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Finally, a CDN-scale, sub-second streaming solution

The live streaming industry has been searching for a lower-than-ultra-low-latency (let’s call it “real-time”) media streaming solution. It’s a must have for betting, bidding, interactive communications, VR, sports, gaming, etc., and a nice to have for everyone else. While WebRTC is known to provide real-time latency, it could not be used on Apple devices and would not scale beyond a couple of thousand viewers, far from the 100+ millions of TV viewers of the Super Bowl in 2018, and generally speaking, the millions of viewers you would expect to sustain with CDNs today. CoSMo and Xirsys came together to design Millicast, a solution that would match current CDNs in terms of price, reach (Apple devices), quality and scale, while keeping the advantage of WebRTC sub-500 milliseconds latency. With patches pending against major browsers, and smart routing and packet loss proprietary protocols running in the back-end, the solution is unique and protected.

Real-time, 1-to-many video streaming is focused on optimizing 5 things: Reach, Quality, Cost, Scale and Latency, in this order.


Until recently, HTTP-based solutions were the only ones which could reach Apple devices, however, Apple began supporting WebRTC in Safari 11, and support for WebRTC in Apple WebView Framework is scheduled for iOS 12 in October 2018, which will bring WebRTC support to all web browser iOS apps (3). This provides a low latency alternative solution to reach all devices without the need for a native app, plug-in or special player.


Quality is important, and usually refers to resolution. It creates a never ending course for a better codec, “better” meaning codecs that can encode higher resolution streams, and codecs with higher compression ratio to stream the same quality with less or equal bandwidth. In some cases, quality means resilience to bad network conditions. All HTTP-based solutions address this problem by buffering, which increases latency. WebRTC utilizes the RTCP protocol with mandatory encryption and provides a mechanism to probe the available bandwidth on both sides, and adapt the encoder settings to bandwidth fluctuations or network conditions, in real-time.


With pre-recorded content, one can leverage an internet HTTP caching infrastructure (5) to lower the cost, but this is not possible with live and real-time content. However, the cost of bandwidth from most major providers is now on par with CDNs, bringing the cost for direct and file-based distribution on par for real-time media.


Scalability has always been the strong point of HTTP-based, file based content which also allows for the reuse of existing CDN systems for fast, reliable, and cheap distribution, at scale. Basically, if you wanted to reach audiences of millions of viewers, HTTP was the best option. Moreover, not being used in production for stream broadcasting, and lacking the tools to stress test to millions of users, everybody assumed that WebRTC was only capable of 1,000 streams, period. A recent blog post by Wowza (2) wrote this assumption as a statement. Xirsys and CoSMo came together to bring an intelligent autoscaling and cascading WebRTC solution that has been load tested with KITE, the Google-sponsored and industry-standard test engine for WebRTC. Now we have clear data that WebRTC scalability is not only beyond the 1,000 streams, but in fact on par with more traditional streaming solutions.


The HLS standard allows you to reach Apple devices, at tens of seconds of latency. One can try to optimize it, but by design, file-based, HTTP-based designs are slow, and must add delay to manage quality. This status quo that the entire industry has adopted can only be maintained as long as there is no alternative. With the Xirsys and CoSMo breakthrough on the WebRTC scalability front, you can now get Real-Time, sub-second latency at the same cost, quality and scalability level you were used to. Who can now afford to compromise, and why should they?

— — —

About Xirsys

Experts in custom Audio and Video infrastructure architecture, design and management, Xirsys has been serving the streaming community as “Influxis” for more than 15 years and counting. Xirsys is one of the few original pioneers of WebRTC infrastructure on-demand with their TURN Server offerings, and have since extended their offer to custom installation and hosting of practically all the possible WebRTC servers in the world: Jitsi video bridge, Janus video room, Medooze, LiveSwitch, Kurento media server, etc. With footprint in 11 data centers across the world, they are the most versatile WebRTC hosting solution on the market.

— — —

About CoSMo

Founded by WebRTC veterans with PhDs, the CoSMo team are well known contributors to the evolution of the WebRTC standard itself, and the internet protocols that underline all media streaming technologies and their security today. They championed the WebRTC-in-WebKit project with Ericsson R&D to bring WebRTC to Apple WebKit, and then Safari, and are recently most famous for handling the daily interoperability WebRTC tests for all browsers, desktop and mobile, as well as Electron apps. In addition, CoSMo designed the double encryption layer on top of WebRTC that powers Symphony Communication and in turn the secure and SEC-compliant communications of 23 banks across the world.

(1) — See here how “real-time” compare to “Live” in terms of latency.

(2) — What is low latency Live Streaming (link)

(3) — “2.5.6 Apps that browse the web must use the appropriate WebKit framework and WebKit Javascript.”

(4) — security in HLS is not mandatory (see chapter 10" Security considerations”, page 56, “SHOULD”) in the informal RFC8216, and is done at the application (HTTP) level. The exchange of the encryption key is not mandated to be secure either.

(5) — HLS specifications: RFC8216: chapter 1, page 4.

(6) —

(7) — “Real-time communication testing evolution with WebRTC 1.0

(8) — “testing webrtc products from clients to servers and infrastructures”​




Real-time video streaming.

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The Fastest Streaming on Earth. Realtime WebRTC CDN built for large-scale video broadcasting on any device with sub-500ms latency.

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