Alternatives to Hecttor

Compare Hecttor alternatives for your business or organization using the curated list below. SourceForge ranks the best alternatives to Hecttor in 2026. Compare features, ratings, user reviews, pricing, and more from Hecttor competitors and alternatives in order to make an informed decision for your business.

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    Google Cloud Speech-to-Text
    Google Cloud’s Speech API processes more than 1 billion voice minutes per month with close to human levels of understanding for many commonly spoken languages. Powered by the best of Google's AI research and technology, Google Cloud's Speech-to-Text API helps you accurately transcribe speech into text in 73 languages and 137 different local variants. Leverage Google’s most advanced deep learning neural network algorithms for automatic speech recognition (ASR) and deploy ASR wherever you need it, whether in the cloud with the API, on-premises with Speech-to-Text On-Prem, or locally on any device with Speech On-Device.
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  • 2
    Speechmatics

    Speechmatics

    Speechmatics

    Best-in-Market Speech-to-Text & Voice AI for Enterprises. Speechmatics delivers industry-leading Speech-to-Text and Voice AI for enterprises needing unrivaled accuracy, security, and flexibility. Our enterprise-grade APIs provide real-time and batch transcription with exceptional precision—across the widest range of languages, dialects, and accents. Powered by Foundational Speech Technology, Speechmatics supports mission-critical voice applications in media, contact centers, finance, healthcare, and more. With on-prem, cloud, and hybrid deployment, businesses maintain full control over data security while unlocking voice insights. Trusted by global leaders, Speechmatics is the top choice for best-in-class transcription and voice intelligence. 🔹 Unmatched Accuracy – Superior transcription across languages & accents 🔹 Flexible Deployment – Cloud, on-prem, and hybrid 🔹 Enterprise-Grade Security – Full data control 🔹 Real-Time & Batch Processing – Scalable transcription
    Starting Price: $0 per month
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    Telnyx

    Telnyx

    Telnyx

    Telnyx is a global communications infrastructure platform that provides voice, messaging, networking, and AI-powered real-time communication capabilities through a fully owned telecom stack. The platform combines carrier-grade networking, programmable identity systems, AI inference, and low-latency communication infrastructure to support real-time conversational AI agents and enterprise communication workflows. Telnyx owns and operates its entire network stack, including physical infrastructure, mobile core systems, edge processing, and AI compute layers, enabling faster performance and lower latency without relying on third-party telecom providers. The platform offers tools such as voice agent builders, speech-to-text, text-to-speech, global phone numbers, AI orchestration, and programmable compliance controls for building intelligent voice and messaging systems.
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    Rev

    Rev

    Rev

    Rev provides premium on-demand, manual and automated transcription, closed caption, and foreign subtitling services. With 170,000+ customers, Rev's clients span from global enterprises to freelance journalists. Rev processes more audio and video than any other provider and has the ability to scale to fit any customer's needs. Pricing is simple starting at just $0.25 per audio/video minute for automated speech-to-text services and $1.25/min for manual with 99% accuracy. Rev also offers Rev.ai which is a speech recognition engine that's available to companies that want it.
    Starting Price: $1.25 per minute
  • 5
    Gemini Audio
    Gemini Audio is a set of advanced real-time audio models built on Gemini's architecture, designed to enable natural, fluid voice interaction and expressive audio generation through simple language prompts. It supports conversational experiences where users can speak, listen, and interact with AI in a seamless loop, combining understanding, reasoning, and response generation in audio form. It is capable of both analyzing and generating audio, allowing applications such as speech-to-text transcription, translation, speaker identification, emotion detection, and detailed audio content analysis. They are optimized for low-latency, real-time use cases, making them suitable for live assistants, voice agents, and interactive systems that require continuous, multi-turn dialogue. Gemini Audio also integrates advanced capabilities like function calling, enabling the model to trigger external tools and incorporate real-time data into responses.
    Starting Price: Free
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    Amazon Nova Sonic
    ​Amazon Nova Sonic is a state-of-the-art speech-to-speech model that delivers real-time, human-like voice conversations with industry-leading price performance. It unifies speech understanding and generation into a single model, enabling developers to create natural, expressive conversational AI experiences with low latency. Nova Sonic adapts its responses based on the prosody of input speech, such as pace and timbre, resulting in more natural dialogue. It supports function calling and agentic workflows to interact with external services and APIs, including knowledge grounding with enterprise data using Retrieval-Augmented Generation (RAG). It provides robust speech understanding for American and British English across various speaking styles and acoustic conditions, with additional languages coming soon. Nova Sonic handles user interruptions gracefully without dropping conversational context and is robust to background noise.
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    Gladia

