Audio jitter buffers are required 101 introductory material for understanding VoIP. libWebRTC’s audio jitter buffer implementation – the one in Chromium – is known as NetEQ. NetEQ is anything but basic. This is good from a user perspective since real-life networks conditions are often challenging. However, this means NetEQ’s esoteric code is complex and difficult […]
libwebrtc
Probing WebRTC Bandwidth Probing – why and how in gcc
Maximizing stream quality on an imperfect network in real-time is a delicate balancing act. If you send too much information then will cause congestion and packet loss. If you send too little then your video quality (or audio) will look like garbage. But how much can you send? One of the techniques used to find […]
Let’s get better at fuzzing in 2019 – here’s how
Fuzzing is a Quality Assurance and security testing technique that provides unexpected, often random data to a program input to try to break it. Natalie Silvanovich from Google’s Project Zero team has had quite some fun fuzzing various different RTP implementations recently. She found vulnerabilities in: WebRTC — mostly issues in the RTP payload Facetime – a […]
Making WebRTC source building not suck (Alex Gouaillard)
One of WebRTC’s benefits is that the source to it is all open source. Building WebRTC from source provides you the ultimate flexibility to do what you want with the code, but it is also crazy difficult for all but the small few VoIP stack developers who have been dedicated to doing this for years. […]