Alternatives to Modulate Velma

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

  • 1
    Dialogflow
    Dialogflow from Google Cloud is a natural language understanding platform that makes it easy to design and integrate a conversational user interface into your mobile app, web application, device, bot, interactive voice response system, and so on. Using Dialogflow, you can provide new and engaging ways for users to interact with your product. Dialogflow can analyze multiple types of input from your customers, including text or audio inputs (like from a phone or voice recording). It can also respond to your customers in a couple of ways, either through text or with synthetic speech. Dialogflow CX and ES provide virtual agent services for chatbots and contact centers. If you have a contact center that employs human agents, you can use Agent Assist to help your human agents. Agent Assist provides real-time suggestions for human agents while they are in conversations with end-user customers.
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    Amazon Lex
    Amazon Lex is a service for building conversational interfaces into any application using voice and text. Amazon Lex provides the advanced deep learning functionalities of automatic speech recognition (ASR) for converting speech to text, and natural language understanding (NLU) to recognize the intent of the text, to enable you to build applications with highly engaging user experiences and lifelike conversational interactions. With Amazon Lex, the same deep learning technologies that power Amazon Alexa are now available to any developer, enabling you to quickly and easily build sophisticated, natural language, conversational bots (“chatbots”). With Amazon Lex, you can build bots to increase contact center productivity, automate simple tasks, and drive operational efficiencies across the enterprise. As a fully managed service, Amazon Lex scales automatically, so you don’t need to worry about managing infrastructure.
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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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    Gemini 2.5 Pro TTS
    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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    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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    Realtime TTS-2
    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
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    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.
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    Amazon Nova 2 Sonic
    Nova 2 Sonic is Amazon’s real-time speech-to-speech model designed to deliver natural, flowing voice interactions without relying on separate systems for text and audio. It combines speech recognition, speech generation, and text processing in a single model, enabling smooth, human-like conversations that can shift effortlessly between voice and text. With expanded multilingual support and expressive voice options, it produces responses that sound more lifelike and contextually aware. Its one-million-token context window allows for long, continuous interactions without losing track of prior details. It supports asynchronous task handling, meaning users can continue speaking, change topics, or ask follow-up questions while background tasks, such as searching for information or completing a request, continue uninterrupted. This makes voice experiences feel more fluid and less bound by traditional turn-based dialog constraints.
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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
  • 10
    Voxtral TTS

    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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    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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    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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    Octave TTS

    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
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    ElevenLabs

    ElevenLabs

    ElevenLabs

    The most realistic and versatile AI speech software, ever. Eleven brings the most compelling, rich and lifelike voices to creators and publishers seeking the ultimate tools for storytelling. Generate top-quality spoken audio in any voice and style with the most advanced and multipurpose AI speech tool out there. Our deep learning model renders human intonation and inflections with unprecedented fidelity and adjusts delivery based on context. Our AI model is built to grasp the logic and emotions behind words. And rather than generate sentences one-by-one, it’s always mindful of how each utterance ties to preceding and succeeding text. This zoomed-out perspective allows it to intonate longer fragments convincingly and with purpose. And finally you can do this with any voice you want.
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    EVI 3