    Gladia

    Gladia

    Gladia is a speech-to-text platform built for production, turning raw audio into structured outputs that power real workflows like meeting summaries, CRM enrichment, contact center QA, and real-time voice assistants. With support for 99+ languages and the ability to handle messy real-world audio—overlapping speakers, accents, code-switching, domain-specific terminology—Gladia is designed for the complexity of actual conversations, not clean studio recordings.
    Starting Price: 10 hours free
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    Gemini 2.5 Flash TTS
    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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    aiOla

    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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    Grok Speech to Text (STT)
    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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    Ctalk

    Ctalk

    Ctalk

    Realize the benefits of contact center, IVR, speech recognition, call recording, unified communications, outbound dialing without replacing your existing telephony platform. The Ctalk contact centre system 'wraps around' your existing PBX seamlessly adding features or more capacity. That's why you don't have to rip and replace. Effectively handle more calls and contacts with the same or reduced resources. Significantly reduce your support costs and dependency on I.T. by empowering multiple administrators with on the fly call management. Dramatically Increase first contact resolution. Know who is calling and why, then route to the right agent every time. 24/7 automated services blend seamlessly with proactive outbound calling.
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    RapportCMS
    RapportCMS is our competitive advantage vs our competitors. We are focused on the intersection between telephony, interaction management and the people who handle the calls. This approach ensures that we make ‘human technology’ designed by and for contact center practitioners. We know that world-class call center technology must be equally adept at addressing what happens after the agent says hello as to how the call is routed to the desktop. As one of the leading Contact Centres in the AUNZ market, we had over 10 years of building, refining and improving our technology before then releasing it to market as a SAAS solution. While most providers have built solutions from a telephony perspective, we recognize what happens after the agent says hello is of equal importance to what happened before.
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    Knovvu Speech Recognition
    Automate customer processes, evaluate agent performances objectively and ensure your operations are 100% efficient. In our connected world, many consumers are interacting with everyday connected appliances in new ways. With a trend in connected devices that often lack a screen, speech is emerging as a natural, intuitive interface for human-machine interaction. Speech recognition is the driving technology behind this development, revolutionizing the way people interact with their devices. With Knovvu Speech Recognition from Sestek, machines and applications can understand user commands in spoken language. With the ability to listen to and interpret spoken demands, users may interact with these devices by speaking aloud rather than inputting buttons and keystrokes. Our automatic speech recognition software has full application. Many organizations use technology to power intuitive and straightforward self-service solutions.
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    Alibaba Cloud Intelligent Speech Interaction
    Intelligent Speech Interaction is developed based on state-of-the-art technologies such as speech recognition, speech synthesis, and natural language understanding. Enterprises can integrate Intelligent Speech Interaction into their products to enable them to listen, understand, and converse with users, providing users with an immersive human-computer interaction experience. Intelligent Speech Interaction is currently available in Mandarin Chinese, Cantonese Chinese, English, Japanese, Korean, French and Indonesian, and please stay tuned for other languages. Intelligent Speech Interaction is suitable for various scenarios, including intelligent Q&A, intelligent quality inspection, real-time subtitling for speeches, and transcription of audio recordings. Intelligent Speech Interaction has been successfully applied in many industries such as finance, insurance, eCommerce and smart home.
    Starting Price: $1.40 per hour
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    Picovoice

    Picovoice

    Picovoice

    Picovoice is the first and only ubiquitous on-device voice AI platform. Picovoice offers speech-to-text, voice search, wake word, Speech-to-Intent (intent detection) and voice activity detection engines. Its stack can run on anything from embedded devices to web browsers, providing an immersive experience not achievable by any Big Tech.
    Starting Price: Free
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    Soniox

    Soniox

    Soniox

    Soniox develops highly accurate foundational speech models that transcribe, translate, and understand speech as it happens, and also provides the developer platform that makes it easy to integrate real-time voice intelligence into any application. Soniox Speech-to-Text API allows you to transcribe speech in 60+ languages in real-time with high accuracy - built for large scale. Soniox also provides regional data residency and is SOC 2 Type 2, GDPR and HIPAA compliant.
    Starting Price: $0.10/hour of audio
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    MAI-Transcribe-1

