Bugs exist. WebRTC is not an exception. Debugging WebRTC applications can be particularly challenging, but not impossible. Arin’s guest for WebRTC Live Episode 49 was our own Senior WebRTC Engineer and Developer Evangelist, Germán Goldenstein. They discuss tools and methodologies for debugging a WebRTC call.
Building in a pre-call test is the most important thing you can do to make your live video application successful. Since we can’t always control call quality, doing a network test, assessing video and audio quality, performing a video and microphone test, and offering a back up plan before the call begins is the best way to ensure customer satisfaction.
The Amazon Chime SDK allows developers to quickly add messaging, audio, video, and screen sharing capabilities with the Amazon Web Services infrastructure as its backbone. Let’s explore its core functionality and try building a simple videoconferencing app.
There are many great open source WebRTC media servers out there. But Janus’ great performance, small footprint, and active open source repository and community make it a popular choice for developers looking to use the latest supported WebRTC functionalities. Alberto Gonzalez takes Janus out for a spin to build a test video conference app.
