Session Initiation Protocol (SIP) and WebRTC are both essential technologies in the field of real-time communications, particularly for voice and video over IP. While they serve complementary roles, they operate differently and have distinct functionalities. In this post, we explore how to architect the integration of WebRTC
At Geekle’s Worldwide Software Architecture Summit ’22 November 15-16, 2022, our CTO Alberto Gonzalez will present a talk on architecting low latency WebRTC applications for scalability, high quality, and usability.
There are a variety of ways you can use traditional telephony in your WebRTC video app, as well as different architectures you can choose to support it from commercial to open source. Let’s explore the reasons you might want to integrate a dial-in or dial-out capability into your WebRTC video or audio application and look at a sample architecture.
WebRTC has growing applications by the day, but some may argue that it’s limited by its domain: the web. The world’s cellular network is more expansive and accessible than the web. Is it possible to connect the two for a true 100% coverage solution? The answer is
