Debugging WebRTC media issues, especially video, often requires access to the unencrypted RTP payloads. We talked about this back in 2017 already and had a great blog post on using the libWebRTC “video_replay” tool. While that post has aged remarkably well, video_replay has improved significantly, in particular since it is now possible to create the […]
Upcoming Livestream 10-Dec: 2024 WebRTC in Open Source Review
Tuesday, December 10 @ 5PM CET / 11AM ET / 8AM PT / 16:00 UTC Join Chad Hart, Editor of webrtcHacks, for an analysis of WebRTC trends in GitHub, StackOverflow, and other open-source communities. Leveraging advanced quantitative analysis techniques, this talk examines millions of GitHub events and developer activity data to uncover key trends […]
Power-up getStats for Client Monitoring
WebRTC’s peer connection includes a getStats method that provides a variety of low-level statistics. Basic apps don’t really need to worry about these stats but many more advanced WebRTC apps use getStats for passive monitoring and even to make active changes. Extracting meaning from the getStats data is not all that straightforward. Luckily return author […]
WebRTC Plumbing with GStreamer
GStreamer is one of the oldest and most established libraries for handling media. As a core media handling element in Linux and WebKit that as launched near the turn of the century, it is not surprising that many early WebRTC projects use various pieces of it. Today, GStreamer has expanded options for helping developers plumb […]
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 […]