    EVI 3

    Hume AI

    Hume AI's EVI 3 is a third-generation speech-language model that streams in user speech and forms natural, expressive speech and language responses. At conversational latency, it produces the same quality of speech as our text-to-speech model, Octave. Simultaneously, it responds with the same intelligence as the most advanced LLMs of similar latency. It also communicates with reasoning models and web search systems as it speaks, “thinking fast and slow” to match the intelligence of any frontier AI system. EVI 3 can instantly generate new voices and personalities instead of being limited to a handful of speakers. For instance, users can speak to any of the more than 100,000 custom voices already created on our text-to-speech platform, each with an inferred personality. No matter the voice, it responds with a wide range of emotions or styles, implicitly or on command.
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    gpt-realtime
    GPT-Realtime is OpenAI’s most advanced, production-ready speech-to-speech model, now accessible through the fully available Realtime API. It delivers remarkably natural, expressive audio with fine-grained control over tone, pace, and accent. The model can comprehend nuanced human audio, including laughter, switch languages mid-sentence, and accurately process alphanumeric details like phone numbers across multiple languages. It significantly improves reasoning and instruction-following (achieving 82.8% on the BigBench Audio benchmark and 30.5% on MultiChallenge) and boasts enhanced function calling, now more reliable, timely, and accurate (scoring 66.5% on ComplexFuncBench). The model supports asynchronous tool invocation so conversations remain fluid even during long-running calls. The Realtime API also offers innovative capabilities such as image input support, SIP phone network integration, remote MCP server connection, and reusable conversation prompts.
    Starting Price: $20 per month
  • 17
    Qwen3.5-Omni
    Qwen3.5-Omni is a next-generation, fully multimodal AI model developed by Alibaba that natively understands and generates text, images, audio, and video within a single unified system, enabling more natural and real-time human-AI interaction. Unlike traditional models that treat modalities separately, it is trained from the ground up on massive audiovisual datasets, allowing it to process complex inputs such as long audio streams, video, and spoken instructions simultaneously while maintaining strong performance across all formats. It supports long-context inputs of up to 256K tokens and can handle over 10 hours of audio or extended video sequences, making it suitable for demanding real-world applications. A key feature is its advanced voice interaction capabilities, including end-to-end speech dialogue, emotional tone control, and voice cloning, enabling highly natural conversational experiences that can whisper, shout, or adapt speaking style dynamically.
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    Ellipsis Health Sage

    Ellipsis Health Sage

    Ellipsis Health

    Ellipsis Health is an AI-powered care management platform centered around its virtual agent, Sage, designed to automate and enhance patient engagement through emotionally intelligent voice interactions that integrate directly into clinical workflows. Sage conducts fully autonomous, multilingual phone conversations with patients, handling tasks such as program enrollment, eligibility verification, copay checks, and answering patient questions, while also performing assessments including health risk evaluations, discharge follow-ups, satisfaction surveys, and outcomes tracking. It supports clinical operations by coordinating care, monitoring adherence, and conducting pre- and post-discharge check-ins, helping healthcare organizations maintain continuity of care and improve quality metrics. It is built on an “empathy engine” that analyzes vocal biomarkers such as tone, pace, and speech patterns to detect emotional and mental health signals.
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    Qwen3-TTS

    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).
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    Chatterbox

    Chatterbox

    Resemble AI

    Chatterbox is a free, open source voice cloning AI model developed by Resemble AI, licensed under MIT. It enables zero-shot voice cloning using just 5 seconds of reference audio, eliminating the need for training. The model offers expressive speech synthesis with unique emotion control, allowing users to adjust the intensity from monotone to dramatically expressive with a single parameter. Chatterbox supports accent control and text-based controllability, ensuring high-quality, human-like text-to-speech conversion. It operates with faster-than-real-time inference, making it suitable for real-time applications, voice assistants, and interactive media. The model is built for production and designed for developers, featuring simple installation via pip and comprehensive documentation. Chatterbox includes built-in watermarking using Resemble AI’s PerTh (Perceptual Threshold) Watermarker, embedding data imperceptibly to protect generated audio content.
    Starting Price: $5 per month
  • 21
    Raven-1