    MAI-Transcribe-1

    Microsoft AI

    MAI-Transcribe-1 is a state-of-the-art speech-to-text model developed by Microsoft and available through Azure AI Foundry, designed to deliver high-accuracy transcription for real-world audio across enterprise and developer use cases. It supports 25 major languages and is optimized to handle diverse accents, dialects, and speaking styles, maintaining consistent performance even in challenging conditions such as background noise, low-quality recordings, or overlapping speech. It is built by Microsoft’s AI Superintelligence team with a dual focus on accuracy and efficiency, enabling fast batch transcription and scalable deployment for production environments. MAI-Transcribe-1 powers a wide range of applications, including meeting transcription, live captions, accessibility tools, call center analytics, and voice-driven agents, making it a foundational component for voice-enabled systems.
    Starting Price: Free
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    Azure AI Speech
    Build voice-enabled apps confidently and quickly with the Speech SDK. Transcribe speech to text with high accuracy, produce natural-sounding text-to-speech voices, translate spoken audio, and use speaker recognition during conversations. Create custom models tailored to your app with Speech studio. Get state-of-the-art speech to text, lifelike text to speech, and award-winning speaker recognition. Your data stays yours, your speech input is not logged during processing. Create custom voices, add specific words to your base vocabulary, or build your own models. Run Speech anywhere, in the cloud or at the edge in containers. Quickly and accurately transcribe audio in more than 92 languages and variants. Gain customer insights with call center transcription, improve experiences with voice-enabled assistants, capture key discussions in meetings and more. Use text to speech to create apps and services that speak conversationally, choosing from more than 215 voices, and 60 languages.
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    Inworld TTS
    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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    GoVivace

    GoVivace

    GoVivace

    Our automatic speech recognition engine supports several English accents and can be localized to any language. Also, the ASR engine supports standard telephony as well as web and mobile applications. Being capable of actioning voice commands given to electronic devices such as computers, tablets, smartphones or telephones with the aid of a microphone, the GoVivace’s Automatic Speech Recognition Engine finds use in diverse applications. This automatic speech recognition engine compares the spoken input with a number of pre-specified possibilities and convert speech to text. The entire set of pre-specified possibilities constitute the application’s grammar, which powers the interface between the dialogue-speaker and the back-end processing. GoVivace’s patented Automatic Speech Recognition solution needs only very simple grammar for its processing. It can also support very large grammars for complex tasks.
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    SpeechPulse
    SpeechPulse uses your computer’s microphone for real-time speech recognition. It can type into your favorite apps, including text editors, web browsers, and office applications. SpeechPulse works fully offline and doesn’t require any internet connectivity. It supports speech recognition in multiple languages, including English, French, Spanish, Italian, German, Japanese, Chinese, and Russian (a total of 100 languages). SpeechPulse supports both auto punctuation and manual punctuation for the English language. It supports auto punctuation for all other languages. SpeechPulse can also generate subtitles for your audio and video files with accurate timestamps. It supports SRT and VTT subtitle formats. You can also customize the width of a subtitle line to include only a limited number of characters. SpeechPulse has a one-time payment. You can pay for the product once and use it forever.
    Starting Price: $59.95/one-time payment
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    GPT‑Realtime‑Whisper
    GPT-Realtime-Whisper is OpenAI’s streaming transcription model built for low-latency speech-to-text experiences in live products. It transcribes audio as people speak, helping voice-enabled apps feel faster, more responsive, and more natural, from captions that appear in the moment to meeting notes that keep up with the conversation. It makes live speech usable inside business workflows as it happens, so teams can power captions for meetings, classrooms, broadcasts, and events, generate notes and summaries while conversations are still in progress, build voice agents that need to understand users continuously, and create faster follow-up workflows for high-volume spoken interactions. It is part of a new generation of real-time voice models in the API that can reason, translate, and transcribe as people speak, moving real-time audio beyond simple call-and-response toward voice interfaces that can listen, translate, transcribe, and take action as a conversation unfolds.
    Starting Price: $0.017 per minute
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    FonadaLabs

