For our 57th episode of WebRTC Live, Arin Sime was joined by Anton Venema, CTO at LiveSwitch Inc for a deep dive into successfully scaling your WebRTC application in today’s technological landscape. They discussed the basics of scalability and media servers, optimizing for client vs. server efficiency, RTP packets and streams, bitrate management, the benefits of using a CPaaS to scale, and more. Watch it here!
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 number of strategies available for enabling WebRTC multi-party connections. The most simple choice is Mesh. But only if you don’t need to support more than 3-5 users on the same call and you don’t want a server in the middle. Our DevOps Engineer, Hector Zelaya, explains.
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.