WebRTC NAT traversal

WebRTC NAT traversal that keeps sessions connected.

WebRTC works best when peers can connect directly. Real networks add symmetric NATs, carrier-grade NAT, VPNs, firewalls, and UDP restrictions. Xirsys gives every session STUN discovery and TURN relay fallback through production infrastructure.

STUN first, TURN when needed

NAT traversal needs both direct discovery and reliable fallback.

A connection layer should help easy sessions stay direct and give difficult sessions a dependable relay path. Xirsys returns standard ICE server configuration for browsers, native apps, SDKs, media servers, and AI agents.

01

STUN discovery

Let WebRTC clients discover viable peer paths across common home, office, and mobile network conditions.

02

TURN relay fallback

Route sessions through Xirsys relay regions when NAT, firewall, VPN, or carrier policies block direct connectivity.

03

ICE credentials

Use API-issued temporary credentials for dynamic clients or static credentials for media servers and long-running infrastructure.

Production reliability

Make NAT traversal part of your infrastructure, not a launch-day surprise.

Direct peer-to-peer WebRTC can look healthy in testing and fail in the field. Xirsys helps teams plan for the percentage of sessions that need relay bandwidth and gives them dashboard visibility when network conditions change.

Reliable fallback

Add WebRTC NAT traversal without changing your app stack.

Use standard ICE configuration and let Xirsys handle the STUN/TURN connectivity layer underneath.