What is WebRTC? Everything you need to know

WebRTC, which stands for Web Real-Time Communication, is a powerful communications technology that enables real-time audio, video, and data sharing directly between web browsers and mobile applications. Since its introduction into the content delivery sphere, WebRTC has simplified and enhanced online communication by removing the need for plugins or external software, making data sharing much more efficient and economical. WebRTC is, in fact, a free, open-source project that is swiftly becoming a standard feature of modern digital communication solutions, such as those offered by System73. 

But how exactly does WebRTC work? What are the benefits of using WebRTC? And what can we expect from the future of data sharing and online communication using this innovative tech? Let’s explore. 

What is WebRTC?

WebRTC is used for a range of applications across multiple industries, such as video chat, file sharing, and screen sharing. Its ability to deliver high-quality, low-latency communication makes it a favorite tool among developers of these kinds of solutions. WebRTC was born out of a Google project with the primary goal of enabling seamless peer-to-peer communication without the need for plugins or additional software. Its architecture is built around three key components: MediaStream, RTCPeerConnection, and RTCDataChannel. MediaStream handles the capture of audio and video, RTCPeerConnection establishes and maintains the connection between peers, and RTCDataChannel allows for direct data transfer.

What are the benefits of using WebRTC?

Aside from WebRTC being a free-to-use and open-source software, it offers several other benefits that make it an attractive choice for real-time communication solutions. First of all, it requires no plug-ins, unlike other technology, which means it can be integrated easily into existing applications. It is commonly used for video conferencing and webinars, online gaming, file sharing, customer support, and live streaming on social media platforms, which, thanks to WebRTC's capabilities, can deliver seamless, real-time communication and data transfer, enhancing user engagement and functionality. WebRTC is also able to achieve low and ultra-low latency results when broadcasting live events to global audiences comprising millions of end-user devices. This ability to ensure minimal delays in content delivery and provide high-quality audio and video are crucial for high-quality experience and user satisfaction, two of broadcasters' main concerns that System73 is able to serve with exceptional results. 

How secure is WebRTC?

WebRTC is proven to be a reliable and secure option for real-time communication thanks to the robust security features it offers, which ensure the confidentiality and integrity of data shared over the network. This is made possible by its security framework based on end-to-end encryption, which protects the data exchanged between peers in a P2P network such as that created by our live content delivery solution, Edge Intelligence. This makes it almost impossible for unauthorized players to access the network and infiltrate the data. It is only received and accessible to the intended recipients. WebRTC also utilizes Datagram Transport Layer Security (DTLS) to secure the initial key exchange process, preventing potential man-in-the-middle attacks. 

What does the future hold for WebRTC development?

Future developments of WebRTC are likely to converge with the ongoing development of AI applications. These powerful tools have the potential to enhance audience-content interaction, participation, and engagement, thanks to AI-powered features such as voice recognition, natural language processing, and sentiment analysis during live events, particularly in professional settings or entertainment applications such as online gaming. 

The impact of these parallel developments could also be felt in industries such as healthcare and education, where applications incorporating virtual reality (VR) and augmented reality (AR) could significantly impact how patients and students interact with digital content. These advancements could lead to more immersive and interactive experiences, transforming telemedicine consultations, remote learning environments, and personalized training programs. As AI, VR, and AR technologies continue to evolve alongside WebRTC, we can anticipate a future where real-time communication is not only more efficient but also more engaging and transformative across various sectors.

Visit system73.com for more information about WebRTC and Edge Intelligence or to book a call with our team.