Fixing an existing WebRTC application is not as much fun as building a new one, but it’s often necessary. Our team typically employs a combination of four fixes: re-architecting the media server or choosing a new CPaaS, solving compounding bugs, re-architecting the application, and improving the UX and error handling.
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.
When choosing your WebRTC application architecture, there are trade-offs between going with a native application against the standard, an open source media server, or using a commercial CPaaS platform. There are, however, some simple rules you can follow to lead you to the right decision for your use case.
In a previous post, we discussed the advent of Microsoft’s new Communication-Platform-as-a-Service (CPaaS) platform, Azure Communication Services. Today, we will cover the basic configuration and elements you need to set up an Azure Communication Service app and get our hands dirty building a Group Video Calling App that can handle up to 50 participants.
