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.

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 […]

The post WebRTC Plumbing with GStreamer appeared first on webrtcHacks.

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 […]

The post Probing WebRTC Bandwidth Probing – why and how in gcc appeared first on webrtcHacks.

We covered End-to-end encryption (E2EE) before, first back in 2020 when Zoom’s claims to do E2EE were demystified (not just by us; they later got fined $85m for this), followed by the quite exciting beta implementation of E2EE in Jitsi using Chromium’s Insertable Streams API. A bit later we had Matrix explain how their approach […]

The post End-to-End Encryption in WebRTC… 4 Years Later appeared first on webrtcHacks.

I am working on a personal Chrome Extension project where I need a way to convert a video file – like your standard mp4 – into a media stream, all within the browser. Adding a file as a src to a Video Element is easy enough. How hard could it be to convert a video […]

The post All the ways to send a video file over WebRTC appeared first on webrtcHacks.

Earlier last week a friend at Google reached out to me asking Does Meet do anything weird with scalabilityMode? Apparently, I am the go-to when it comes to Google Meet behaving weirdly :). Well, I do have a decade of history observing Meet’s implementation, so this makes some sense! It turned out that this was […]

The post The Hidden AV1 Gift in Google Meet appeared first on webrtcHacks.

Open Broadcast Studio or OBS is an extremely popular open-source program used for streaming to broadcast platforms and for local recording. WebRTC is the open-source real time video communications stack built into every modern browser and used by billions for their regular video communications needs. Somehow these two have not formally intersected – that is […]

The post WebRTC cracks the WHIP on OBS appeared first on webrtcHacks.

webrtcHacks celebrates our 10th birthday today 🎂. To commemorate this day, I’ll cover 2 topics here: Our new merch store Some stats and trends looking back on 10 years of posts We have the Merch In the early days of webrtcHacks, co-founder Reid Stidolph ordered a bunch of stickers which proved to be extremely popular. […]

The post 10 Years of webrtcHacks – merch and stats appeared first on webrtcHacks.

Explore the future of Real-Time Communications with WebrtcHacks as we delve into the use of WebCodecs and WebTransport as alternatives to WebRTC's RTCPeerConnection. This comprehensive blog post features interviews with industry experts, a review of potential WebCodecs+WebTransport architecture, and a discussion on real-time media processing challenges. We also examine performance measurements, hardware encoder issues, and the practicality of these new technologies.

The post WebCodecs, WebTransport, and the Future of WebRTC appeared first on webrtcHacks.

Here at webrtcHacks we are always exploring what’s next in the world of Real Time Communications. One area we have touched on a few times is the use of WebCodecs with WebTransport as an alternative to WebRTC’s RTCPeerConnection. There have been several recent experiments by Bernard Aboba – WebRTC & WebTransport Co-Chair and webrtcHacks regular, […]

The post Livestream this Friday: WebCodecs, WebTransport, and the Future of WebRTC appeared first on webrtcHacks.

Note: Chinese translation thanks to Xueyuan Jia and Xiaoqian Wu of the W3C. See the English version here.  W3C Web 技术标准专家 François Daoust 和 Dominique Hazaël-Massieux（Dom）先前与我们探讨了如何使用 WebCodecs 和 Streams 进行实时视频处理。那篇文章重点介绍了如何设置流水线以应付来自摄像头、WebRTC 流或其他来源的视频帧低延迟处理。演示了一些处理示例 — 改变颜色、覆盖图像，甚至是改变视频编解码。引用的其他用例还包括机器学习处理，例如添加虚拟背景。 今天，他们将重点讨论可用于进行实际视频处理的诸多技术选项。有很多技术用来读取和更改视频帧内的像素。他们全面回顾了当前基于 Web 的所有技术选项 — JavaScript、WebAssembly （wasm）、WebGPU、WebGL、WebCodecs、Web 神经网络（WebNN）和 WebTransport。其中一些技术已经存在一段时间，许多则是新出现的。 这是一篇关于与视频分析与操作的文章。感谢 François 和 Dominique 与我们分享他们的研究，测试 Web 上可用的进行视频处理的完整技术目录。 正文内容 视频帧处理选项 使用 JavaScript 像素格式 性能 其他考虑 使用 WebAssembly 演示代码 […]

The post Web 上的视频帧处理 – WebAssembly、WebGPU、WebGL、WebCodecs、WebNN 和 WebTransport appeared first on webrtcHacks.

