Since our “Real-time communication testing evolution with WebRTC 1.0” article was accepted for publication by IEEE, not many new scientific papers have been published on WebRTC testing apart from [1] and [2] ([3] is also available, but without indication wether it has been peer-reviewed and accepted for publication or not), but a lot has happen with KITE.
Beyond being the official testing tool for webrtc.org, and adding tests for multistream and simulcast to better cover the last pieces of the WebRTC 1.0 spec, support for mobile browsers and mobile app testing, as well as for Electron apps testing have been added. KITE can now support up to 20 clients configurations, making it the most complete and most versatile #webrtc testing tool known to date.
Beyond the Interoperability Mode, a Load Testing mode has been added to KITE, specifically to stress test webrtc servers and infrastructures, with a capacity of 1 millions users. It has been tested in production by several customers in a one-to-many streaming environment.
Of course, you can have CoSMo run KITE for you (hosted and managed), or you can run it yourself, on premises. More details in this Post.
Month: May 2018
Streaming protocols and ultra-low latency including #webrtc
In their latest blog post, Wowza is doing a great job at explaining in simple words latency, and the use cases that could benefit for having under 500ms, a.k.a. “real-time”, latency. However, the section about streaming protocol is somehow confusing me. This blog post is an attempt to put those protocols back into perspective to have a fair comparison.