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.
Real Time Weekly #252: October 22, 2018 Welcome to Real Time Weekly! Hello everyone! This week we’ve got a new announcement from Plivo about their new multiple incoming calls support, an overview to the most popular open source WebRTC media servers from Linagora, and tutorials from Pusher, Pubnub,
Webphones are web-based applications that allow you to make phone calls directly in your Internet browser like Firefox, Chrome, Safari and many more. Webphones can make and receive calls from traditional phones (PSTN, Public Switch Telephone Network). The Real-Time Communications (RTC) capabilities also allows audio, video, messaging
