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!
For our 56th episode of WebRTC Live, Arin Sime was joined by Wonder Co-Founder Leonard Witteler. Leonard and his team have used WebRTC to build a virtual space for groups to meet, talk, exchange ideas, and work together that is so much more than a networking tool for virtual business conferences. He discussed the evolution of the technology and architectures that made Wonder possible and their use of multiple CPaaS providers, as well as MediaSoup, to handle their varied use cases. Watch it here!
For our 55th episode of WebRTC Live, Arin Sime was joined by Manik Sachdeva from Around. Manik provided insight into the features, talent, technologies, and tech stacks that Around is employing to keep us connected in the new normal of collaborative meetings and hybrid work environments. They discussed noise and echo suppression, audio-only meetings, latency, scaling, UI, load balancing, the decision to use Electron, Chromium, and Mediasoup, and much more. Watch it here!
For our 54th episode of WebRTC Live, Arin Sime was joined by Sergio Garcia Murillo, founder and main developer for Meedoze technology, CoSMo’s Media Server Tech Lead, and Millicast’s Principal Engineer and Solution Architect, to explore enabling the next generation of live video architectures with Real-Time AV1 SVC.
