Interview with Danny Burns, Founder and CTO, Ceeblue
Ceeblue has been making waves in the real-time streaming space. Can you start by telling us what WebRTS is and what led to its development?
WebRTS is our latest real-time streaming framework, designed to deliver real-time video at scale with sub-500 ms end-to-end latency, while solving some of the biggest pain points in live streaming, such as network congestion.
Where did you come up with the idea for WebRTS?
At Ceeblue, we’ve integrated nearly every protocol you can imagine, and we have found that none of the current streaming protocols were really designed for live streaming at scale. For example, some low-latency protocols make very inefficient use of the origin, or were designed for communications and peer-to-peer connections instead of streaming, while others have components focused on VOD and are proprietary in nature.
WebRTS (which stands for Web Real-Time Streaming), on the other hand, was built from the ground up for real-time, sub-500ms end-to-end live streaming. It is compatible out of the box with existing CDNs and has no royalties to pay, making it both technically efficient as well as cost-effective. This is particularly relevant as real-time streaming is now a growing priority for industries that need scalable, seamless content delivery, from major sports broadcasters to online betting platforms.
What makes WebRTS unique is that it challenges the notion that WebRTC, and to a lesser extent HESP and MoQ, are the best solutions for mass-scale real-time streaming. While WebRTC has dominated this space, it transmits its data over UDP. WebRTS, on the other hand, can now support real-time streaming over HTTP-based delivery methods, leveraging TCP while still matching WebRTC’s performance for low-latency deployments. Our real-world deployments have shown WebRTS achieving sub-500ms end-to-end latencies even under suboptimal network conditions.
What specific problems does WebRTS solve compared to existing real-time streaming technologies?
There are a few major ones. First, WebRTS offers true real-time latency without compromising video quality. A significant innovation we’ve introduced is partially reliable streaming, which balances continuity with real-time delivery. WebRTS can be configured to prioritize either stream continuity using TCP’s retransmission and buffering mechanisms or, alternatively, implement selective frame skipping to maintain low latency even in suboptimal network conditions.
It’s quite amazing. WebRTS’s seamless frame-skipping mechanisms maintain the stability and quality of the stream, even when the network is throttled to a bitrate that is below the bitrate of the lowest available rendition.
Second, WebRTS is built for traditional CDNs, and doesn’t require special configurations or specialty delivery infrastructure (as WebRTC does). This, along with the fact that it is an open-source framework with no royalties, brings the total cost of ownership of a WebRTS solution far below that of other real-time technologies.
Finally, it’s extremely efficient—our new containerless format uses 5% less bandwidth than CMAF, and is incredibly light on the origin, which reduces cost and complexity, improving scalability.
That sounds like a major leap forward. Can you break down some of WebRTS’s key technical features?
Absolutely. A few standout features include:
- Partial Reliability Mode: This allows WebRTS to drop non essential frames when needed, ensuring smooth playback even under poor network conditions.
- Advanced Congestion Control: Our predictive buffer state machine and adaptive playback mechanisms optimize streaming quality dynamically.
- Compatibility: WebRTS is codec-, and transport protocol agnostic, making it easy to integrate with existing workflows.
- CDN Compatibility: Compatible with traditional CDNs, reducing delivery cost and workflow complexity.
- Lower Origin Load: By reducing queries and using an efficient demuxing system, WebRTS puts less strain on origin servers, lowering infrastructure costs.
- Adaptive Streaming with MBR: WebRTS dynamically reduces stream bitrate and resolution when network congestion is detected, ensuring consistent playback before resorting to frame skipping.
- Support for DRM: Unlike WebRTC, WebRTS is fully compatible with the major digital rights management solutions, making it more viable for commercial streaming applications. (We’ll be showing off real-time DRM at EZDRM’s booth at NAB Las Vegas this year.)
Who stands to benefit the most from WebRTS?
WebRTS is a game-changer for any industry where ultra-low latency is critical. Sports OTT broadcasting benefits from real-time delivery, as it eliminates the “spoiler effect” from social media.
iGaming companies love WebRTS because their customers demand immediate interactivity and the highest image quality and resilience. There is a demonstrable correlation between quality of experience and revenue in this industry.
Sports betting platforms appreciate the real-time latencies, which permit a larger betting window, and therefore increased revenue. Real-time latencies also eliminate the possibility of cheating, making sports betting platforms far more secure. A huge consideration for these platforms is WebRTS’s compatibility with their current CDN providers, which allows for far more cost-efficient scalability than specialty WebRTC CDNs.
Live auctions can operate seamlessly with remote bidders.
Even corporate communications and town halls see improved engagement thanks to higher-quality, buffer-free video streams. The fact that WebRTS easily traverses corporate networks, unlike WebRTC, is a huge benefit that Ceeblue’s corporate events customers are taking advantage of.
Can you share some real-world use cases where WebRTS is making an impact?
We’re already seeing strong interest from live sports, live events, and betting platforms, where real-time interaction is key. For example, in the case of iGaming operators, WebRTS increases gross-gaming revenue (GGR). Many platforms have long used technologies that provide 2.5 - 6 seconds of latency. But with WebRTS they achieve sub-500ms latencies, increasing engagement and betting frequency.
Looking ahead, what’s next for WebRTS and Ceeblue?
We’re continuing to refine WebRTS and expand its capabilities. We’re working on additional optimizations to further reduce bandwidth usage and enhance playback stability under extreme network conditions. We’re also collaborating with industry partners to integrate WebRTS into more commercial streaming platforms. Ultimately, our goal is to make real-time streaming more efficient, scalable, and accessible for businesses of all sizes.
In the near future, we are studying the possibility of extending WebRTS beyond TCP and into UDP-based transport with support for switching from one to the other while streaming. This will make WebRTS even more flexible as new real-time streaming standards evolve.
Sounds exciting! Where can people learn more about WebRTS?
They can visit our official WebRTS page at ceeblue.net/webrts.
We’re always happy to connect and discuss how WebRTS can fit into new and existing workflows!
We will soon be spearheading an open source initiative that will support many community tools for companies and developers that want to learn more about and contribute to the WebRTS project. Stay tuned!
Thanks, Danny! We appreciate your time and insights.
My pleasure! Thanks for having me.
Danny Burns has spent over 30 years in the IT arena, accruing Technical Support & Infrastructure experience. He has been a part of Senior Management of teams of technical engineers in large customer-facing IT support roles, has run Infrastructure and Operations programs, has led Network Design Operations for large multi-national 7x24 blue chip operations, has designed Infrastructure for Hotels and Broadcasters, and has a strong technical background coupled with sound Operational Process experience.
http://ceeblue.net/
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