guides and information for WebRTC developers| webrtcHacks
Explore WebRTC’s NetEQ jitter buffer with Meta’s Fengdeng Lyu. This post starts with concepts such as jitter, loss, playout, and concealment. It then reviews the Chromium libWebRTC implementation and various algorithms in detail.| webrtcHacks
What SDP munging is, where in your WebRTC code you're allowed to do it and why you shouldn't do it.| webrtcHacks
OpenAI is utilizing WebRTC for its Realtime API! Even better, webrtcHacks friend and Pion founder Sean DuBois helped to develop it and agreed to a Q&A about the implementation. It is not often a massive WebRTC use case like this emerges so rapidly. In addition, Sean was extremely transparent about his work at OpenAI. In […]| webrtcHacks
How to measure OpenAI's response latency using WebRTC and VoIP tools with an analysis of the results| webrtcHacks
Walkthrough on how to use the OpenAI Realtime API with WebRTC including tips on WebRTC, overall flow, data channel messaging, and functions| webrtcHacks
Data analysis of millions of GitHub events to track developer activity and tech trends driving the evolution of open-source WebRTC| webrtcHacks
Learn how to capture and replay video streams for effective debugging in this step-by-step guide on WebRTC's video_replay tool.| webrtcHacks
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 […] The post Power-up getStats for Client Monitoring appeared first on webrtcHacks.| webrtcHacks
overview, history, and future of GStreamer's support of WebRTC and its many plug-ins for building WebRTC into media pipelines| webrtcHacks
Why bandwidth probing is needed, Google’s Congestion control (GCC) implementation, and look at bandwidth estimation (BWE) debugging.| webrtcHacks
Philipp "Fippo" Hancke examines End-to-End Encryption (E2EE) adoption and standardization progress in WebRTC.| webrtcHacks
Authored by Philipp Hancke, investigation prompted by unusual behavior in Google Meet's handling of the scalabilityMode statistic. Hancke reveals the use of AV1 during pre-call stages, AV1 with VP9 SVC, and provides data on the advantages of AV1 screen sharing.| webrtcHacks
Review and analysis of the WebRTC WHIP implementation added to OBS, how it compares to RTMP, and what to expect next.| webrtcHacks