    Raven-1

    Tavus

    Raven-1 is a multimodal, real-time perceptual AI model from Tavus designed to bring emotional intelligence to artificial intelligence by interpreting human audio, visual, and temporal signals together instead of reducing communication to text alone. It unifies tone, facial expression, body language, hesitation, and contextual dynamics into a rich, unified representation of user intent and state, enabling conversational AI to understand how people communicate in real time with nuanced natural language descriptions rather than static emotion labels. It was engineered to overcome the limitations of traditional systems that rely on transcripts and limited emotion scoring by capturing subtle cues, such as emphasis, sarcasm, engagement shifts, and evolving emotional arcs, and continuously updating this understanding with low latency so responses align with the true context of the interaction.
    Starting Price: $59 per month
  • 22
    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
  • 23
    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.
  • 24
    Cartesia Ink-Whisper
    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
  • 25
    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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    Grok Voice Think Fast 1.0
    Grok Voice Think Fast 1.0 is an advanced voice AI model developed by xAI, designed to handle complex, real-world conversational workflows. It excels in multi-step tasks across customer support, sales, and enterprise applications. The model is built for fast, natural conversations while maintaining high accuracy and responsiveness. It supports real-time reasoning without adding latency, allowing it to process and respond intelligently during live interactions. Grok Voice can accurately capture and confirm structured data such as names, addresses, and account details, even in noisy or challenging conditions. It is optimized for global use with support for over 25 languages. The model is capable of handling interruptions, accents, and ambiguous inputs with ease. Overall, it enables businesses to deploy efficient, scalable voice agents for high-volume interactions.
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    Orpheus TTS

    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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    Cartesia Sonic
    Sonic is the fastest, ultra-realistic generative voice API, powered by our next-gen state space model and purpose-built for developers. With a time-to-first audio of 90ms, Sonic is the fastest generative voice model, with best-in-class quality and controllability. Built for streaming using our first-of-its-kind low-latency state space model stack. Fine-grained control over pitch, speed, emotion, and pronunciation. Sonic ranks #1 in quality in independent evaluations of quality. Sonic supports seamless speech in 13 languages, with more added to every release. From Japanese to German, any language you need, we’ve got it. Localize a given voice to any accent or language. Power support experiences that delight your customers. Bring your storytelling to life with immersive voices. Create content that engages viewers and drives clicks. Narrate content for podcasts, news, and publishing, and empower healthcare with voices that patients trust.
    Starting Price: $5 per month
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    Hume AI

    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
  • 30
    All Voice Lab

    All Voice Lab

    All Voice Lab

    All Voice Lab is an innovative AI tool that reshapes audio workflows with a range of AI-powered solutions. The tool offers text to speech technology, voice cloning and voice altering capabilities that bring authenticity and lifelikeness to audio projects. Text to Speech technology can be utilized for various applications, from audiobooks to video voiceovers, it enhances the overall output by offering realistically engaging voices. Advanced emotion recognition and voice style modelling enable the AI to adapt to text sentiment and adjust the tone, pitch, and rhythm in real-time, thereby resulting in natural and emotionally expressive speech. The tool supports 33 languages - providing consistent tone and style across different languages and perfect for global content creation. With the voice cloning technology, users can achieve precise replication of their tone, pitch and rhythm, and multilingual capabilities.
    Starting Price: $3/month
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    GPT-Realtime-Translate
    GPT-Realtime-Translate is OpenAI’s live translation model for building multilingual voice experiences where each person can speak in their preferred language, hear the conversation translated in real time, and read real-time transcriptions. It supports more than 70 input languages and 13 output languages, making it useful for customer support, cross-border sales, education, events, media, and creator platforms serving global audiences. It is designed to preserve meaning while keeping pace with the speaker, even when people speak naturally, switch context, use regional pronunciation, or rely on domain-specific language. GPT-Realtime-Translate helps cross-language conversations feel more natural by combining lower latency, stronger fluency, and real-time speech translation in one API workflow. It can support live multilingual voice interactions, translate conversations as they happen, and make spoken content accessible to audiences.
    Starting Price: $0.034 per minute
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    VoiceBun

    VoiceBun

    VoiceBun

    VoiceBun is an open source, no-code voice-agent builder that lets you create, configure, and deploy AI-powered conversational assistants entirely via natural-language prompts. It combines speech-to-text, large-language models, and text-to-speech into a unified platform where you define your agent’s goals, initial greeting, tool integrations and data sources; VoiceBun automatically generates the underlying conversational logic, state management and API connectors needed to handle inbound and outbound calls for support, scheduling, lead qualification and more. The web-based interface gives you mobile-friendly access and isolated deployments through user-specific subdomains, while built-in analytics surface call transcripts, usage metrics, success rates, and sentiment trends. Integration includes options for telephony, webhook actions for external workflows, and role-based access controls with encrypted credentials for enterprise security.
    Starting Price: $20 per month
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    PlayAI

