If you’re looking to integrate Zoom into your web application, you have two main options: the Meeting SDK and the Video SDK. While both run in the browser, support popular frameworks like React, Vue, and Angular, and use JWT for authentication, they’re designed for very different purposes.
When WebRTC was first introduced by Google over a decade ago, it came with the promise of simplicity. “Just drop in a little JavaScript and you’ve got video chat in the browser with no downloads necessary!” While that vision helped kickstart a wave of innovation in real-time
Voice assistants powered by real-time AI are increasingly being used to automate phone-based customer interactions. Whether for contact centers, internal help desks, or voice-driven workflows, a reliable architecture needs to support low-latency audio streaming, accurate speech-to-text (STT), intelligent response generation, and real-time speech synthesis. In this post,
Bringing WebRTC and SIP together is a powerful way to connect modern web applications with traditional phone systems. Whether you’re enabling voice and video in the browser, or linking your app to a PBX and SIP trunk, WebRTC SIP integration allows users to communicate across platforms without
