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.
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!
The demand for real-time video applications has never been greater. If you can’t wait for our expert team to free up or if you are simply low on funding, Arin Sime has compiled a list of resources to help you learn more about WebRTC development on your own.