    PlayAI

    PlayAI

    PlayAI is a voice intelligence platform that enables businesses to create highly realistic, human-like AI voices for a variety of applications. The platform provides tools for building voice agents that can be deployed across web platforms, mobile apps, and phone systems. PlayAI's voice models are designed to sound fluid and emotive, enhancing customer support, personal assistance, and even front desk interactions. With flexible deployment options, the platform supports applications like voiceover creation, podcasts, and more, making it an ideal solution for companies looking to integrate conversational AI into their services.
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    Azure Text to Speech
    Build apps and services that speak naturally. Differentiate your brand with a customized, realistic voice generator, and access voices with different speaking styles and emotional tones to fit your use case—from text readers and talkers to customer support chatbots. Enable fluid, natural-sounding text to speech that matches the intonation and emotion of human voices. Tune voice output for your scenarios by easily adjusting rate, pitch, pronunciation, pauses, and more. Engage global audiences by using 400 neural voices across 140 languages and variants. Bring your scenarios like text readers and voice-enabled assistants to life with highly expressive and human-like voices. Neural Text to Speech supports several speaking styles including newscast, customer service, shouting, whispering, and emotions like cheerful and sad.
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    Voicebridge

    Voicebridge

    Voicebridge

    VoiceBridge AI is the world’s first web‑based, hands‑free voice interviewing platform powered by empathetic AI agents that conduct multiple conversational interviews simultaneously. Users set objectives and share a participation link, and “Ava”, the multilingual AI agent, leads natural voice dialogues, capturing responses which are instantly converted into transcripts, emotional insights, summaries, authentic quote posters, and authenticated testimonials. It scales to hundreds of interviews at once, supports synthetic persona testing and global panels, and delivers real‑time analytics with theme detection. It emphasizes privacy with encryption and identity masking, enabling product teams, marketers, HR professionals, and research groups to quickly surface high-quality voice feedback for churn reduction, product‑market fit, employee engagement, and content creation, all within minutes and without complex setup.
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    Chikka.ai

    Chikka.ai

    Chikka.ai

    Chikka.ai is an AI-powered voice interviewing platform featuring “Ava,” an empathetic, multilingual AI voice agent that conducts dynamic and natural voice interviews at scale. Users simply define objectives, invite participants via a shareable link, and Ava leads the conversation, capturing authentic feedback securely. Chikka.ai instantly converts recordings into transcripts, emotional insights, summaries, shareable quote posters, and marketing-ready testimonials authenticated by its VoiceVerify engine to ensure credibility. It supports hundreds of interviews concurrently, offers synthetic persona test-runs, global respondent panels, and robust privacy protections with encryption and identity masking. Real-time analytics and theme detection help teams uncover hidden opportunities, reduce churn, inform product-market fit, refine employee engagement, and generate content-driven marketing materials.
    Starting Price: $19.90 per month
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    Vocode

    Vocode

    Vocode

    Vocode is an open source library that simplifies the creation of voice-based applications leveraging large language models. Developers can build real-time streaming conversations with LLMs and deploy them to phone calls, Zoom meetings, and more. Vocode provides easy abstractions and integrations so that everything you need is in a single library. It offers out-of-the-box integrations with leading speech-to-text and text-to-speech providers, including AssemblyAI, Deepgram, Google Cloud, Microsoft Azure, and Whisper. The platform supports cross-platform deployment across telephony, web, and Zoom, enabling applications like LLM-powered phone calls, personal assistants, and voice-based games. Vocode's modular design allows for seamless integration of various AI models and services, providing developers with the flexibility to choose the best components for their applications. The platform also supports multilingual capabilities.
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    gpt-4o-mini Realtime
    The gpt-4o-mini-realtime-preview model is a compact, lower-cost, realtime variant of GPT-4o designed to power speech and text interactions with low latency. It supports both text and audio inputs and outputs, enabling “speech in, speech out” conversational experiences via a persistent WebSocket or WebRTC connection. Unlike larger GPT-4o models, it currently does not support image or structured output modalities, focusing strictly on real-time voice/text use cases. Developers can open a real-time session via the /realtime/sessions endpoint to obtain an ephemeral key, then stream user audio (or text) and receive responses in real time over the same connection. The model is part of the early preview family (version 2024-12-17), intended primarily for testing and feedback rather than full production loads. Usage is subject to rate limits and may evolve during the preview period. Because it is multimodal in audio/text only, it enables use cases such as conversational voice agents.
    Starting Price: $0.60 per input
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    Voicing AI