There are a lot of options for reading and changing the pixels inside a video frame. In this post, W3C specialists François Daoust and Dominique Hazaël-Massieux (Dom) review every web-based option for processing video frames on the web available today - JavaScript, WebAssembly (wasm), WebGPU, WebGL, WebCodecs, Web Neural Networks (WebNN), and WebTransport.

The post Video Frame Processing on the Web – WebAssembly, WebGPU, WebGL, WebCodecs, WebNN, and WebTransport appeared first on webrtcHacks.

WebRTC used to be about capturing some media and sending it from Point A to Point B. Machine Learning has changed this. Now it is common to use ML to analyze and manipulate media in real time for things like virtual backgrounds, augmented reality, noise suppression, intelligent cropping, and much more. To better accommodate this […]

The post Real-Time Video Processing with WebCodecs and Streams: Processing Pipelines (Part 1) appeared first on webrtcHacks.

New coturn project leads Gustavo Garcia and Pavel Punsky give an update on the popular TURN server project, what's new in STUN and TURN standards, and the roadmap for the project

The post coturn: No Time to Die – Q&A with new project leads appeared first on webrtcHacks.

I interviewed mediasoup’s co-founder, Iñaki Baz Castillo, about how the project got started, what makes it different, their recent Rust support, and how he maintains a developer community there despite the project’s relative unapproachability. mediasoup was one of the second-generation Selective Forwarding Units (SFUs). This second generation emerged to incorporate different approaches or address different use cases a few years after the first generation of SFUs came to market. mediasoup was and is different. It is node.js-based, built as a library to be part of a serve app, and incorporated the Object-oriented approaches used by ORTC – the alternative spec to WebRTC at the time. Today, mediasoup is a popular SFU choice among skilled WebRTC developers. mediasoup’s low-level native means this skill is required.

The post Revealing mediasoup’s core ingredients: Q&A with Iñaki Baz Castillo appeared first on webrtcHacks.

WebRTC blackbox reverse engineering experts Gustavo and Fippo take a look at Cloudflare's new WebRTC implementation, how Cloudflare uses the new WebRTC-based streaming standards WHIP and WHEP, and the bold pronouncement that they can be a replacement to open source solutions.

The post How Cloudflare Glares at WebRTC with WHIP and WHEP appeared first on webrtcHacks.

WebRTC had its peaks during the pandemic, but how is it doing now? Did all those new projects die, putting the community back at pre-pandemic “normal” levels or is WebRTC still going strong? I built and analyzed a dataset from over a million GitHub’s events since 2019 to help answer are there many new WebRTC-related repos, how many new users is WebRTC attracting, is the community coding as much as it used to, how are new API's like Insertable Streams and WebCodecs doing?

The post Post-Peak WebRTC Developer Trends: An Open Source Analysis appeared first on webrtcHacks.

Balázs Kreith of the open-source WebRTC monitoring project, ObserveRTC shows how to calculate WebRTC latency - aka Round Trip Time (RTT) - in p2p scenarios and end-to-end across one or more with SFUs. WebRTC's getStats provides relatively easy access to RTT values, bu using those values in a real-world environment for accurate results is more difficult. He provides a step-by-step guide using some simple Docke examples that compute end-to-end RTT with a single SFU and in cascaded SFU environments.

The post Calculating True End-to-End RTT (Balázs Kreith) appeared first on webrtcHacks.

A full review and guide to all of the Jitsi Meet-related projects, services, and development options including self-install, using meet.jit.si, 8x8.vc, Jitsi as a Service (JaaS), the External iFrame API, lib-jitsi-meet, and the Jitsi React libraries among others.

The post The Ultimate Guide to Jitisi Meet and JaaS appeared first on webrtcHacks.

A very detailed look at the WebRTC implementations of Google Meet and Google Duo and how they compare using webrtc-internals and some reverse engineering.

The post Meet vs. Duo – 2 faces of Google’s WebRTC appeared first on webrtcHacks.