Building real-time voice AI solutions using WebSockets

Real-time media streaming lets you send live audio instantly over a steady connection, enabling immediate processing and response. Unlike traditional methods that rely on multiple connections and add delays, WebSocket technology keeps a direct, two-way link open, ensuring ultra-low latency. This makes it a key component in voice AI applications, where speed and accuracy are critical.

In this post, we’ll break down how real-time streaming works, how Telnyx uses WebSockets to power AI-driven voice solutions, and how you can set up streaming for applications like transcription, sentiment analysis, and live translation.

The technical basics of real-time media streaming

Real-time media streaming involves capturing live audio data and transmitting it instantly using WebSocket technology. Unlike HTTP, which requires multiple connections for continuous data exchange, WebSockets maintain an open, two-way communication channel over a single TCP connection. This approach significantly reduces latency by eliminating the need for repeated connections.

WebSockets handle data efficiently due to their binary protocol, offering faster data exchange and reduced delays compared to traditional text-based methods. This efficiency is particularly important for voice applications that require minimal delay to function effectively.​

Real-world examples of real-time streaming

AI-powered speech processing

Real-time audio streaming significantly enhances AI-powered speech processing applications. Voice assistants and chatbots benefit from immediate analysis and quick responses, making interactions feel more natural and responsive.

Keyword detection, another application, enables automated monitoring and reaction to specific phrases instantly. Sentiment analysis capabilities allow immediate recognition of emotional tones in conversations, providing valuable insights into customer interactions and enabling timely interventions.

Live translation and transcription

Immediate speech-to-text conversions and real-time translations are major benefits of live audio transmission. Live transmission significantly improves communication in multilingual contexts, valuable in international business interactions, customer support, large-scale webinars, virtual conferences, or global broadcasts. Quick transcription and translation ensure smoother communication to reduce misunderstandings and improve efficiency.

Fraud detection and security

Security applications, especially fraud detection, greatly benefit from live audio streaming. Voice biometric authentication uses streaming technology to verify caller identities instantly, preventing unauthorized access and fraud. Immediate processing of audio streams allows real-time analysis of voice patterns, significantly enhancing security in telecommunication and customer verification processes.

Real-time coaching and call whispering

In customer support environments, instant streaming simplifies real-time coaching and call whispering. Supervisors can listen in, guide agents, or provide immediate assistance without complicated setups. Being able to coach and call whisper improves training effectiveness, enhances customer interactions, and ensures agents receive timely support during interactions.

Integrating real-time streaming with AI services

Telnyx’s streaming technology is particularly effective for AI integration. You can connect Telnyx WebSocket streams directly with AI services, enabling immediate analysis, interactive responses, and voice synthesis.

AI-powered chatbots become more responsive, with calls instantly analyzed by AI platforms such as OpenAI, Gemini, or Deepseek. Immediate keyword or emotion detection provides instant feedback and support, significantly enhancing user experiences.

Additionally, real-time voice-based coaching becomes more impactful by analyzing live conversations, assessing speech patterns, and delivering instantaneous recommendations to support agents.

How to set up real-time media streaming

You can get started with real-time streaming in just four steps:

1. Sign up for a Telnyx account.
2. Create a Voice API or TeXML configuration to manage your call connections.
3. Set up an Outbound Voice Profile for your outbound call settings linked to your Credential Connection.
4. Create a TeXML bin or use dial endpoint to place your first call.

After completing these prerequisites, you can set up a WebSocket server URL to manage incoming and outgoing audio streams. Audio data from Telnyx’s service is then processed through your AI applications, with generated responses immediately returned to the ongoing call.

You can select between MP3-based streaming for less demanding scenarios or longer playback, and RTP streaming, ideal for low-latency, interactive use cases. For more detailed information on setting up media streaming over WebSockets, check out our developer guidelines.

Technical considerations

Proper audio encoding is essential for compatibility and performance. Telnyx supports multiple codecs, specifically PCMU (8 kHz, default), PCMA (8 kHz), G722 (8 kHz), OPUS (8 kHz, 16 kHz), and AMR-WB (8 kHz, 16 kHz). Appropriate encoding ensures optimal performance and enhances transcription accuracy.

The MP3 streaming method is easier to implement for general use cases, while RTP streaming offers lower latency, ideal for real-time interactions and immediate audio feedback.​

Improve your voice AI with real-time streaming

Real-time media streaming over WebSockets enhances AI-driven applications by providing instant, high-quality audio data for immediate analysis and response. Telnyx’s Media Streaming API enables you to build advanced, scalable, and responsive AI solutions, significantly improving communication experiences.