What is the role of WebRTC in the broadcast industry in 2023? To find out, WebRTC.ventures CEO and founder Arin Sime attended the National Association of Broadcasting (NAB) annual conference, NABShow 2023, in Las Vegas on April 17th and 18th. For a video overview of the conference, Arin recorded this report, which we then expand on in the blog post below.

While attending the NABShow, I had the chance to attend a number of talks in the Remote Broadcasting mini-conference within the larger NABShow conference. I spent a lot of time walking the multiple exhibit halls, talking with vendors, partners, and even meeting up with one of our clients!

While the NABShow exhibit halls are mainly stocked with vendors selling camera, lighting, and audio equipment, there is much more than hardware here. Many of the software vendors utilize WebRTC technologies (even though WebRTC was not originally designed for broadcasting). This is especially true in the Remote Production aspect. 

Remote Production via WebRTC

During one session on a Remote Production case study, presented by Michael Clayton of Base, he discussed the history of broadcast production. He described the pandemic era of remote production as “nomadic.” Due to sudden changes in the work environment, studios had to cobble together remote production workflows as best as they could. While they were remarkably effective with this “DIY” strategy, it was not sustainable. Now we’ve entered the post-pandemic phase of true cloud-first virtual remote production techniques. This emphasis on cloud-first production in the browser was evident on the showroom floor. Remote Production is here to stay.

One booth I visited was TheSwitch (recently acquired by Tata Communications). They showed a very impressive remote production tool that runs entirely in the browser. It allows for multiple video feeds to be monitored and produced entirely in the browser over a connection of no more than 5MB. (Knowing that I’m a WebRTC person, they even showed me the webrtc internals tab during their demo. It did not go over 3MB in usage!) In addition to handling traditional RTMP video feeds from studios, remote guests could also join a broadcast via a WebRTC meeting link and enter the stream via their Guest Conferencing module. 

I also attended a talk by our friends at Dolby.io, where Ryan Jesperson showed off the Dolby Streaming platform. This is a WebRTC CDN based on their acquisition of Millicast a couple of years ago. Ryan was a guest on WebRTC Live in the past, as well as Sergio Garcia Murillo, where they’ve talked about different aspects of using WebRTC for real-time streaming and AV1. WebRTC was not necessarily designed for streaming. Millicast and Dolby have done a fantastic job of making a very low latency implementation that allows for very interactive broadcasting. 

HLS broadcasting is great for watching recorded content which is not live. But it’s not good enough in live streaming of sports events where latency needs to be zero, especially when combined with use cases like gambling or remote commentary. The Dolby WebRTC streaming solution is quite impressive. At the booth, they showed a competing solution (still using WebRTC!) that had latency of up to 15 seconds next to their solution which had no visible delay as we moved around in front of the camera.

Ryan’s presentation also talked about different remote production tools that have been built on their platform. They showed functionality like their multi-view capabilities that allow the producer to toggle between different camera angles to choose which to send out on the stream. This same functionality could be used on the user side if you wanted an implementation where each individual user can choose the camera angle they want to see, or the language track they want to listen to.

DRM and WebRTC

Another topic that I learned more about at the NABShow is Digital Rights Management. DRM is a layer of security that can be built into video to ensure that the licensing for that content is not violated. While WebRTC already has encryption built into the streams between two peers, that does not provide protection over how the content is used. DRM is most commonly used in streaming of content like movies and subscription media services, to ensure that the content cannot be pirated and displayed in other players. It’s an extra layer of encryption and key management on top of what WebRTC would provide, and can be used to ensure that the content is only played back in authorized players.

I typically thought of this as only being relevant in use cases like streaming watch party tools, where a group of remote viewers uses a WebRTC-based group chat to watch Netflix together, or stream a concert, or similar entertainment and media use cases. While that is the primary use case, there are more WebRTC applications that could benefit from this.

