supported languages, helping cloned voices remain recognizable and consistent across linguistic contexts. Developers can direct delivery through ordinary natural-language instructions instead of manually tuning acoustic parameters, controlling emotion, role, scenario, pacing, projection, and tone with simple prompts. Inline tags provide fine-grained control over breaths, laughter, emotional shifts, and other non-verbal details, making the model useful for narration, games, character dialogue, and dubbing. -
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Qwen-Audio-3.0-TTS-Flash
Alibaba
Qwen-Audio-3.0-TTS-Flash is the real-time variant of Qwen-Audio-3.0-TTS, tuned for interactive applications with first-packet latency at the 300 ms level. It supports 16 languages, along with improved fidelity for several Chinese dialects. Across multilingual evaluations, Flash delivers the lowest average WER/CER in the family at 3.87, showing strong intelligibility while preserving speaker identity across diverse languages. Developers can guide delivery with plain-language instructions instead of manually adjusting acoustic parameters, controlling emotion, role, scenario, pace, projection, and tone through simple prompts. Inline tags add precise non-verbal details, making the model well-suited to conversational agents, narration, games, dubbing, and other expressive speech experiences. Voice cloning is designed to work with imperfect reference audio; targeted acoustic simulation suppresses noise and reverberation while retaining the original speaker’s timbre. -
22
MiniMax
MiniMax AI
MiniMax is a global AI technology company that develops advanced multimodal foundation models and AI-powered products for individuals, developers, and enterprises. Its flagship model, MiniMax M3, combines frontier-level coding capabilities, agentic task execution, native multimodal understanding, and support for up to 1 million tokens of context through its proprietary MiniMax Sparse Attention (MSA) architecture. The company offers a comprehensive ecosystem that includes coding assistants, AI agents, video generation, speech synthesis, music generation, and developer APIs. Through products such as MiniMax Code, Hailuo AI, MiniMax Audio, Talkie, and its enterprise platform, users can automate workflows, generate content, build applications, and deploy AI-powered solutions at scale. MiniMax helps organizations and developers improve productivity, accelerate software development, and create intelligent experiences across text, audio, image, video, and music. -
23
Gemini 2.5 Flash TTS
Google
Gemini 2.5 Flash TTS is the latest text-to-speech (TTS) model variant in Google’s Gemini 2.5 lineup, designed for faster, low-latency speech synthesis with expressive, controllable audio output. It offers significant enhancements in tone versatility and expressivity so that developers can generate speech that better matches style prompts, from storytelling narrations to character voices, with more natural emotional range. It features precision pacing, which allows it to adjust speech tempo based on context, delivering faster sections or slowing for emphasis more accurately according to instructions. It also supports multi-speaker dialogues with consistent character voices for scenarios like podcasts, interviews, or conversational agents, and improved multilingual handling so each speaker’s unique tone and style persist across languages. Gemini 2.5 Flash TTS is optimized for lower latency, making it ideal for interactive applications and real-time voice interfaces. -
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Gemini 2.5 Pro TTS
Google
Gemini 2.5 Pro TTS is Google’s advanced text-to-speech model in the Gemini 2.5 family, optimized for high-quality, expressive, controllable speech synthesis for structured and professional audio generation tasks. The model delivers natural-sounding voice output with enhanced expressivity, tone control, pacing, and pronunciation fidelity, enabling developers to dictate style, accent, rhythm, and emotional nuance through text-based prompts, making it suitable for applications like podcasts, audiobooks, customer assistance, tutorials, and multimedia narration that require premium audio output. It supports both single-speaker and multi-speaker audio, allowing distinct voices and conversational flows in the same output, and can synthesize speech across multiple languages with consistent style adherence. Compared with lower-latency variants like Flash TTS, the Pro TTS model prioritizes sound quality, depth of expression, and nuanced control. -
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Google has released updated Gemini audio models that significantly expand the platform’s capabilities for natural, expressive voice interactions and real-time conversational AI with the introduction of Gemini 2.5 Flash Native Audio and improved text-to-speech technology. The updated native audio model powers live voice agents that can handle complex workflows, follow detailed user instructions more reliably, and maintain smoother multi-turn conversations by better recalling context from previous turns. It is now available across Google AI Studio,Gemini Enterprise Agent Platform, Gemini Live, and Search Live, enabling developers and products to build interactive voice experiences such as intelligent assistants and enterprise voice agents. In addition to the real-time voice improvements, Google enhanced the underlying Text-to-Speech (TTS) models in the Gemini 2.5 family to offer greater expressivity, tone control, pacing adjustments, and multilingual support.
