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
WebRTC & SIP: The Demo! WebRTC and SIP are two of the most important technologies in today’s real-time communication ecosystem. Session Initiation Protocol (SIP) is heavily used in VoIP technology; webRTC is used for browsers, mobile devices and native communication capabilities without additional software plugins. In this