WebRTC server infrastructure

WebRTC server infrastructure for the connection layer.

Xirsys gives real-time products the STUN/TURN infrastructure underneath WebRTC: global relay routing, API credentials, static server credentials, NAT traversal, and dashboard analytics without replacing your app or media server.

Bring your stack

Keep your media, signaling, and app experience. Add dependable connectivity.

Some products need an SFU, media server, SDK, model provider, or custom backend. Xirsys does not replace those pieces. It provides the neutral connectivity service they rely on when users need STUN/TURN access.

Browsers and apps

Return standard ICE configuration to browser WebRTC and native client applications.

Media servers

Support server-side deployments with static non-expiring TURN credentials.

SDK products

Add connectivity fallback under calling, collaboration, streaming, and support workflows.

AI agents

Give voice and video agents a relay layer for restrictive user networks.

Server platform intent

A WebRTC server platform should be easy to integrate and easy to leave alone.

Teams searching for WebRTC server infrastructure usually need reliability, coverage, and operating leverage. Xirsys focuses on that infrastructure layer: credentials, regions, relay bandwidth, uptime, and support.

01

Global routing

Production regions route users to healthy relay capacity close to their network path.

02

Credential API

Automate short-lived ICE credentials or use static credentials for infrastructure services.

03

Bandwidth planning

Size plans around TURN-relayed traffic while direct STUN sessions remain free.