What is WebRTC and how to make online communication and streaming more comfortable?
In live streaming services such as blogging, distant education, gaming and adult- streaming active interaction between the author and viewers is essential because it allows them to exchange opinions and make important decisions in real-time. There are benefits for all participants in such broadcasts, but the main resource for generating revenue for the streaming providers is the viewers and authors. Therefore, the platform must meet the requirements of both sides, otherwise, they would prefer another platform.
What is WebRTC?
WebRTC, which stands for Web Real-Time Communication, is an open-source project that provides web browsers and mobile applications with real-time communication via simple application programming interfaces (APIs). It allows for audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. Supported by Google, Microsoft, Mozilla, and Opera, WebRTC is being standardized through the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).
The way WebRTC works is intriguing in its ability to negotiate connections and transfer data between peers. When initiating a WebRTC session, one client needs to send an offer to another client, detailing how the media streams will be encoded. This negotiation is conducted using the Session Description Protocol (SDP). To facilitate the communication between peers, WebRTC utilizes Interactive Connectivity Establishment (ICE) to find the best path for the data to travel through, which could involve navigating through firewalls or Network Address Translators (NATs). Moreover, it uses STUN (Session Traversal Utilities for NAT) servers to find out the public IP address of each peer, and TURN (Traversal Using Relays around NAT) servers are used as a fallback to relay data if a direct peer-to-peer connection fails.
In a nutshell, WebRTC facilitates direct, peer-to-peer communication between users, significantly reducing latency and enhancing the quality of video and audio transmissions. The simplicity, alongside the low latency and high-quality communication provided by WebRTC, makes it a sought-after technology for real-time interactions on the web, enabling rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allowing them all to communicate via a common set of protocols.
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WebRTC is an open standard that allows for independent development and integration into a streaming platform. However, a more effective solution for WebRTC integration is to use a ready-made and proven tool. Flussonic Media Server is an example of a platform that has successfully implemented WebRTC technology for a long time. We have taken the best of WebRTC technologies and even a little more, so that streaming platforms can adapt to the required use cases and real-life conditions of their authors and viewers.
By incorporating Flussonic Media Server for one-to-many WebRTC communication on your streaming platform, you can reap several benefits that enhance the live streaming experience by providing a greater sense of immersion and interactivity.
- Security. Publishing content directly from the browser to the website ensures data security since you don’t need to provide your data to the streaming applications
- Stable (consistent) low latency, which does not fluctuate, is a necessary condition for timely feedback from author to viewers (and vice versa) that creates a sense of real communication.
- Guaranteed sound quality with minimal latency, sufficient for phone calls, will help maintain excellent audio communication and reaction speed during live streaming.
- Using WebRTC allows reducing the latency for iOS device users to sub-second values. iOS doesn’t support MSE, and WebRTC becomes the only way to get video with low latency. This is especially important for live streaming, as any delay can greatly affect the perception of content.
In addition, the Flussonic Media Server:
- Enables accurate load balancing by taking into account node utilization, allowing the stream to be directed to the appropriate server and optimizing hardware resource utilization.
- Automatically switches between TCP and UDP and enables selecting the optimal protocol for data transmission depending on the type of data and network conditions.
- Constantly analyzes the Internet connection and allows quickly responding to changes in the outgoing video’s quality.
- Provides excellent scalability and ensures uninterrupted delivery of video with the maximum possible quality for thousands, hundreds of thousands, and millions of viewers.
- Supports WebRTC ABR: for some viewers, the maximum possible quality will only be in low resolution (their Internet connection cannot handle more), while others can consume the maximum possible bitrate without problems.
When to Choose Flussonic with WebRTC for Your Live Streaming Needs?
There are several widely used streaming platforms, including Twitch, YouTube, Periscope, Zoom, and Facebook, that offer live streaming and interactive features for content creators and their audience. But there are situations when these platforms are limited by user content and technologies that may not meet specific requirements.
For example, in some countries, Zoom is blocked or has restrictions for use by certain organizations, particularly those affiliated with the government. Popular services like Twitch and YouTube may restrict certain content, such as adult streaming, which may not pass pre-moderation. People who have scheduled appointments with a psychologist or virtual medical consultation are unlikely to use public services to protect their data. Companies conducting live training sessions may also want to protect their content. It is essential to note that YouTube has strict copyright rules, and the platform has the full right to block an author’s content if it violates these rules, which may have severe consequences for the author, even resulting in the closure of their channel.
Speaking of technologies, most platforms use widely adopted protocols such as MPEG-DASH, HLS and RTMP, but they have their limitations concerning sound quality, adjustment to unstable internet channels, and not all of them provide the sub-second latency on iOS. And most importantly, platforms like YouTube, Periscope, Facebook require the installation of applications to start streaming, which is not always welcomed by authors and viewers.
Weak points of WebRTC
Although WebRTC has many significant advantages, in some cases it is more profitable to use TCP communication. Let’s have a look at which ones:
WebRTC uses UDP as its protocol. This makes communication easier, but reduces reliability from a security perspective. As mentioned above, TCP is a series of steps from the moment the client begins to connect to the server until the other party acknowledges that the signal has been received. In other words, availability is guaranteed by the ability to confirm every time whether the other party has received the data. On the other hand, UDP, which sends data continuously, cannot confirm whether the other side has actually received it.
All WebRTC solutions are not compatible with each other, since the standard describes only methods for transmitting video and sound, leaving the implementation of methods for addressing subscribers, tracking their availability, exchanging messages and files, scheduling, and other things to the developer. In other words, you will not be able to call from one WebRTC application to another.
WebRTC determines the real IP addresses of users. At the same time, neither a proxy nor the use of the Tor network will help maintain anonymity. You can hide your IP address using various VPN services, as well as using a TURN server.
While not every live streaming scenario requires the use of WebRTC technology, if achieving a strong sense of audience engagement and interactivity is critical to your business, and if you require real-time responsiveness, high-quality audio, adaptability to varying internet conditions, or if you’re facing restrictions with other platforms, then building your streaming solution with WebRTC is the answer. With Flussonic Media Server, you get all the advantages of the WebRTC technology stack without having to develop infrastructure from scratch. It is a ready-made platform that allows you to focus on your business, content monetization, attracting authors and viewers who need the “true-to-life experience,” while we take care of the technical aspects of streaming.
Flussonic Media Server Team Lead
At the forefront of Flussonic innovations: responsible for development of Flussonic Media Server, video analytics & UI services