    FonadaLabs

    FonadaLabs

    FonadaLabs is a voice AI platform that provides enterprise-grade infrastructure and APIs for building voice agents on Indian telephony networks. The platform offers a complete voice pipeline that includes telephony hosting, noise cancellation, speech recognition, voice models, and text-to-speech capabilities within a unified API environment. FonadaLabs supports over 23 Indian languages with speech recognition optimized for regional accents and telephony use cases. The platform enables real-time voice streaming with ultra-low latency, enterprise security, and India-based data residency for compliance and sovereignty requirements. Businesses can also leverage specialized voice agent language models, tool-calling support, and natural-sounding Indian voice generation for customer interactions and automation.
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    Azure Voice Live API
    Azure Voice Live API is a fully managed solution for building low-latency, high-quality speech-to-speech agents through one unified interface. It combines speech recognition, generative AI, and text-to-speech, allowing developers to send audio input and receive audio output, synchronized avatar visuals, and action triggers without manually orchestrating separate backend components or deploying the underlying models. It supports more than 140 speech-to-text locales and over 600 standard voices across 150+ text-to-speech locales, with options for phrase lists, custom speech, custom voices, and brand-aligned avatars. Developers can choose among multiple generative AI models, including GPT-Realtime, GPT-5, GPT-4.1, GPT-4o, Phi, and compatible bring-your-own models, depending on the intelligence, speed, and latency required. Advanced conversational features include noise suppression, echo cancellation, robust interruption detection, and end-of-turn detection.
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    Yactraq

    Yactraq

    Yactraq

    Yactraq is the industry value leader in speech analytics software. Our customers typically realize benefits across two broad functional areas. Marketing teams looking to extend their Voice-of-the-Customer (VoC) capabilities beyond the feedback form and social media now want to mine sales and customer service phone calls as part of their omni-channel capability. Contact Center Quality Management teams typically use speech analytics / audio mining as a way of leveraging AI / Machine Learning to evaluate the performance of their call agents. Yactraq offers customized free trials based on a clients own data so they can experience the value of our software before deciding to buy. Our products are cost-effectively priced to suit the needs of end customers as well as partners in the Business Process Outsourcing (BPO), Contact Center as a Service (CCAS), Voice-of-the-Customer (VoC), CRM Software and Network Service Provider businesses.
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    SpeechText.AI

    SpeechText.AI

    SpeechText.AI

    Transcribe audio and video into text. Get accurate transcriptions of podcasts with domain-specific speech recognition. SpeechText.AI is a powerful artificial intelligence software for speech to text conversion and audio transcription. Upload audio or video files. AI transcription software supports various file formats and transcribes from speech to text in any language. Select domain. Select industry domain and audio type from predefined categories to improve the recognition accuracy of domain-specific words. Transcribe. Our speech transcription engine uses state-of-the-art deep neural network models to convert from audio to text with close to human accuracy. Edit & Export. Search, modify and verify audio transcriptions using interactive editing tools. Export your content in different formats. Why SpeechText.AI? Set of amazing features to help you transcribe audio and video in seconds. Speech recognition. Powerful speech-to-text tech.
    Starting Price: $19 one-time payment
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    OpenAI Realtime API
    The OpenAI Realtime API is a newly introduced API, announced in 2024, that allows developers to create applications that facilitate real-time, low-latency interactions, such as speech-to-speech conversations. This API is designed for use cases like customer support agents, AI voice assistants, and language learning apps. Unlike previous implementations that required multiple models for speech recognition and text-to-speech conversion, the Realtime API handles these processes seamlessly in one call, enabling applications to handle voice interactions much faster and with more natural flow.
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    Vision Agents
    Vision Agents is an open source Python framework for building low-latency voice and video AI agents with any model. It lets developers plug in LLM, speech, and vision models from more than 25 providers and ship real-time agents for telehealth, voice support, live coaching, video analysis, interactive avatars, security monitoring, sports commentary, and other multimodal applications. It is designed to help teams build agents that can listen, speak, see, process media, call tools, and respond in real time while running on Stream’s global edge network with sub-500ms latency. Developers can build a first agent in minutes, using a small Python setup with Gemini Realtime, OpenAI, Deepgram, ElevenLabs, Stream, or other supported providers. Vision Agents supports both real-time speech-to-speech models and custom STT/LLM/TTS pipelines, giving teams either the fastest path to a working voice agent or full control over speech recognition, language reasoning, text-to-speech, etc.
    Starting Price: Free
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    Gemini 3.1 Flash TTS
    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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    Solventum Fluency Direct
    Solventum Fluency Direct is an AI-powered speech recognition and conversational documentation solution designed to help clinicians create clinical notes directly within electronic health record (EHR) systems. The platform uses advanced speech understanding and natural language processing technologies to allow physicians to dictate, review, edit, and sign medical documentation through natural conversation. By reducing manual typing and administrative workload, Solventum Fluency Direct helps clinicians document patient encounters more efficiently while improving documentation accuracy. Integrated computer-assisted physician documentation (CAPD) functionality provides real-time insights and prompts that help physicians capture a complete and compliant patient story during the documentation process.
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    Speech Recognition Cloud

