What is WebRTC-to-SIP?
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.
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.
Interactive Powers - Streamline your business communications