WebRTC is more popular than ever, and is supported by more browsers than ever. But that doesn't make building with WebRTC any easier!
For our 74th episode of WebRTC Live, Arin was joined by our WebRTC.ventures CTO Alberto Gonzalez to discuss the challenges inherent in architecting low latency WebRTC applications including complex, scalable and high availability media servers; stateful system complexities; and server provisioning.
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!
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.
