For a long time, content creators and broadcasters had to choose between the immediacy of live streaming or the polish of edited, on-demand video. Today, innovations in browser APIs and real-time technologies—especially around WebRTC—are closing that gap. By leveraging powerful JavaScript APIs, modern libraries, and emerging protocols, it’s now possible to deliver feature-rich, low-latency video experiences that merge live interaction with the sophistication of post-production workflows.

This convergence of live and produced content represents one of the most exciting shifts in digital media. I recently explored these developments on WebRTC Live, joining three colleagues to discuss The Changing WebRTC Landscape. While you can watch the full episode and read the highlights on the WebRTC.ventures blog, this post will take you through some of the ways WebRTC is reshaping Video Publishing and Production for industries ranging from broadcast media to education to healthcare.

Standard Tools and Libraries for Rapid Development

On the application side, React remains a reliable foundation for many of our web projects, often paired with open source component libraries such as Chakra UI to streamline UI development. For real-time media handling, we frequently integrate Janus or a CPaaS platform to incorporate WebRTC features. This stack allows for fast development cycles, while recent advances in JavaScript APIs enable more complex real-time video features without reinventing the wheel.

Advanced Browser APIs for Real-Time Video Effects

The JavaScript APIs available in modern browsers make it easier to build compelling real-time integrations. For instance, the Insertable Streams API has evolved from an experimental feature to a stable one suitable for production. We’ve used it to implement video filters on live WebRTC calls—think virtual backgrounds or real-time overlays—giving creators a quick path to applying sophisticated effects that once required specialized media servers or post-production tools.

Broadcasting to Large Audiences: HLS and Low-Latency WebRTC

When it comes to scaling video to bigger audiences, we’ve found success using CDN-based approaches with HLS. This remains a cost-effective way to maintain high-quality streams while controlling latency. However, if minimizing delay is the priority, we explore WebRTC broadcasting. Protocols like WHIP (WebRTC-HTTP Ingestion Protocol) and WHEP (WebRTC-HTTP Egress Protocol) have taken center stage for ultra-low-latency streaming directly to browsers. While some solutions—such as cascading SFUs—remain experimental for massive scale, it’s exciting to see how open source communities are rapidly innovating in this space.

Real-Time Video Composition with Headless Browsers

For advanced features like live video composition, we frequently turn to headless browser setups using libraries such as Puppeteer. By running a browser instance without a visible UI, we can ingest and send WebRTC streams in real time, layering in graphics or mixing feeds for broadcast. Since this approach often relies on the same libraries used for front-end web development, it’s easy to iterate quickly and share knowledge with our clients—everyone’s already familiar with the standard JavaScript tooling and frameworks.

Lower-Level Video Processing and the Rise of WebAssembly

Not all video processing has to happen in a headless browser. Tools like FFmpeg and GStreamer remain invaluable for more specialized or heavier processing tasks. With WebAssembly, we can now run FFmpeg directly in the browser, unlocking client-side video editing and processing. This flexibility lets us choose between client and server processing, depending on performance needs or cost considerations. We also track other promising WebAssembly libraries—such as MP4Box from the GPAC project—to further refine in-browser video manipulation.

WHIP, WHEP, and the Transition from RTMP

RTMP has been a standard in video streaming for years and remains ubiquitous in many encoding devices. However, with the deprecation of Flash, browser-based RTMP solutions are no longer viable. Protocols like WHIP and WHEP combine the low latency of WebRTC with simpler HTTP-based signaling, making direct browser ingestion easier. Popular tools—OBS, GStreamer, and FFmpeg—have already adopted these protocols, indicating a broader shift away from RTMP. We’ve successfully integrated MediaMTX with WHIP/WHEP to minimize latency when bridging RTMP streams into WebRTC applications, but the ultimate goal is to skip RTMP altogether.

Looking Ahead: WebCodecs, WebTransport, and Node.js QUIC

WebCodecs, WebTransport, and WebAssembly promise even more advanced features for browser-based media in the coming years. WebCodecs opens up efficient video decoding and encoding directly in the browser, while WebTransport can enable new forms of low-latency data streaming over HTTP/3. Yet adoption can be slower on the server side, particularly with Node.js, because QUIC support relies on ongoing developments in OpenSSL. As Node.js and its ecosystem catch up, we’ll see more widespread use of HTTP/3-powered applications and tighter integrations for real-time video processing.

For further reading: What’s Next for WebRTC in 2025? A Look Ahead

Ready to Transform Your Video Production Workflow?

At WebRTC.ventures, we’re excited to be at the forefront of these developments, continually exploring new ways to enhance the content creation process and empower broadcasters with cutting-edge tools and techniques.

Whether you’re a broadcaster, content creator, or developer, now is the time to leverage these innovative solutions. Contact WebRTC.ventures today to help you implement state-of-the-art WebRTC integrations that will set your projects apart in an increasingly dynamic digital landscape.