Top 5 Features for WebRTC Broadcasts - Update Digest
Top 5 Features for WebRTC Broadcasts
The Flussonic team releases more than a hundred updates every month, and to help you stay in the loop with all these changes, we’ve put together summaries of the most significant updates from the past year.
Today, we’ll dive into the WebRTC technology features available in Flussonic Media Server and explore how they benefit our customers. In this article, we’ve handpicked the top 5 most intriguing features, from our perspective. These updates are particularly relevant for those planning to launch streaming services catering to user-generated content such as blogs, adult entertainment, and gaming. Additionally, they prove invaluable for those aiming to stream online with minimal latency, encompassing events like concerts, sporting spectacles, and betting platforms.
Interestingly, some of our IPTV/OTT clients, initially not associated with user-generated content streaming, have ventured into the realm of WebRTC. They’re establishing their own web conferencing and similar services. Despite not being the most common application of WebRTC among our clients, we wholeheartedly support such endeavors by offering the essential tools for their implementation. This is one of the key strengths of our product—the ability to run various services in parallel on the same server, all under the same license!
WebRTC is indeed a publicly available standard and technology, but there are compelling reasons why customers often find our solution to be the superior choice. Our journey with WebRTC dates back to 2014 when we initially experimented with it. At that time, we couldn’t foresee the transformative impact it would have on video streaming and online communications. Nevertheless, our engineering acumen and unwavering belief in WebRTC enabled us to swiftly and effectively integrate this technology.
By 2019, our first customers were already utilizing WebRTC, and ever since, we’ve been actively expanding its capabilities for user-generated content (UGC) streaming. While clients certainly have the option to embark on their own WebRTC integration journey, the trend among companies and startups is to opt for off-the-shelf solutions like Flussonic. Why? Well, Flussonic goes beyond merely offering support and pre-built solutions for WebRTC; it delivers an all-inclusive package, complete with the essential components and tools needed to develop highly competitive services.
- Experience and Expertise: Flussonic boasts a wealth of experience, having worked with WebRTC for nearly a decade. Our seasoned team of developers possesses the know-how required to successfully implement WebRTC in diverse applications.
- Rapid Implementation: Developing a WebRTC solution from scratch can be a time-consuming endeavor. Flussonic’s ready-made integration significantly accelerates the implementation process, allowing customers to go live sooner.
- Skilled Developer Availability: WebRTC specialists are in high demand, and their services often come at a premium. Flussonic eliminates the need for customers to hunt for specialized talent by providing a team of skilled developers as part of the package.
- Ongoing Enhancement: We’re committed to the continuous improvement of our WebRTC functionality for UGC streaming. Customers who choose our solution benefit from regular updates and refinements without the hassle of managing these updates themselves.
- Complexity Simplified: While major platforms like Discord and Twitch can independently integrate WebRTC, the process can be complex and resource-intensive. Flussonic simplifies this complexity, offering a user-friendly solution for a wide range of applications.
We can safely say that among media servers, Flussonic Media Server offers the most comprehensive WebRTC solution. Many companies provide WebRTC in their products or services for integrating this technology into video services, but only Flussonic provides ready-made functionality for capturing video from various sources, transcoding (which is necessary for implementing adaptive bitrate), and a ready-made open-source player/publisher.
In the following sections, we’ll delve into these features in more depth, highlighting how they enable not just the utilization of WebRTC but the creation of exceptionally competitive services.
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WebRTC ABR - Adaptive Bitrate
For our WebRTC clients, providing a top-quality viewing experience across varying Internet speeds is crucial. Quality means viewers can enjoy video at the best possible quality their network can handle—some may get only lower resolutions due to network constraints, while others enjoy high bitrates without issues.
To tackle this, we developed an adaptive bitrate solution. Our WebRTC and MSE-LD player can adjust video quality based on the viewer’s internet speed. If there’s a slow connection, it reduces video resolution to prevent buffering, ensuring smooth playback. As the connection improves, it automatically enhances video quality for a better viewing experience.
However, we didn’t just focus on automatic quality control. We also gave users the option to choose their preferred video quality. We added choices like “auto,” “1080p,” “720p,” “480p,” and more, allowing users to pick the quality that suits them best.
We had to make significant changes to our media server core to seamlessly integrate it with the new player and provide access to different quality options. When streaming through WebRTC, we use the RTP protocol to send video and audio frames. We measure bandwidth using two methods: REMB and TWCC. In our Adaptive Bitrate (ABR) algorithm, Flussobnic decides whether to switch to a higher bitrate based on any of these measurements.
