WebRTC to SIP Delivery Network

Voxbone offers support for WebRTC calling via its global VoIP backbone, which enables customers to easily deploy high quality and secure WebRTC services on existing infrastructure, without complex software development or network investment.

WebRTC (Web Real-Time Communication) is an API that supports voice calling, video, chat, and P2P file sharing from a browser without plugins. This service is especially suitable for audio conferencing providers, contact centers, or unified communications, but is also beneficial to any organization interested in opening up their communication channels to the web with voice calling capabilities.

Excellent call quality over Voxbone's SIP trunk

With its WebRTC to SIP gateway, Voxbone introduces two capabilities to WebRTC: Firstly, Voxbone routes WebRTC calls over its private global backbone instead of the public Internet, adding high levels of quality of service and security. Secondly, calls are delivered over a regular SIP trunk, allowing Voxbone customers to start using WebRTC without the need for additional investment in network infrastructure.

As a Voxbone customer you gain access to Voxbone’s WebRTC Javascript library and authentication service, which enables you to rapidly develop your web applications and connect these applications to the nearest Voxbone PoP over the shortest Internet path.

  • Cloud WebRTC-to-SIP gateway
  • No WebRTC infrastructure needed
  • Authentication service for TDoS protection
  • Global private backbone for QoS
  • Javascript library for rapid client development
  • Configuration and activation via web portal and API
WebRTC to SIP Framework