Our experience

Hundreds of international projects successfully implemented

Over 20 years of Kamailio development experience in the team

Deep IT Infrastructure, IT Security and Project Management knowledge

Infrastructure hosted in Germany with high security and data privacy standards

WebRTC Gateway

Our WebRTC Gateway is an add-on product that brings browser- and app-based voice and video calling to your existing VoIP infrastructure, without replacing it.

It registers directly to your existing SIP server as a standard SIP endpoint, bridging WebRTC clients to your telephony landscape transparently.

Calls made from a web browser or mobile app reach your existing users, extensions, and PSTN connections just like any other SIP call.

WebRTC Gateway Platform

Overview of the GILAWA WebRTC Gateway, a Kamailio-based add-on that integrates seamlessly into your existing SIP landscape.

  • WebRTC-to-SIP signalling bridge based on a proven stable Kamailio version
  • Media transcoding and relay via RTPEngine — handles codec negotiation between WebRTC and SIP endpoints
  • Secure WebSocket (WSS) transport for browser and app clients
  • Registers to your existing SIP-server with your existing account, provides Caching and better performance, while no change to your current infrastructure needed
  • Support for voice and video telephony from browser or native app
  • TLS certificate provisioning via Let's Encrypt for secure signalling
  • Common codec support including OPUS (WebRTC), G.711, G.722, and VP8/VP9 for video
  • Local firewall and fail2ban configuration included
  • Delivery includes: deployment on one server, functional test verification, interconnection test with your SIP server
  • Optional additional services
    • TURN/STUN server setup for NAT traversal in restrictive network environments
    • REST API interface for provisioning and call control
    • Failover and high-availability configuration
    • Integration with your web or mobile application (SDK guidance)
    • Maintenance contract and updates

System requirements

  • Virtual machine at least 2 vCPUs, 4 GB RAM, 20 GB disk
  • Operating system Linux Debian 12, Debian 13, Ubuntu 22.04 or 24.04
  • One public IPv4 network address with forward DNS configured
  • Existing SIP server reachable from the gateway (SIP trunk or registration credentials)
  • One E-Mail address for certificates and system messages
  • SSH root access for installation and test support
  • Docker or Kubernetes deployment available as part of a project

Interested in this product?

We are looking forward to speak with you!

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