    Speech Recognition Cloud

    Speech Recognition Cloud

    Speech Recognition Cloud is a cloud-based speech recognition and dictation application for Windows. It converts speech to text in real time and types directly at the cursor in most applications (Word, Outlook, browsers and web forms). It supports automatic punctuation, spoken formatting commands (new lines, paragraphs, bullet and numbered lists), configurable hotkeys/hold-to-talk, and custom vocabulary and text expansion. Processing occurs in the cloud, so users can dictate on standard PCs without high-end hardware. An optional Medical edition supports clinical terminology for healthcare documentation. An internet connection is required.
    Starting Price: $6/month
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    wolkvox

    wolkvox

    Microsyslabs

    wolkvox is a cloud-based call center management software that helps businesses streamline communications across numerous web chat applications and social media channels such as Telegram, WhatsApp, Line, Twitter, Facebook, and Instagram. Organizations can manage interactions using video calls, landline, mobile devices, SMS, email and more. wolkvox enables enterprises to create and monitor multiple customer categories, record and analyze client interactions and generate reports to track the performance of campaigns and agents. It offers a variety of features including a drag-and-drop interface, simultaneous calling, Artificial Intelligence (AI)-enabled speech analytics, gamification, and more. Additionally, administrators can use the predictive dialer to establish custom rules for virtual agents, call routing and messages and design templates for email and SMS campaigns. wolkvox supports integration with various third-party ERP, business intelligence, CRM, and information systems.
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    Cartesia Sonic-3
    Cartesia Sonic-3 is a real-time, streaming text-to-speech (TTS) model designed to generate ultra-realistic, expressive voice output with extremely low latency, enabling AI systems to speak as fluidly as humans in live interactions. Built on advanced state space model architecture, Sonic delivers high-quality speech while achieving near-instant response times, with audio generation beginning in as little as 40–100 milliseconds, making conversations feel seamless rather than delayed. It is optimized for conversational AI use cases, acting as the “voice layer” for AI agents by converting text into natural-sounding speech that includes emotional nuance such as excitement, empathy, or even laughter. It supports more than 40 languages with native-level voices and accent localization, allowing developers to build globally accessible applications with consistent quality across regions.
    Starting Price: $4 per month
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    NeoSound

    NeoSound

    NeoSound Intelligence

    NeoSound Intelligence is an AI tech company that turns emotions into actionable insights in order to create a world with better conversations between organizations and consumers. ​We intend to make all conversations better between consumers and organizations. By providing AI-powered speech analytics tools, we help call center companies to optimize their customer communication. Turn calls into revenue. Optimise customer communication by listening to customer calls automatically. NeoSound tools turn phone conversations into meaningful actionable insights to make customer communication better. NeoSound tools do not only speech-to-text translation. Smart algorithms do acoustics and intonation analysis. The machine listens to how people speak not only what they say. That is why our trained machines can easily address your company-specific needs. NeoSound offers a unique combination of speech-to-text semantic analytics and acoustic analysis of intonation.
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    Gemini 2.5 Flash Native Audio
    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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    AppTek

    AppTek

    AppTek

    AppTek is a global leader in artificial intelligence (AI) and machine learning (ML) technologies for automatic speech recognition (ASR), neural machine translation (NMT), and natural language understanding (NLU). The AppTek platform delivers industry-leading, real-time streaming and batch technology solutions in the cloud or on-premise for organizations across a breadth of worldwide markets such as media and entertainment, call centers, government, enterprise business, and more. Built by scientists and research engineers who are recognized among the best in the world, AppTek’s solutions cover a wide array of languages, dialects, and channels. AppTek utilizes deep neural networks to transcribe and understand speech and text data, delivering more accurate and efficient tools.
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    Transcribe

