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[10 years of] guides and information for WebRTC developers

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Search Results for: SFU

WebRTC cracks the WHIP on OBS

Chad Hart · August 22, 2023 · 2 Comments

A logo of a company Description automatically generated with medium confidence

Open Broadcaster Software – Studio or OBS Studio 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 […]

WebCodecs, WebTransport, and the Future of WebRTC

Chad Hart · July 18, 2023 · 1 Comment

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.

Revealing mediasoup’s core ingredients: Q&A with Iñaki Baz Castillo

Chad Hart · November 16, 2022 · 1 Comment

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.

How Cloudflare Glares at WebRTC with WHIP and WHEP

Gustavo Garcia · October 25, 2022 · Leave a Comment

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.

Calculating True End-to-End RTT (Balázs Kreith)

Balázs Kreith · July 10, 2022 · 2 Comments

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.

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