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.
There are many different ways to handle the video and audio streams in your WebRTC application. In this post, Arin Sime considers the line of decisions around open source media servers. First, whether to use one at all, as opposed to pure peer-to-peer architecture. Then, whether to choose an SFU or an MCU. The answers, as they usually do, rest in your use case.
