Make calls easily with our SIP to WebRTC gateway
ConnexCS offers a turnkey WebRTC solution to allow your existing SIP infrastructure to integrate with external WebRTC clients effortlessly. Allowing you to offer WebRTC as a Service.
We Transcode media between ulaw, alaw, g729, g722, g723, speex and opus. Easily connecting calls using different media types.
By only allowing the ConnexCS Gateway to your existing network, we will protect any unauthorised traffic from hitting your network.
Automatically deploy hundreds of WebRTC users with the help of our API.
Segment your WebRTC users into separate companies.
Should you need more than a gateway, ConnexCS Carrier grade switching and billing is all available to you.
Don’t have your own WebRTC Client? No worries!
Our Progressive Web Application (PWA) runs in all major browsers and can also be installed to mobile devices for ease of use.
Our user friendly portal makes it easy to provision & manage end user devices.
Allowing you to offer WebRTC as a Service.
Improvement is continuous. Our instant roll out solution ensures that all changes and upgrades are quickly made available to all for the end clients, ensuring that problem and features are deployed as fast as possible.
We can customise the WebPhone and bring any agent panel inside the application.
We also have software development kits (SDK’s) available to allow you to add your own developments.
If you have any ideas or customisation that you would like us to do, please get in touch.
We offer a range of services to help you and your business with your VoIP requirements.
From hour-long conference calls to full-day consultations, we can develop a solution that best meets your business need