We also optimized Flussonic to handle variable bitrate video (MBR) and convert audio to the OPUS format, ensuring top-notch audio quality.
Most importantly, we integrated ABR directly into our platform, enhancing it with multi-source video capture and transcoding capabilities. All these improvements were aimed at offering our platform’s viewers the flexibility and high-quality video experience they desire.
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WebRTC AV1: Optimization and Integration
WebRTC AV1 represents the latest leap forward in video coding, and Flussonic Media Server is at the forefront of adopting this cutting-edge technology. AV1 has been designed with WebRTC in mind, and its integration is making a significant impact in the world of online video.
The television industry is known for its slow adoption of new codecs due to legacy codec support on many TV devices, with updates often taking years or even decades. In contrast, modern mobile devices and computers have long embraced AV1, and most web browsers now utilize it for video streaming. Leading video conferencing applications like Zoom and Skype have also actively embraced AV1 to enhance video quality and reduce data usage.
Given this context, it’s only natural that, in 2023, we encourage customers to move away from outdated codecs when it comes to online streaming. Modern computers can efficiently process video using AV1, and the integration of this codec into WebRTC was a logical step forward in optimizing and advancing video streaming technology.
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JIT-Packaging: Automatic Audio Codec Selection Based on WebRTC, RTMP, or HLS Protocols
Here, we want to introduce you to a unique feature of Flussonic Media Server—a mechanism that allows, within a single installation and under one license, the processing of both H.264 video streams with AAC audio (via the RTMP protocol) and H.264 video streams with OPUS audio (via the WebRTC protocol).
So, we have two groups of video stream sources:
- Browsers capable of publishing video in H.264 format and audio in OPUS format using the WebRTC protocol.
- OBS and similar publishers that use the RTMP protocol to provide video in H.264 format with AAC audio.
The challenge here is that you can’t simply take an H.264 stream in RTMP and publish it on a website via WebRTC, as it’s impossible to publish AAC audio streams on the site.
Flussonic addresses this issue by automatically generating both variations of audio tracks, irrespective of the audio format received at the input. When H.264 video arrives via WebRTC with OPUS audio, Flussonic adds an AAC audio track, and conversely, if the video comes through RTMP with AAC audio, an OPUS audio track is appended.
This ensures that we always have both audio track options (AAC and OPUS) available for H.264 video. Consequently, we can offer playback compatibility across various protocols, including WebRTC and HLS (which requires AAC). This approach caters to diverse device and browser requirements, giving content creators more options and flexibility when organizing online broadcasts while accommodating the limitations of viewers’ devices and browsers.
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WebRTC Canvas: Real-Time Creativity
Inspired by our clients’ requests, we’ve integrated Canvas tools into WebRTC, offering content creators more tools to craft unique and engaging content. Canvas is an impressive tool that enables real-time customization of video streams by adding individual elements and effects. With this feature, content creators can design remarkable visual solutions right from their browsers.
The WebRTC publisher (player) now comes equipped with the capability to add various effects to the camera’s video stream, as provided by the website’s developers. This functionality empowers users to effortlessly infuse their videos with a distinctive style. Examples of such effects include overlaying a company logo, adding scrolling text, applying video blurs, creating chat overlays, and much more.
WebRTC Canvas not only brings creative ideas to life but does so without overburdening the server-side during transcoding. This reduces the load on both users and providers. While server-side logo embedding is an option, using Canvas is the preferred choice for user convenience and provider resource optimization.
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WebRTC Load Balancer: Streamlining Traffic Optimization and Management
In the past, Flussonic Media Server utilized WebRTC through the web-socket protocol, which posed challenges for effective load balancing since web-sockets lacked built-in load balancing capabilities. Balancing WebRTC connections via web-socket was essentially a random process.
Now, Flussonic has introduced its own load balancing feature that accommodates various protocols, including HLS, DASH, and WebRTC. This has been made possible by transitioning to the implementation of WebRTC through the WHIP/WHEP (WebRTC HTTP Ingest/Egress Protocol) protocols, which are built on standard HTTP protocols. HTTP protocols are better equipped for efficient load balancing.
Flussonic’s load balancer can now intelligently select the optimal server for both transmitting and viewing video streams using WebRTC. This balancing feature enhances flexibility and optimization for video streaming, benefiting both viewers and publishers. It leads to more efficient utilization of server resources and an improved quality of service.
To try out the top 5 WebRTC features and more - request a free trial key or use your current Flussonic Media Server license.