How does XirSys help me?

XirSys provides the server-side infrastructure required to deploy your WebRTC applications and services in a production-grade, expert supported, reliable, fault-tolerant environment.


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Requirements

WebRTC requires server-side functionality covering the following four items, all of which are provided by the XirSys platform:

User discovery & communication

Signaling

NAT/firewall traversal

Relay servers

Real-time communication

The STUN protocol and its extension TURN are used by the ICE framework to enable WebRTC peer connection to handle NAT traversal and other network conditions.

ICE is a framework for connecting peers, such as two video chat clients. Initially, ICE tries to connect peers directly to each other with the lowest possible latency, via UDP. If these initial STUN attempts fail because of restrictive NATs or firewalls, ICE will use an intermediary (relay) TURN server.

With these server requirements it is critical for production-quality services and applications to deploy on a WebRTC server infrastructure with guaranteed performance, reliability, security, scalability, and this is exactly what XirSys provides.

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Developer Tools

We provide the tools you need to help you build and deploy your WebRTC applications including API's, quick start guides, demos and client components. And if you need more, just contact us, we're happy to help.

  • Initiate a WebRTC connection with 3 easy steps, view our Quick Start Guide
  • Establish and manage WebRTC sessions and security with the XirSys API
  • Use any third-party API with XirSys, see how to use SimpleWebRTC's API
  • Check out our Developer Page for more tools