WebRTC SIP Gateway
TeleFinity WebRTC-SIP Gateway allows your website visitors to place calls directly to your existing Call Manager/Call Center or traditional PBX from anywhere at zero cost. It can turn the users’ browser in desktop computers and mobile devices into a secure phone terminal.
TeleFinity WebRTC to SIP Gateway* is available on the cloud as well as on-premises. Our cloud WebRTC to SIP Gateway simplifies the implementation and speeds it up in less than 10 minutes. At the same time, the on-premises are available when your organizational policy requests it to be implemented within the organization’s data center.
All you need is to copy-paste a little piece of JavaScript code into your website. The JavaScript code will be generated and shared with you once the service is activated.
WebRTC is a technology that enables secure Real-Time Communication (RTC) capabilities in browsers and mobile platforms.
*WebRTC-SIP Gateway or SIP WebRTC Gateway are the same. it converts the WebRTC traffic into SIP and vice versa
Try It
Starter | Standard | Enterprise | |
Concurrent Calls (Channel) | 2+ | 5+ | 10+ |
Single Call Button | Yes | Yes | Yes |
Multiple Call Buttons | – | Yes | Yes |
Customized Call Buttons* | – | – | Yes |
DTMF Capture | Yes | Yes | Yes |
Dashboard | Yes | Yes | Yes |
CDR | Yes | Yes | Yes |
WordPress Plugin | Yes | Yes | Yes |
Call Recording** | – | Yes | Yes |
Location Restriction | – | – | Yes |
Price | 20 USD /Channel /Month | 27 USD /Channel /Month | 35 USD /Channel /Month |