Article·Oct 27, 2025

What Is Voice AI & How It Works

Voice AI processes thousands of calls per hour for enterprises. Learn how speech-to-text, voice agents, and TTS work in production at scale.

8 min read

By Bridget McGillivray

Last Updated

Voice AI processes thousands of concurrent calls per hour, enabling organizations to handle workloads beyond human capacity. This guide explains how the technology works, the metrics it moves for companies running it at scale, and the technical criteria that matter when evaluating providers for production deployment.

What is Voice AI?

Voice AI is a technology that enables computers to understand speech, interpret intent, and respond naturally in conversation. Legacy IVR (Interactive Voice Response) systems were never able to achieve such capabilities.

Modern voice systems chain together four specialized engines that work in sequence:

  1. Speech-to-text converts spoken words into text that software can process. Neural models handle accents, crosstalk, and background noise because they're trained on massive speech datasets from actual customer interactions.
  2. Natural Language Processing and Understanding interpret intent and extract entities, typically powered by large language models that maintain context across conversation turns.
  3. Dialogue management tracks conversation state, determines the next action, and orchestrates backend system calls.
  4. Text-to-speech synthesizes natural-sounding responses with appropriate prosody and emotion.

This pipeline powers everything from everyday interactions like "Hey Siri" or "Alexa, reorder coffee" all the way to enterprise use cases like triaging support calls, authenticating customers, and gathering analytics at scales human teams can't reach.

Let’s look at each of the four stages in detail.

Speech-To-Text (STT)

Microphones capture raw audio that immediately undergoes digital signal processing to filter out background noise and normalizing volume levels. The cleaned audio then gets fed into STT models that map sounds to words. These models, which are trained on thousands of hours of audio data, handle accents, filler words, and rapid speech. And they can generate transcripts in less than a second to keep the conversation flowing.

The real-world accuracy of STT depends on signal quality and speaker variables. Audio issues like low-bit-rate files, shaky microphones, and background chatter can raise word-error rates, while accents, rapid speech, and industry jargon create additional challenges.

Then there’s the constant tension between accuracy and latency. High-accuracy systems require more processing complexity, which increases latency. Low-latency systems sacrifice some accuracy to achieve faster response times.

Natural Language Processing and Understanding

Large language models parse the transcript to understand what the caller actually wants, not just what words they said.

This involves two critical operations: entity extraction and intent classification. Entity extraction identifies specific data points like flight numbers, dates, account IDs, and policy numbers from natural speech. Intent classification determines the user's goal, whether they want to reschedule an appointment, check their account balance, or report a problem.

Context trackers maintain conversation history across multiple turns, enabling users to interrupt mid-sentence, ask follow-up questions, or change direction naturally without repeating information. When a caller says "actually, make that Tuesday instead," the system understands "that" refers to the appointment date mentioned earlier.

Dialogue Management

Once intent is clear, the system executes the appropriate function and determines the next action. Dialogue management tracks conversation state across the entire interaction, knowing what information has been collected, what's still needed, and which backend systems to call.

For example, if a caller wants to reschedule a flight, the system might need to check availability, verify account status, and confirm payment method before completing the transaction.

This orchestration layer coordinates multiple backend system calls while maintaining conversational flow. Rather than forcing users through rigid menu trees, dialogue management can adapt to conversational patterns. It can handle ambiguous responses, ask clarifying questions when needed, and recover gracefully when users provide unexpected input or change their minds mid-conversation.

Text-To-Speech (TTS)

Finally, TTS converts the response into natural audio, adding appropriate pauses and emphasis so it sounds professional rather than robotic.

Production TTS systems must handle specialized content correctly. Entity-aware processing is responsible for ensuring addresses, account numbers, and regulatory terminology are pronounced naturally on the first attempt. This is crucial to avoid situations like where a customer might hear "one-two-three Main Street" instead of "123 Main Street" or mispronounced medication names in healthcare calls.

Of course, real-world conditions complicate this entire process. Contact centers, for example, deal with overlapping speakers, equipment noise, and industry-specific terminology that prove challenging for voice AI systems. Modern pipelines compensate with strategies like noise-robust training and dynamic vocabulary injection, which are crucial when enterprise systems need to handle high-volume concurrent calls without degrading accuracy or response time.

Tying It All Together With Conversational AI Agents (Voicebots)

Conversational AI agents act as complete voice AI systems, handling customer calls by listening, understanding intent beyond literal words, and executing requests in real time. Unlike legacy IVR systems (and even some generic voice AI systems) that force callers through rigid menus, these agents use end-to-end speech processing that skips text conversion and deliver real-time responses while handling natural interruptions mid-sentence.

This experience depends on four technical capabilities to work in production environments:

  • Context retention lets the agent remember details from earlier in the conversation, so customers don't repeat themselves when switching topics
  • Interruption handling mirrors natural conversation flow. Customers can jump in without breaking the interaction or forcing restarts
  • Function calling triggers backend actions like order lookups or password resets while maintaining conversational flow
  • Entity extraction validates account numbers, policy IDs, and dates automatically without explicit confirmation steps

Early chatbots locked users into predetermined scripts that were vulnerable to conversations going off-track. Current voice agents powered by large language models can adapt questions dynamically, recognize emotional context, and switch languages on-the-fly. This is the exact technology enterprise deployments use to process high volumes of concurrent calls while maintaining accuracy.

How Does Voice AI Benefit Enterprises?

The business case for voice AI centers on four improvements that directly impact bottom-line performance:

  • Cost reduction: Contact centers that deploy voice agents significantly reduce routine call-handling costs. Organizations pay for compute minutes instead of headcount, and that spend stays linear even when call volumes surge.
  • Customer satisfaction: Voice agents answer rapidly (typically in under one second), understand open-ended questions, and never put anyone on hold. This drives higher CSAT scores than menu-based IVR systems because instant engagement becomes the baseline.
  • Data capture: Speech recognition transcribes and analyzes every word in real time. This data become immediately available for search and review.
  • Scalability: Voice AI can handle both high volumes and multilingual applications, meaning operations can scale in numbers and across regions.

Why Deepgram is The Best Choice Among Voice AI APIs

Deepgram delivers voice AI that works in production through three core differentiators: scale, flexibility, and cost transparency.

  • Deepgram's infrastructure is built to handle production workloads that generic APIs are unable to match. The platform is capable of processing thousands of concurrent calls while maintaining sub-second latency and accuracy, even with noisy call-center audio, accented speech, and industry-specific terminology.
  • An API-first design means Deepgram can be integrated with WebSocket connections immediately, then deployed to private cloud or on-premises environments as requirements evolve. This flexibility is critical for teams navigating HIPAA, GDPR, or internal data-residency requirements.
  • Deepgram’s pricing stays predictable through transparent per-minute rates. There’s even a free tier that allows prototyping without procurement delays.

Organizations building products that handle real customer conversations can sign up for a free Deepgram console account and get $200 in credits to test production-grade speech infrastructure.