    Voicing AI

    Voicing AI

    Voicing AI is an enterprise-grade agentic voice AI platform designed to automate customer interactions through humanlike voice agents that can both converse and take real-time actions during calls. It enables businesses to handle inbound and outbound phone calls 24/7 using AI agents that understand queries, respond naturally, and execute tasks such as updating CRM systems, retrieving data, or completing workflows without human intervention. It is built around proprietary “large action models” that allow agents not only to communicate but also to perform operations across integrated systems, significantly accelerating task execution. It supports multilingual conversations in over 20–30 languages and incorporates high emotional and contextual intelligence to handle complex customer interactions with accuracy and empathy.
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    Respeecher

    Respeecher

    Respeecher

    Create speech that's indistinguishable from the original speaker. Replicate voices for any media project — from a Hollywood movie to an engaging video game. Our machine-learning technology masters every aspect of your target voice to create a spot-on match. Our system leverages recent revolutionary advances in artificial intelligence. We combine classical digital signal processing algorithms with proprietary deep generative modeling techniques to learn your target voice inside and out. Make changes to the script of the performance anytime during the creative process without re-recording the target voice. Edit a plot line on the fly. Bring back the voice of a beloved actor who has passed away. Whatever the reason, Respeecher can ensure that your creative vision is achieved. Our voice swaps are virtually indistinguishable from the original — and never sound robotic. They convey all the nuances and emotions of human speech and have the highest production value.
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    Gemini Live API
    ​The Gemini Live API is a preview feature that enables low-latency, bidirectional voice and video interactions with Gemini. It allows end users to experience natural, human-like voice conversations and provides the ability to interrupt the model's responses using voice commands. The model can process text, audio, and video input, and it can provide text and audio output. New capabilities include two new voices and 30 new languages with configurable output language, configurable image resolutions (66/256 tokens), configurable turn coverage (send all inputs all the time or only when the user is speaking), configurable interruption settings, configurable voice activity detection, new client events for end-of-turn signaling, token counts, a client event for signaling the end of stream, text streaming, configurable session resumption with session data stored on the server for 24 hours, and longer session support with a sliding context window.
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    ElevenAgents

    ElevenAgents

    ElevenLabs

    ElevenLabs Agents is a platform for building, deploying, and scaling intelligent conversational AI agents that can speak, type, and take action across phone, web, and application environments. It enables developers and teams to create real-time agents that interact naturally with users through voice and text, combining speech-to-text, large language models, and text-to-speech into a unified system that functions like a human conversation partner. It allows agents to resolve customer issues, automate workflows, answer questions, and execute tasks based on connected data sources and predefined logic, making interactions both accurate and context-aware. These agents can be customized with knowledge bases, system prompts, and tools that enable them to access external systems, execute custom logic, and perform actions beyond simple responses. They support multimodal capabilities, meaning they can read, speak, and interpret inputs while handling conversational dynamics.
    Starting Price: $5 per month
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    Uservox