    Transcribe

    Wreally

    Transcribe saves thousands of hours every month in transcription time for journalists, lawyers, podcasters, students and professional transcriptionists all over the world. Increase your productivity & save mountains of time when converting your interviews, audio notes, lectures, speeches, podcasts and any recorded speech to text. Put on your headphones, load your audio, slow it down and speak out what you hear. It's that simple. Our dictation engine will convert your speech to text on the fly. This is way faster than typing. We support English, Spanish, French, Hindi and almost all other European & Asian languages.
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    Diktamen

    Diktamen

    Diktamen

    Diktamen is a cloud-based digital dictation and transcription platform designed to streamline voice capture, task management, and workflow automation across professional sectors. The solution enables users to dictate audio from any location, via mobile, desktop, or dedicated devices, and securely transmit that audio for transcription, speech recognition, and task assignment. It supports industry-specific workflows (notably in legal and healthcare), allows integration with existing systems, and features centralized management for submissions, status tracking, and BI reporting with AI-driven forecasting. Clients benefit from cost reduction in dictation infrastructure, efficient transcription turnaround through outsourced partner networks, real-time task routing, and a flexible SaaS deployment model with minimal local installation or maintenance. Diktamen holds ISO 27001 certification and adheres to GDPR for data security and compliance.
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    RocketWhisper

    RocketWhisper

    Mojosoft Co., Ltd.

    RocketWhisper is a powerful desktop speech recognition and transcription application that runs 100% offline on your computer. Your voice data never leaves your machine - complete privacy guaranteed. Powered by OpenAI's Whisper engine with NVIDIA GPU (CUDA) acceleration, RocketWhisper delivers fast and accurate speech-to-text conversion for professionals, content creators, and anyone who works with voice and text. Key Features: - 100% offline processing - voice data never leaves your PC - OpenAI Whisper engine for high-accuracy speech recognition - NVIDIA CUDA GPU acceleration - up to 10x faster than CPU - Real-time voice-to-text input with global hotkey (Push-to-Talk with Right Alt) - Batch transcription of multiple audio/video files (MP3, WAV, M4A, MP4, MKV, AVI, etc.) - SRT/VTT subtitle export for video content - AI text formatting with LLM integration (OpenAI, Anthropic, Google Gemini, Grok, local LLM)
    Starting Price: $32 one-time
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    Gemini 3.1 Flash Live
    Gemini 3.1 Flash Live is Google’s most advanced real-time audio model, designed to deliver natural, reliable, and low-latency voice interactions for the next generation of conversational AI. It is optimized for real-time dialogue, enabling fluid, human-like conversations with improved precision, faster response times, and a more natural rhythm that better reflects how people actually speak. It enhances tonal understanding, allowing it to recognize nuances such as pitch, pace, and emotional cues, and dynamically adapt responses to user intent, including frustration or confusion. Built for both developers and enterprises, it can be accessed through the Gemini Live API in Google AI Studio, as well as integrated into production environments to power voice-first agents capable of handling complex, multi-step tasks at scale. It supports multimodal inputs including text, audio, images, and video, and produces both text and audio outputs, enabling richer, context-aware interactions.
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    MiniMax Speech 2.8
    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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    Yandex SpeechKit
    Speech technologies based on machine learning to create voice assistants, automate call centers, monitor service quality, and perform other tasks. Leverage the advanced technology behind the wildly successful Alice voice assistant, now ready for use in your business. In a fraction of a second, SpeechKit accurately recognizes speech, allowing our clients' voice assistants to communicate quickly and easily. Choose the right version for you, the full version creates a smart voice assistant while the adaptive version gives your brand a unique voice in just a month. A solution for the most demanding customers who need to control speech processing and synthesis within their own infrastructure. SpeechKit’s ML models can now be deployed to your infrastructure. We offer both hybrid options and 100% on-premise deployments for sensitive traffic. The service can recognize audio in MP3, LPCM, and OggOpus formats.
    Starting Price: $0.000020 per unit
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    Cartesia Sonic-3.5
    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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    Rev.ai

