The RTC standard enables dynamic connections directly on the browser to capture and transfer audio, video, and real-time exchange of data. It can also be implemented in online browser-based gaming applications and, security and identity management.
This way you can engage and collaborate in real-time without the need to install plugins and drivers or download third-party components.
In the past, browsers weren’t equipped to play media or even support sound. That’s why many websites relied on Flash plugins. Over the years, browsers became more advanced. By adding new functionalities like video and audio processing, new network protocols support and APIs, browsers are now also capable of delivering high-quality RTC.Today, browsers manage this complexity by relying on the following APIs:
getUserMedia (also known as MediaStream)
Enables your application to access users' media devices. The user will notice a notification asking for permission and after granting it, the stream will be available to use from the code.
By building an RTC build connection object, you will be able to send the media peer-to-peer using SRTP protocol allowing the data to go straight to the other browsers without storing it in the middle while also keeping it encrypted.
Audio and video aren’t the only type of content that webRTC is able to transmit. This technology also supports chats, screen sharing, browser online videogames and data exchanges.
WebRTC is free. In order to use it, you need a web browser that supports this technology and grants relevant permissions like access to your microphone, speakers and camera. In case you want to use WebRTC over a local network, you will still need a server for signaling server even if you are establishing a peer-to-peer connection.
All modern browsers support WebRTC, among those:
Establishing connections between users using only a browser is advantageous for many reasons. Eliminating the need to install plugins, third party software and updates is a real time-saver that allows you to initiate communication fast. With WebRTC dealing with video and audio encoding/decoding, data transfer and data loss, all comes built-in into the browser OOTB.
Its popularity will increase since more and more browsers are embedded in smart TVs, for example.
WebRTC has an open-source nature while enjoying support from the big industry players and thousands of independent developers that constantly update its framework and functionality.
WebRTC doesn’t consist of a single technology, but rather a collection of open-source protocols and standards. Google initiated the original idea in 2011. The Internet Engineering Task Force (IETF) defines the protocol set while new APIs are standardized by the World Wide Web Consortium (W3C).
This means this technology is constantly evolving by undergoing active development. Furthermore, since WebRTC is platform/device-independent, we will see it integrated into more and more applications in the future.
In order to create high real-time performance, WebRTC adapts to network conditions and creates efficient use of bandwidth delivering quality media streams. Furthermore, all of the processes are done directly in the browser because WebRTC technology can dynamically adjust its parameters.
Once the web application obtains the optimized media, it then gets distributed to its peers or post-processed using one of the media APIs.
With 100–500ms ultra-low-latency, WebRTC is one of the most efficient protocols on the market due to the fact that it’s based on UDP (User Datagram Protocol). This means that the data transfer is enabled before the receiving party provides an agreement. This makes your stream appear faster and more interactive.
WebRTC is a true peer-to-peer protocol as it can create direct connections between two parties over the internet without the need for a third party. However, in order to scale it, this technology needs to rely on a media server in order to avoid bandwidth bottlenecks. That’s why Digital Samba can help you scale your business with our video engine API. Check it out here!
WebRTC is secure because it guarantees safe encryption and it safeguards the transmission of sensitive information. WebRTC uses two standardized encrypting protocols: Datagram Transport Layer Security (DTLS) and Secure Real-time Transport Protocol (SRTP).
The IETF enforces mandatory encryption and security standards on all WebRTC communications. In fact, creating an unencrypted network is a big security concern, and it’s prohibited.
While maintaining and controlling your own servers can bring you certain advantages, nowadays, cloud-based APIs are rapidly overtaking the market as the preferred solutions. They are optimizing IT costs since the customer pays only for the rented space on the hosted servers. The IT team can save time and focus on other parts of the development without setting up, maintaining, or securing the servers.
Keeping communication alive is essential in today’s changing landscape. WebRTC brings a whole new set of opportunities directly on your browser from one-on-one voice and video calling to virtual workshops and hybrid events.
As Justin Uberti, Tech Lead on WebRTC at Google says:
WebRTC fills a critical gap in the web platform as you can communicate in real-time just by loading a web page.”
Considering its emerging technology, we will see more and more companies embrace WebRTC in the future!
Photo by Jacob Lund from Noun Project
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