Voice

Last updated 10 Sept 2025

Build voice AI agents with real-time media streaming

These days, every product pitch mentions AI. Most teams have a model, a chatbot, or some form of automation. But building AI agents that handle live phone conversations is a different challenge from training an LLM. It demands real-time processing, natural speech, and voice infrastructure that keeps up with the pace of human interaction.

If your agent lags, interrupts, or sounds robotic, callers notice. They hang up. That’s why real-time media streaming is essential. It gives your AI immediate access to call audio and enables it to respond just as quickly. Let’s look at how it works, and what you need to build it.

Why media streaming matters for low-latency voice AI

For voice AI to sound natural, latency must be between 200-500ms anything longer and conversations begin to feel disconnected. Media streaming keeps latency low by sending audio to your application the instant it’s spoken. The speed lets your AI agent process and reply within the same interaction, keeping conversations fluid. You might be thinking, do I really need to keep latency that low?

Imagine you’re building a sales coaching agent. As the rep speaks with a prospect, the audio is streamed to your AI engine. The speech is transcribed, analyzed for sentiment and intent, and used to generate live coaching prompts during the call.

This kind of in-call enablement is only possible with instant access to audio and the ability to inject responses inside the same session.Once your system can operate at this level, new use cases open up: escalating when tone shifts, verifying identity through voice, or translating instantly.

All of this relies on sub-second processing and a reliable telephony backbone. That’s the foundation Telnyx was built on. To deliver on that promise, you need the right architecture.

How real-time streaming works

At its core, streaming is about giving your AI agent live access to call audio. The provider will take the media from an active phone call and create a parallel copy, which it streams over a WebSocket connection. This ensures your application can analyze and respond without interrupting the call itself. From there, you can connect the stream to speech-to-text, text-to-speech, or other AI services.

Here’s a quick look at the pipeline:

  1. Capture: Audio is picked up from the call
  2. Encode: Compressed with a low-latency codec
  3. Transport: Sent over a WebSocket
  4. Process: Transcribed and analyzed by your AI stack
  5. Respond: Audio is injected back into the live call

If you’d like a deeper dive into the fundamentals of WebSocket streaming, we’ve covered that in an earlier post. What matters here is what comes next: how Telnyx extends this basic pipeline with codec flexibility, call control, and a global network to make it production-ready.

Real-time Media Streaming Presentation.png

Build production-ready voice AI Agents with Telnyx

Streaming audio into your AI stack is powerful, but it’s only the start. To make your agent reliable in real-world conditions, you also need flexible audio handling, intelligent call control, and a network engineered for real-time performance. Telnyx delivers all three.

Codec flexibility

Different AI services expect different audio formats. Traditionally, that meant the codec used on your phone call had to match what your AI could accept and if it didn’t, you had to build converters just to make things work.

Telnyx removes that barrier. You can request audio in one codec for the call and another for the stream, no extra processing required. We’ve also added support for L16, giving you even more options for high-quality integrations.

Currently supported codecs for bidirectional streaming include:

  • PCMU (default)
  • PCMA
  • G722
  • OPUS (8/16 kHz)
  • AMR-WB (8/16 kHz)
  • L16 (16 kHz)

Some providers don’t support certain codecs at all. For example, Twilio does not support PCMA on WebSocket streaming. This is a limitation across Europe, the Middle East, and Asia, making it harder to scale your AI-powered communications globally.

Call control

Real-time streaming gives your AI the ability to listen and respond. But what happens when the assistant needs to act? With the Telnyx Voice API, you can control the call itself; answer or end it, transfer participants, collect keypad inputs, or even trigger actions based on sentiment and phrases.

Telnyx also provides extra call features that help your AI Agents perform optimally:

  • HD Voice to provide a clear sound with a broader range of sound frequencies
  • Noise suppression to improve transcription accuracy
  • Answering machine detection so your AI knows who it’s talking to

👉 Explore the full list of Voice API features here.

A network built for real-time performance

For over a decade, Telnyx has engineered its own global telephony network to deliver ultra low-latency communications. Unlike providers that depend on third parties, Telnyx owns and operates its backbone. That means consistent call quality, cost efficiency, and the reliability to scale voice AI worldwide.

Getting started

Now that you know what’s possible, here’s how to get started.

Option 1: Stream with your own AI stack

If you already have your own AI model, Telnyx can stream live audio directly to your engine or preferred LLM. You can also mix in our speech services or connect third-party tools for transcription, synthesis, or analytics.

Getting started is simple:

  • Create a Telnyx account and buy a number
  • Configure media streaming with your WebSocket server
  • Initiate a stream with the Voice API or TeXML. See the difference between Voice API and TeXML so you can choose the best fit.

👉 For full setup instructions, see the Voice API quick start guide, TeXML application quick start guide, and the media streaming guide.

Technical Notes for Developers

  • Streaming Modes
    inbound_track: Caller audio
    outbound_track: Agent or assistant audio
    both_tracks: Full conversation context
  • Stream Payloads
    Telnyx delivers base64-encoded RTP packets enriched with metadata (timestamps, codec info, stream IDs). Easily fed into STT, TTS, or analytics engines.
  • Streaming Methods
    RTP: Best for conversational AI needing instant response
    MP3: Best for pre-recorded prompts or hold messages

Option 2: Build fully on Telnyx

For teams that want a faster path, Telnyx offers a no-code AI Assistant Builder. Design call flows, configure AI Agent responses, and go live in minutes. It runs on the same foundation of telephony, call control, and media streaming, and adds TTS, STT, and GPU-powered AI orchestration. That means you can launch a voice AI agent quickly on one platform.

Check out our guide on how to build great AI voice agents for best practices, architecture examples, and tips to scale from prototype to production.

Build voice AI that works

Real-time voice AI is hard to get right but with the right tools, it doesn’t have to be. Telnyx gives you a full-stack solution for streaming, control, and global connectivity, so your AI Agents can sound natural and respond fast. From prototype to production, Telnyx helps you move faster and deliver a better experience on every call.

Ready to build voice AI with real-time media streaming? Contact our team to get started!
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Deniz-Yakışıklı-AvatarDeniz Yakışıklı

Sr. Product Marketing Manager

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