    Rev.ai

    Rev.ai

    Rev.ai was built by leading speech recognition experts from millions of hours of accurate human-transcribed content. We began in 2011 with Rev.com, providing human transcription services. We are now the world's largest transcription vendor, with over 35,000 contractors who transcribe millions of minutes of audio each month. In 2017 we launched Temi, an automated speech-to-text transcription and editing service. Temi has already transcribed 20 million minutes of content and was named the best transcription service by Wirecutter. Today our best-in-class speech engine is available to everyone as Rev.ai. We're helping companies get the most out of their audio and video content by making it searchable and accessible.
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    Voci

    Voci

    Medallia

    Companies engage with customers by phone more than any other channel, and these interactions represent a gold mine of untapped information. Listening to every customer call is costly and time-consuming and not physically practical. As a result, only a fraction of randomly selected calls is typically reviewed. These voice interactions reveal the true voice of your customers and enable you to get to the heart of their concerns. With our highly accurate, automated speech-to-text transcription, you can transform your unstructured voice data into transcripts that can be integrated into your analytics platforms. Voci enables you to improve agent quality monitoring, enhance the customer experience, extract competitive intelligence and ensure compliance.
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    AccuSpeechMobile

    AccuSpeechMobile

    AccuSpeechMobile

    AccuSpeechMobile's modern, robust speech recognition is optimized for mobile devices in over 40 languages. Designed for industry workflows, cutting edge noise abatement technology delivers outstanding recognition in noisy environments. A speaker-independent voice engine works for all users out-of-the-box, without the need to voice train or maintain voice files for each user. AccuSpeechMobile is a 100% device-based solution. No voice server or middleware is required and no changes are needed to the backend system (WMS, ERP, EAM, CMMS). Cloud or network connection is not required to use the full functionality of device-based data collection. AccuSpeechMobile fully supports multi-modal capabilities so that users can hear spoken information and speak commands in tandem with the use of intelligent scanners. The ability to reference additional information on the device screen is also always available in conjunction with speech-to-text and text-to-speech commands.
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    Virtual Speech Center

    Virtual Speech Center

    Virtual Speech Center

    Virtual Speech Center offers innovative speech therapy apps and software for schools, private practices, independent speech pathologists and parents. We offer a wide range of mobile applications for speech therapy developed for IPad and IPhone devices. Some of our apps are offered at no charge to speech pathologists. Virtual Speech Center is a pioneer in taking speech and language therapy apps to the next level by incorporating games as reward components. The games featured in our apps include puzzles, board games, and games with sports and carnival themes. Our apps can be purchased individually or in bundles. Virtual Speech Center's TheraPlatform speech therapy software includes telepractice, documentation, billing, intake forms and e-claim submission modules designed for speech and language pathologists. Virtual Speech Center offers innovative speech therapy apps for schools, private practices, independent speech pathologists and parents.
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    OpenAI Whisper
    Whisper is an automatic speech recognition (ASR) system developed by OpenAI for converting spoken language into text. It is trained on 680,000 hours of multilingual and multitask audio data collected from the web. The model is designed to handle diverse accents, background noise, and technical language with high accuracy. Whisper supports transcription in multiple languages as well as translation into English. It uses an encoder-decoder Transformer architecture to process audio inputs and generate text outputs. The system can also perform tasks like language identification and timestamp generation. Overall, Whisper enables developers to build robust voice-enabled applications with ease.
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    Verbatim

    Verbatim

    Saince

    Speech recognition and radiology reporting solution that everyone can afford. Verbatim is the industry’s newest and technically most advanced speech recognition and radiology reporting solution that does not burn a hole in your pocket. With accuracy of 99% and built-in intuitive workflows, you can complete your reports fast and easy.
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    Oxlo.ai

    Oxlo.ai

    Oxlo.ai

    Oxlo.ai is a privacy-first inference stack for agents, built to run frontier-class open-source models with unlimited agentic tool calls, secure failover, and zero data retention or training. It gives developers request-based access to curated open models through a unified HTTP API designed for predictable usage, low-latency inference, and clean integration into production systems. Teams can call models through OpenAI-compatible endpoints, switch from another provider by changing the base URL and API key, and keep support for streaming, function calling, JSON mode, vision models, embeddings, and image generation. Oxlo.ai supports more than 40 models across text, chat, reasoning, coding, image generation, audio, embeddings, computer vision, vision-language, speech-to-text, text-to-speech, long-context, and detection workflows.
    Starting Price: $80 per month