WebRTC is a technology which allows voice and video communication directly through a web browser. WebRTC stands for Web Real Time Communications.
Although WebRTC has been around for quite some time, it’s popularity is growing. The driver for this is context base communication where you would like to enable many different ways of communicating and collaborating including voice, video and even screen-sharing. Improving customer interaction is one of the main drivers of introducing WebRTC communication.
Anyone with an internet browser can gain access to a VoIP call or video conference in no time, without the hassle of apps and applications. Therefore, end users can choose to work on virtually any device or operating system. With 5G now being rolled out with low latency and increased bandwidth it might be another trigger for further WebRTC growth.
Our SIP to WebRTC 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 webpages. The SIP to Web-RTC interface is provided as a function without need for any hardware or servers.
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.