Quality with Real-Time OTT media? It’s all about the feedback!

There are many that happened this week, both during IETF sessions and in the ecosystem, that troubled me enough to write a dedicated blog post. People from the streaming industry, and from the webrtc industry alike, are approaching OTT media in general, and webrtc specifically, the wrong way. It works, but it does not work well enough. It connects, it streams in perfect conditions, but it does not stream with good quality, at scale and in real-time. What is missing?

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#WebRTC 101: 1st assignment

Now that we understand the basis of libwebrtc code management, we can start answering otherwise problematic questions. This week I was at CommConUK, and was discussing the number of contributors to libwebrtc, pointing my interlocutor to the AUTHORS file to start with. “Less than 100” was the other party position, and to be honest, I had never checked. So, who would risk a guess as to how many contributors to the webrtc stack there were in the past three years, and more importantly, how to check?

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#WebRTC 101: Fetch the source

I first started writing about libwebrtc source management, build and test systems almost 5 years ago. While the posts are still here, and mostly accurate, people forget, and/or the system as changed just enough that we need to update what is the de-facto reference for libwebrtc. As we are writing a book, with examples and illustration to be used in classrooms to teach the underlying principles, and in companies for e.g. on boarding Engineers, we though we should put some extract here.

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Real-Time AV1 (in #WebRTC) is now Production ready!

After the release of the Codec Spec in march 2018 (“frozen bitstream” and reference decoder), the next step qs to show that AV1 could be used in production. The decoder had to be made fast enough on commodity hardware, hardware vendor had to integrate AV1 support in their chips, for the base profile. Then advanced profiles (SVC, …) and modes (lossless, Real-Time) would deliver. For the work on real-time and SVC modes, a specific subgroup was created to continued the work beyond just the codec Spec. Integration with the Real-Time-Protocol (i.e. writing an AV1 RTP Payload specification), and usage of SVC in conjunction with Media Servers (RTP Header extensions, …) needed to happen. On Halloween 2018, CoSMo demo’ed live the first AV1 RTP integration in WebRTC. It did not support SVC and was not Real-Time. On June 26 2019, Cisco demonstrated live from New York the first Real-Time AV1 RTP Integration in WebRTC, through a modified version of their flagship product: webex. It denotes a new step in the evolution of AV1, one that happens 12 months earlier than anybody thought it would.

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Wowza’s marketing at work again. #webrtc.

Let me be honest, I really dislike marketing in general. Public researcher by training, I’m looking forward to the truth, to reproducible results, and claims that are backed up by data and processes I can access by myself to reproduce the results and the conclusions of the analysis. Marketing is very often the opposite of researching of the truth: it aims at making the market buy your product(s) and service(s). Very often, the end justify the mean, and FUD, deceiving and/or unsubstantiated self-serving claims become the norm.
Let’s be clear, if you don’t have a killer feature, you might as well pretend that either a/ you have it, b/ it is not something important, c/ what you have is so much better. Eventually, as all lawyers know, if you can’t argue the facts, go after the credibility of those bringing them to light. Microsoft have made this approach very famous and even gave it a name: Fear, Uncertainty and Doubt: FUD.
When I read some Wowza marketing piece I cannot help but noticing a strange correlation with those technics, and it’s frankly upsetting. We have just made a “Fact checking” session about WebRTC at Live Streaming West in New York. I think it’s time to do it again here, so at least people can have access to enough verifiable information to make their own mind.

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Video CDN and Satyagraha: admitting defeat in the course to perfect live streaming.

Satyagraha, (Sanskrit and Hindi: “holding onto truth”), was the philosophy proned by gandhi when addressing the British Imperialism: non-violence, self-scrunity and research of truth. It lead to citations like “First they ignore you, then they laugh at you, then they fight you, then you win”, and more simple saying like “believe what they do, not what they say”, which should be applied any marketing material, really.

For the past two years, all CDNs and most streaming server vendors, have been trying to downplay the disruptive impact that WebRTC would have on their business. It wouldn’t scale. There would be no market for it. It would not be needed. It has now changed.

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KITE 2.0 – The best WebRTC testing engine now with more free content

WebRTC testing through KITE is getting more and more popular. As we updated the cosmo website, KITE is also receiving a lot of new features in this 2.0 release. Still the only end-to-end testing and load testing on the market that can be run on-premises, and to support web apps (desktop browsers and mobile browsers) as well as native apps, KITE has been leading the market from the clients application support, and price, points of view, but was still slightly being in term of usability. It was an expert tool made for experts. This new release add many free features, as well as many new commercial options for automated load testing on top of your own AWS account, for minimum cost.

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WebRTC 1.0 Simulcast vs ABR

In a public comment to millicast recent post about simulcast, Chris Allen, CEO of infrared mentioned that they have been supporting ABR with WebRTC in their Red5 Pro Product for a long time. While his claim is valid, and many in the streaming industry use a variation of what they do, there are two very important distinctions that needs to be made between ABR and simulcast. We made the distinction about latency quickly in our presentation at streaming media west last year, however possibly too quickly, and we never really explain the distinction about end-to-end encryption, so we though we should dedicate a full post this time around. WebRTC with simulcast is the only way to achieve the lowest latency possible, and real end-to-end security, with a higher flexibility than DRM can provide.

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