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Gemini 3.1 Flash TTS
Google
Gemini 3.1 Flash TTS is Google’s latest text-to-speech model designed to deliver highly expressive, controllable, and scalable AI-generated speech for developers and enterprises. Available in Google AI Studio and Gemini Enterprise Agent Platform, it focuses on precise control over how audio is generated, allowing users to shape delivery through natural language prompts and an extensive system of more than 200 audio tags that define pacing, tone, emotion, and style. It supports over 70 languages and regional variants, along with a library of 30 prebuilt voices, enabling users to generate speech ranging from professional narration to conversational or stylized performances. Developers can embed instructions directly into text inputs to guide vocal expression, combining pacing, emotion, and pauses in a structured prompting framework that produces nuanced, high-fidelity audio output. Gemini 3.1 Flash TTS is optimized for real-world applications. -
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MAI-Voice-2
Microsoft AI
MAI-Voice-2 is Microsoft AI’s most expressive and natural-sounding text-to-speech model to date, built for production voice experiences where fidelity, language coverage, speaker consistency, and emotional range directly shape the user experience. It is designed for assistants, customer support, audiobooks, accessibility experiences, games, podcasts, courses, simulations, and creator workflows where voice quality must sound natural, fluid, and trustworthy. It expands from English-only support to 15 languages while maintaining naturalness and expressiveness, with support for English, Italian, French, German, Hindi, Spanish, Portuguese, Korean, Chinese, Turkish, Russian, Thai, Dutch, Romanian, and Hungarian. MAI-Voice-2 offers granular emotion control through tags such as sad, whispered, and excited, along with role-based expressive speech for experiences like motivational trainers, sports commentators, or character voices. -
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Miso TTS
Miso TTS
Miso Labs builds emotive foundation models for voice, designed to help developers create voice agents that feel fast, warm, and human instead of robotic or delayed. Its flagship model, Miso TTS, is an 8-billion-parameter transformer model for state-of-the-art emotive speech and dialogue generation, with open source weights available on Hugging Face and API access coming soon. Miso is built for real-time conversational voice, responding in 110ms to preserve natural flow and avoid the awkward pauses common in AI voice agents. It supports one-shot voice cloning, allowing users to clone a voice from a ten-second audio clip while keeping the agent’s voice consistent from the first second of a call to the last. Miso Labs also emphasizes local and sovereign deployment, with open source models built for local use and on-premises hosting and support available for enterprise teams that need to keep sensitive data in-house. -
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Cartesia Sonic-3.5
Cartesia
Sonic 3.5 is Cartesia’s fastest, most natural text-to-speech model, built for expressive, real-time voice generation with sub-90ms latency and native support for 42 languages. It is designed to follow transcripts faithfully, voice confirmation codes, and heteronyms correctly without preprocessing, and stay expressive enough to carry a real conversation. It supports languages intended to deliver native-quality speech. Sonic 3.5 focuses on clean audio across every language and voice, with no artifacts to edit out, making it practical for production voice experiences where quality, speed, and consistency matter. Its expressive conversational delivery provides strong pacing and real emotional range, tuned for support and agent transcripts. Alphanumerics such as order numbers, phone numbers, IDs, and emails are spoken naturally in every language, while context-aware English pronunciation helps words like read, bass, and bow land correctly from the surrounding text. -
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Cartesia Ink 2
Cartesia
Ink 2 is Cartesia’s fastest, most accurate streaming speech-to-text model, built for production voice agents with the lowest word error rate and best turn detection of any streaming STT. It is designed to transcribe structured data such as phone numbers, dates, and emails correctly the first time, while also knowing when a speaker starts and finishes without requiring a separate voice activity detection system. Turn detection is built directly into the model, so voice agents can react to events instead of managing raw transcript segments. Ink 2 emits a full lifecycle of turn events, giving an agent clear signals for when to listen, interrupt, think, prepare a reply, cancel a premature response, or speak. The transcript property is cumulative within a turn, meaning each update contains the full text transcribed so far rather than a delta, and emitted text is final once sent.