    Uservox

    Uservox

    Uservox.ai is an AI voice automation platform to transforms customer engagement. It automates routine voice conversations, letting teams focus on high value interactions. The AI voice agents sound natural, understand context, and handle real customer interactions across multiple languages, managing Level 1 support, lead qualification, payment reminders, feedback collection, and CRM updates without human intervention. The platform captures every call and lead while providing actionable insights into customer behavior and increasing operational efficiency. Unlike traditional IVRs, this delivers a completely human like experience, understanding intent, tone, and emotion, while being available 24/7. Businesses handling high call volumes can automate up to 80% of routine interactions, reduce operational costs, scale their reach, and improve efficiency while delivering a real conversational experience that customers trust.
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    Grok Voice Agent
    The Grok Voice Agent API is xAI’s new developer platform for building fast, intelligent, and multilingual voice agents. It is powered by the same in-house voice technology used by Grok Voice in mobile apps and Tesla vehicles. The API enables voice agents to speak dozens of languages, call tools, and search real-time data. Grok Voice Agents are engineered for low latency, delivering audio responses in under one second. The platform ranks first on the Big Bench Audio benchmark for voice reasoning performance. Developers benefit from a simple, flat pricing model based on connection time. The Grok Voice Agent API brings production-proven voice intelligence to custom applications.
    Starting Price: $0.05 per minute
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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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    AI Voicer
    Get ready to unlock the extraordinary with AI Voicer, the game-changing text-to-speech app that's redefining the way you speak. Transform written words into captivating spoken narratives with unmatched clarity and emotion. Download AI Voicer, powered by ElevenLabs, and embark on a journey of text-to-speech mastery, voice cloning, dictation, and more. Elevate your voice with AI Voicer – where your words come alive and cover new horizons in the world of TTS and voiceovers. Step into the future of voiceover with our remarkable cloning technology.
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    ERNIE 5.0
    ERNIE 5.0 is a next-generation conversational AI platform developed by Baidu, designed to deliver natural, human-like interactions across multiple domains. Built on Baidu’s Enhanced Representation through Knowledge Integration (ERNIE) framework, it fuses advanced natural language processing (NLP) with deep contextual understanding. The model supports multimodal capabilities, allowing it to process and generate text, images, and voice seamlessly. ERNIE 5.0’s refined contextual awareness enables it to handle complex conversations with greater precision and nuance. Its applications span customer service, content generation, and enterprise automation, enhancing both user engagement and productivity. With its robust architecture, ERNIE 5.0 represents a major step forward in Baidu’s pursuit of intelligent, knowledge-driven AI systems.
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    Ori

    Ori

    Ori

    Ori is an enterprise-grade generative-AI platform built to automate and scale customer interactions across voice, chat, email, and messaging channels, with full compliance, auditability, and multilingual support. It delivers AI-powered chatbots and voice bots capable of handling the full customer journey; lead qualification, conversational sales, onboarding, customer support, collections, renewals, and retention. Its core features include multilingual and omnichannel support, intelligent conversation flows with context awareness and sentiment detection, real-time compliance and script adherence (for regulated industries like finance and insurance), full audit trails, and seamless handoffs to human agents when needed. It supports voice-based conversations (speech recognition, natural-language responses), chat/text conversations, email responders, and hybrid bot-plus-live-agent workflows.
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    OpenAI.fm
    OpenAI.fm is an innovative platform from OpenAI, enabling users to explore and experiment with their latest audio models. It serves as an interactive space where users can try out, tweak, and share text-to-speech transformation features. The platform offers various voice options and gives users the ability to customize speaking styles, including altering emotional tone and character voices. Targeted at developers, content creators, and AI enthusiasts, OpenAI.fm provides a hands-on environment for those interested in discovering and working with AI-generated voices.
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    MetaSoul

    MetaSoul

    MetaSoul

    MetaSoul® is the revolutionary technology that brings emotional depth and Personas to Artificial Intelligence. They help to understand and make sense of experiences; they provide a sense of direction and motivation. Make your avatars unique and more autonomous with a MetaSoul®; multiply their value as they develop skill sets. Introducing MetaSoul Azure API: Revolutionizing Emotional AI Voices and OpenAI Ehanced Persona Do you want to avoid the complexities and challenges when combining OpenAI and Microsoft Neural Text to Speech to achieve nuanced emotions in your applications? Managing emotions and persona for each phrase and adjusting intensity in real-time can be cumbersome. Fear not, as we present MetaSoul Azure API, the ultimate solution for effortless integration and unparalleled emotional AI voices and faces.
    Starting Price: $5 per month per user