Last updated 10 Sept 2025
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.
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.
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:
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.
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.
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:
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.
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:
👉 Explore the full list of Voice API features here.
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.
Now that you know what’s possible, here’s how to get started.
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:
👉 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
inbound_track
: Caller audio outbound_track
: Agent or assistant audio both_tracks
: Full conversation contextFor 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.
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.
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