What is WebRTC-to-SIP?

WebRTC-to-SIP trunking

Receive Audio Calls from Web Browsers or Mobile Devices in your Contact Center.


Video Gateways (WebRTC) developed by Interactive Powers include a SPLIT module which enable to control all media streams (video - audio - data) on any communication between users and a SIP softswitch or PBX/ACD. This development is based on SIP and WebRTC peers communications for the integration of voice and data.

Connect a Web Widget for any existing Call Centers.

How does WebRTC-to-SIP work?

WebRTC-to-SIP (Trunking) enables to convert Video Real-Time Communications from any Web Browsers or Mobile Devices into a standard SIP trunk for your Call Center.

WebRTC-to-SIP Diagram

Step 1. Audio Streaming & Signaling

Step 2. WebRTC to SIP Converting

Step 3. DTMF enabled

Step 4. SIP headers overload

What are Web Browsers supported?

All modern Web Browsers (WebRTC compliant) are supported like:

  • Chrome
  • Firefox
  • MS Edge
  • Apple Safari
  • Opera

What are Mobile OS supported?

Leading Market Mobile OS are compliant without any plugin:

  • Apple iOS (iPhone, iPad…)
  • Android OS (Smartphones, Kiosks and Tablets)

What ACD/PBX can be connected?

You can connect any kind of SIP compliant PBX or ACD (Automatic Call Distribution) solution like and not limited to:

  • Altitude Software
  • Enghouse / Presence
  • Genesys PureCloud
  • Genesys Engage
  • Motion XCally
  • Cisco CM
  • ICR Evolution
  • Avaya PBX / CC
  • Audiocodes
  • Kolmisoft
  • InConcert
  • 3CX PBX / Call Center
  • Asterisk PBX

Need to add it right now?

Please contact our Sales Support team for pricing and more information.


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