I visited the booth of CastLabs, where they showed me a demo of their JavaScript DRM library for WebRTC. The library will apply DRM on top of any WebRTC stream. They needed this functionality for themselves and their clients, and then decided to open it up for others to use. If you were to try and join the stream without having the correct DRM key management in place, you would see a garbled green screen instead of the actual video. It also prevents you from screen capturing the video via your laptop, as the screen capture will also be garbled.

In addition to streaming movies, you could use this in other common WebRTC use cases like EdTech. Imagine that you have an online course behind a paywall or subscription service. Without DRM, someone could buy one subscription to your course, and then steal the content via screen capture and distribute that video content under their own name or paid service. Using DRM in this use case will prevent such theft, and could be used for other secure or proprietary communication tools such as sensitive corporate meetings or as an extra layer of privacy in a telehealth call.

Using AI to combat Packet Loss

Finally, another area that I learned more about at NABShow is the possibility of using AI to create intelligent routing networks to handle streaming media content on the public internet. I attended an interesting presentation by Michael Yang, SVP for Corporate Strategy at Caton.

Michael talked about their technology for transporting broadcast-quality video and large files over the public internet. It provides more efficient media streaming with very low packet loss. They provide SLA’s of 99.9999% which is better than a leased line, and uses AI to intelligently route traffic over their globally distributed network of POPS.

Michael stated that their solution reroutes traffic in less than 20ms when it detects a network problem. In some of their more complicated scenarios, this might mean that an IP broadcast over their network makes thousands of network switches per day with zero-errors. 

While most of the examples given were around broadcast media, this same technology could also be used in other WebRTC media connections that require very high reliability. Michael talked about a Brazilian healthcare scenario where surgery is being broadcast over their network, and remote specialists need to be able to comment on the surgery in real-time with zero transmission errors.

We have not tried Caton’s offerings yet. But the presentation was certainly intriguing and it reminded me of SubSpace, a company that we interviewed on WebRTC Live in 2022 about a “parallel network optimized for video traffic.” Unfortunately SubSpace ultimately folded as a company. As I said in that episode, I do see a need for services like this and so perhaps Caton has an offering that can be commercially viable and useful for highly reliable WebRTC video needs.

Using Offset for production collaboration

While at NABShow, I was happy to meet up again with our client Todd Ducharme, co-founder of Offset, a remote production collaboration tool. Our team has built the initial version of their application which allows production teams to collaborate together in real-time in a hybrid model. Some production staff are on the set, and others are “off set”. Remote members are still able to see a livestream of the camera feeds and collaborate both with each other and the on-set staff in real time using various channels and multi-modal communication of audio, text, and voice.

Part of the WebRTC.ventures team working with Todd Ducharme in a design workshop on the Offset application.

Be sure to watch our video update from NABShow 2023 to hear Todd discuss Offset more, and how he and his co-founders (who are themselves filmmakers) are able to save studios hundreds of thousands of dollars in travel costs while still allowing key figures like advertising executives to participate in the filming of a commercial in real-time.

Building custom broadcasting applications with WebRTC

NABShow 2023 was a great event, and I’ll definitely try to return to it next year. It was interesting to learn more about film and broadcast. This was true even on topics outside of WebRTC. One panel discussion I attended was about how the camera crews captured the incredible aerial footage in Top Gun: Maverick, and how “The Rings of Power” is fueling new cloud-based collaboration tech.

Even just staying focused on WebRTC, there was so much ground to cover. It’s incredible to see the number of broadcasting applications that now use WebRTC. They are on the market and changing the way film, TV, sports, and podcast production happens. It’s a great example of how much WebRTC has matured over the last few years and how widely it is being used.

There is still so much room for innovative applications in the broadcasting space, like our client Offset. Regardless if you are building an application for live streaming, online education, interactive broadcasting, remote production, or many other use cases, our team of experts at WebRTC.ventures can help. We will apply our years of experience and our deep relationships with industry partners to help you build an innovative and transformative broadcast experience. Contact us today!

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