WebRTC enable your legacy communication infrastructure without need to manage any hardware or software.

Web enable your legacy communication infrastructure

Modern software is built utilizing the power of public cloud infrastructure and designed around a set of microservices. In todays fragmented communication world many applications have a monolithic platform structure with limited opportunities to communicate over voice and audio, outside their own network or apps.

Our SIP to Web-RTC gateway, provided as a function, allows Enterprise applications and platforms supporting SIP to establish peer-to-peer real time communication such as voice and video to web-pages.  The SIP to Web-RTC interface is provided as a function without need for any hardware or servers

Bridging WebRTC and SIP worlds

Direct your modern WebRTC communication application towards the WebRTC gateway function and we do the bridging back to the legacy SIP world.

Towards the legacy platform the WebRTC application will look like an ordinary SIP endpoint. The function will proxy SIP Register and Invite requests and bridge the differences required by the WebRTC enabled browsers.

Use case examples

The function allows web based applications such as:

  • Web softphone and dialer applications
  • Web contact center applications
  • Web collaboration services

The web application hooks in with existing backend communication applications and ordinary phone systems and trunk services.

Painless integration

The function enables easy integration with WebRTC applications using SIP over WebSockets. All that need to be done is to add a route header pointing back to legacy system.

The function takes care of the rest of complexity seamlessly! It automatically scales to meet your capacity needs. And you don’t have to worry about server management and availability.

Specification

WebRTC to SIP gateway function

  • Bridges from SIP/Websocket to SIP over UDP, TCP or TLS
  • Bridges from SRTP to RTP media
  • Modifies SDP to enable interworking between legacy systems and WebRTC enabled browsers
  • Integration via SIP Route header
  • Supports voice and video calling

Browser support:

  • Chrome
  • Safari
  • Firefox

Codec support:

  • G.711
  • OPUS
  • VP8
  • H.264

Protocol support:

  • SIP
  • WebSocket
  • RTP
  • SRTP
  • DTLS
  • ICE