Real-Time Bi-Directional Audio
Architecting high-throughput, low-latency WebSocket servers to handle continuous bi-directional audio streams for AI.
Why WebSocket Audio Streaming Matters
Voice AI cannot use standard HTTP requests. It requires a persistent, bi-directional connection to stream audio bytes in real-time as the user speaks. WebSockets are the foundation of this.
Employer Demand
Required for Senior Backend Engineers building real-time collaboration or voice applications.
How We Use It
We build highly optimized Node.js and Rust WebSocket servers that ingest 8kHz/16kHz audio streams, process them, and emit TTS audio chunks back to the telephony provider.
Real World Example
We engineered a clustered WebSocket architecture that maintains stateful connections for up to 5,000 concurrent AI voice calls without dropping frames.
The Slickrock Advantage
"We manage memory aggressively to prevent memory leaks associated with long-lived WebSocket connections, ensuring flawless uptime."
Frequently Asked Questions
Why not use WebRTC instead of WebSockets?
WebRTC is excellent for browser-to-browser communication, but WebSockets are often preferred for server-to-server telephony integrations (like Twilio Media Streams) due to protocol simplicity.