Live video should be fast, scalable, secure, and simple. That’s the promise of WebRTS (Web Real-Time Streaming).
Born out of years of hands-on experience integrating real-time streaming technologies that scale, WebRTS is Ceeblue’s answer to a problem that has plagued live streaming for years: how do you deliver sub-second video reliably over the web, without compromising on scale, security, or viewer experience?
The answer: We rethought the delivery model to fix real-world, real-time problems, but without having to reinvent the wheel.
Real-Time That Actually Works Everywhere
WebRTS delivers live video with sub-500 millisecond glass-to-glass latency without needing to install a plugin, use a specialty CDN, or fire up a proprietary player.
This is web-native real-time. WebRTS live video plays directly in the browser using standard HTML5 players, including video.js, using our open client library. There’s no proprietary lock-in and no outdated, 90’s-style installation requirements. Just real-time playback, on any device, wherever your audience happens to be.
It scales like HLS or DASH, but it delivers like WebRTC.
Built for Scale: Over Any CDN
WebRTS doesn’t need special infrastructure or custom delivery networks. It works with traditional CDNs. It scales like HLS or DASH, but it delivers like WebRTC. That means you get real-time delivery without sacrificing the reliability, reach, and cost-efficiency of standard delivery pipelines.
WebRTS is also format-agnostic. You’re not locked into a single transport or codec or vendor. You have the freedom to choose the best path for your pipeline.
What Is Quality, Really?
One of the first internal debates we had while building WebRTS was a deceptively simple question: What does “quality” actually mean when it comes to live video?
Ask a network engineer, and you’re more than likely to get an answer based on reliability: quality means every packet delivered, every frame accounted for, in perfect sequence. No drops, no skips.
But ask a viewer and you’ll get something very different. For viewers, quality means the video doesn’t freeze during a goal. The audio doesn’t cut out in the middle of a speech. The stream doesn’t collapse the moment they enter a tunnel or dip into 3G. There are no artifacts. If a frame gets dropped while they’re blinking, who cares?
Both perspectives are valid. And with WebRTS, they don’t have to be at odds.
WebRTS includes an optional adaptive mode that intelligently balances both definitions of quality. When conditions are stable, it operates in fully reliable mode, delivering every frame with surgical precision. But when the network falters—when congestion spikes, when packets get lost—WebRTS can shift gears. It employs seamless Adaptive Live-Point Recovery, prioritizing audio continuity, and using smart congestion control to keep playback fluid without sacrificing the Quality of Experience (QoE).
The result: a method for delivering live, real-time video that satisfies both the network engineer’s need for precision and the viewer’s demand for uninterrupted immersion.
Yes, It Uses TCP—And No, That’s Not a Problem
One of the biggest misconceptions about WebRTS is in its use of TCP-based delivery. Critics often point to TCP’s susceptibility to head-of-line blocking as a fatal flaw for real-time video. But WebRTS fixes that.
WebRTS incorporates intelligent congestion control and adaptive playback strategies that actively avoid head-of-line blocking—even over TCP. Depending on the transport layer, it uses a combination of a number of methodologies to bypass problematic segments and maintain the live point without stalling or buffering. No queuing, buffering, or head-of-line blocking, even if the available bandwidth is below the bitrate of the lowest-quality rendition.
Playback remains smooth even when the network isn’t, effectively solving the last-mile problem.
TCP has just flashed forward into what modern streaming should be: real-time that’s actually resilient to real-world network congestion.
Real-time video that satisfies both the network engineer’s need for precision and the viewer’s demand for uninterrupted immersion.
Designed for the Real World, Not the Lab
WebRTS was built for the challenges that broadcasters actually face: real eyeballs on messy networks, watching high-stakes events like live sports, auctions, or betting streams. When conditions get rough—up to a very respectable threshold—WebRTS doesn’t crash or freeze. Its adaptive mode adapts. Seamlessly.
In controlled environments, sure, many protocols perform fine. But WebRTS shines in uncontrolled, unpredictable, large-scale deployments. That’s why it’s already in use in demanding verticals like corporate communications.
Corporate communications are a particularly good fit, as WebRTS slips by corporate firewalls’ port blocking with ease, and is currently in production with eCDNs like Quanteec. These events are often high-stakes one-off productions, with important stakeholders’ perceptions on the line. QoE is of the utmost importance, making WebRTS the ideal solution.
Cost-Effective by Design
WebRTS wasn’t just built for performance, it was built to be efficient, both technically and economically.
At scale, the hidden costs of real-time video can be enormous: bloated bandwidth usage, excessive load on origin servers, and tightly coupled infrastructure that’s expensive to maintain or scale. WebRTS addresses all of that head-on.
Thanks to its new containerless format, WebRTS uses up to 5% less bandwidth than CMAF, while also reducing overhead at the origin. That’s not just good engineering—it’s a direct cost savings for any broadcaster running high-volume live events.
It’s also designed to work seamlessly over standard CDN infrastructure, which means you’re not locked into expensive, specialized delivery services. Just plug it into your existing stack and go.
In short: WebRTS does more, with less.
Security Without the Latency
Most real-time protocols struggle with digital rights management (DRM). WebRTS doesn’t. It was designed with premium content protection in mind, supporting Widevine, PlayReady, and FairPlay right out of the box. You get true sub-second latency and full DRM support. No trade-offs.
WebRTS is also tokenization-friendly, including Bring Your Own Token (BYOT) support, public-to-private-to-public switching, individual viewer callbacks, customizable session limitations, and real-time session deauthorization and on-the-fly token deletion, making it ideal for broadcasters and platforms with complex entitlement systems.
For too long, developers have had to choose between speed, scale, and simplicity.
Open Where It Matters
While the WebRTS framework itself includes some proprietary server-side components (for the time being), the client library—which contains most of the heavy lifting, intelligence, and logic—is fully open source under the AGPL v3.0 license.
We believe in interoperability, transparency, and building something the community can contribute to and grow with. That’s why we’ll be launching a WebRTS community portal dedicated to the open-source library, documentation, resources, best practices, tutorials, and developer community.
You can use WebRTS with your own infrastructure, your preferred CDN, and your customized workflows. The client library can be integrated easily, there’s a video.js plugin available, and we provide a user-friendly API. We’re here to accelerate innovation, not control it.
Why This Matters
Live video enables some of the most valuable and QoE-sensitive moments online: OTT live sports, betting, iGaming, live shopping, live casinos, eLearning, corporate communications, and more. But for too long, developers have had to choose between speed, scale, and simplicity.
WebRTS changes that. It’s a drop-in upgrade for real-time delivery:
- Sub-second latency
- Full DRM support plus tokenization
- Standard CDN compatibility
- No client install or app required
- Open-source client
- Resilient in the face of network congestion
- Cost-effective and Efficient
In short: it works. At scale. In the real world.
Want to see it in action? Learn more about WebRTS or contact Ceeblue.
Keep an eye on this space for the upcoming launch of the WebRTS open-source community…
