In the last two years, real-time communication has been more important to unified communications (UC) and collaboration than ever before. From remote learning to team discussions, from customer support to events, a variety of business activities now rely on the real-time transmission of audio and video.
This is where WebRTC comes in. It is an open-source, standardized way to enable video experiences to run on all WebRTC-compatible browsers with minimal effort.
What is WebRTC? Definition
WebRTC (Web Real-Time Communication) is an open-source technology created by Google that enables browser-to-browser real-time communication and data exchange, primarily focused on audio and video traffic.
Without WebRTC, devices cannot connect with each other, unless there is an intermediate server. One device transmits the information to a server, and then the server delivers it to the second device; thus, both devices must have the same plug-in or software loaded for communication to occur.
That is why, shortly after the release of Google Chrome, the company aimed to develop the essential requirements for seamless data transmission on a standardized platform, hence removing the requirement of any third-party applications or plug-ins. Mozilla, Microsoft, Opera, and Apple all came on board within a few years. Today, it is a free and open-source project that anyone can leverage, providing websites and mobile applications around the world with real-time communication (RTC) capabilities. To understand what is WebRTC better, it is important to know the context in which the protocol emerged.
How Did WebRTC Come into Being?
Shortly after the release of Google Chrome, its development team observed that the Web’s infrastructure was insufficient for real-time interactions. There were no standard implementations in any browser, as well as the absence of a browser-wide protocol, that enabled direct data transfers among individuals. Google aimed to provide the essential requirements for seamless data transmission on a standardized platform, thereby removing the need for third-party applications or plug-ins.
Google acquired Global IP Solutions or GIPS in May 2010, a VoIP and videoconferencing solutions provider that has created several RTC-required components, including codecs and echo cancellation algorithms. Google has also open-sourced the GIPS tech and collaborated with the IETF and W3C to achieve widespread industry consensus.
WebRTC (which as discussed is an open-source project for browser-based real-time communication) was announced by Google in May 2011. Ericsson Labs created the first version of WebRTC by working on a modified WebKit library in the same year. This makes it a collaborative effort by prominent technological giants, which is now a common occurrence in unified communications.
How Does WebRTC Work?
WebRTC offers a real-time peer-to-peer link between two or more browsers for the sharing of private voice, video, and data. It employs three major elements to do this:
- The media stream – The media stream is an application programming interface or API that enables access to the device’s camera and microphone. It manages the multimedia activity and data consumption of the device. The Media stream manages the device’s information on media capture and rendering. Ideally, it facilitates the streaming of audio and visual data via the devices.
- The data channel – The RTC data channel facilitates the transport of arbitrary data in both directions between peers. This is valid SCTP (Stream Control Transmission Protocol). A data channel is meant to alleviate congestion on networks such as UDP. It guarantees consistent stream distribution via the Internet.
- Peer connections – WebRTC was created to build a peer-to-peer connection over the internet. The basic purpose of an RTC peer connection is to establish direct contact without the need of an intermediate connection. Peers may acquire or consume material, particularly music and video, as well as create it.
WebRTC Use Cases for the Enterprise
There are several apps that use (or may utilize) WebRTC technology. WebRTC is most often used for video conferencing as the foundation for video chat apps. WebRTC is often used if a meeting tool is accessed using a web browser (rather than by installing an app). Additionally, it may be used for contact centers, in-context communications, telemedicine, social media, and peer-to-peer data transfers.
WebRTC is widely used in telehealth and telemedicine since it has built-in encryption. This is great for HIPAA compliance and other healthcare privacy and security considerations. WebRTC is also useful when some component of customer service would be more effectively actioned through video so that the agent can see what the consumer is viewing. Using WebRTC SDKs from Telnyx or other firms, it is also possible to make a call to any phone number, anywhere around the globe, via a web browser.
What is the Benefit of Using WebRTC?
Because WebRTC needs no plugins, frameworks, or programs, all one needs is a WebRTC-compatible browser. For the end user, WebRTC apps function “out of the box.”
WebRTC is completely peer-to-peer, therefore there are no costs associated with utilizing bandwidth across a network. Moreover, since WebRTC is fully browser-to-browser, consumers get the best speed and lowest possible latency. Consider the case of a user attempting to send a file. Without WebRTC, one must upload the file to a server, then the receiver must download it. With WebRTC, the file is sent directly over the WebRTC Data Channel, without the need for servers or infrastructure.
That is why nearly every leading UC company today uses WebRTC, owing to its ease of use, security, and cost benefits.