Given the virtual milieu all around, almost everyone has been looking forward to point-to-point and instant communication. Businesses are trying to make their communication system as smooth as possible. And that is exactly where WebRTC comes into play. It enables you to conduct barrier-free real-time communication directly through a browser. Sounds interesting, right?
In this article, we are going to discuss every important aspect related to WebRTC. Think of it as a WebRTC tutorial without codes. It is meant to cater to the needs of different kinds of readers such as individuals who are curious about WebRTC, novice developers who are new to the concept, developers who wish to dive deep into WebRTC, developers looking for a specific WebRTC solution, or someone who wants clarity on debugging. Are you one of these? Great! Stay with us and read on.
What is WebRTC? – Understanding the Meaning in Detail
WebRTC stands for Web Real-Time Communication. It can be defined as an open-source protocol. It was built by Google back in the year 2011. Not only does WebRTC enable real-time communication but also data sharing between different browsers and devices. It tends to empower peer-to-peer communication in web browsers and devices by leveraging application programming interfaces.
Gone are the days when you needed external plugins to conduct virtual communications. WebRTC involves voice, video, and data transfers and clears the way for two-way communication between two web browsers. A simple WebRTC video, audio, or data sharing technology empowers users to access the camera and microphone on their phone and laptop. It also enables entire screen display and remote screen recording. In a nutshell, WebRTC has been a boon to corporate businesses when it comes to conducting interactions in a virtual setup.
Wondering How WebRTC Works? – 3 Components You Must Know About
If you’ve always been curious to understand the functioning of WebRTC, this section is for you! Now, WebRTC works on three primary components – media capture & stream API, peer connection, and data channel. All these three components together account for peer-to-peer communication. Let us closely look at each one of them:
Media Capture & Stream API
The media stream API is meant to provide support for a simple WebRTC video chat or audio call. With media capture and stream API, one can easily access the camera and microphone of the device. It also tends to offer security. Think of user permissions that involve asking for the user’s consent before any web application can start streaming something. To put it in a nutshell, the media stream tends to facilitate audio and video data streaming via the devices.
Peer connection is one of the most essential parts of the WebRTC specifications. It tends to connect two applications on separate computers to interact leveraging peer-to-peer protocol. The objective of RTC peer connection is to conduct direct communication and avoid taking the help of any intermediary connection.
Data exchange between two browsers can prove to be a complex process. But thanks to the RTC data channel API, data can be directly transferred from one peer to another. It tends to function on Stream Control Transmission Protocol. The best part about the data channel is that it encourages reliable stream delivery over the web.
How is WebRTC Video Streaming, Audio Callingand Data Sharing Done? – 4 Important Steps
Have you been wondering how audio calling and data sharing is performed via WebRTC technology? Or how does a video chat app using WebRTC protocol function? Don’t worry! We are here to help you. In this section, we are going to discuss the steps involved in establishing communication through WebRTC.
All the steps are sequentially arranged. Only after the completion of the first step, the next step happens, and then the third one and so on and so forth. Now, every set is a combination of certain protocols. Let us look at each step one by one to understand how a simple WebRTC video chat, audio call, or data share works.
WebRTC signaling may be described as the process of setting up, controlling, and terminating a particular interaction session. The peers involved in the real-time communication tend to send their stream to the server. The server encodes and delivers the same to the receiving peer. Signaling tends to follow a SDP (Session Description Protocol) format. It is a plain text format and contains the list of media sections. Now peers can share four different details with each other:
- IP address of both agents
- Audio and video tracks produced and transferred by one agent
- Video and audio tracks consumed by another agent
- Data channels that determine the type of media
Any audio calling or data sharing or video chat app using WebRTC requires browser support. And the browsers tend to communicate via the signals they send to the servers. This is where signaling has a role to play.
It is defined as bi-directional interaction between two peers. A WebRTC video or audio communication is a P2P connection. It is not a client-server connection. Let’s have a look at the major protocols involved.
- ICE servers – It tends to find the best way in which two agents can be connected.
- STUN – It stands for Standard Traversal Utilities for NAT and allows WebRTC clients to find their own public IP address.
- TURN – It stands for Traversal Using Relays around NAT and enables connection between two agents when firewall restrictions prevent the same from happening.
Next, WebRTC focuses on security. It makes sure that all the communication that happens between two agents is encrypted. For this, it has two pre-existing protocols in place.
- DTLS – It stands for Datagram Transport Layer Security and involves certain values known as ciphers. The client and server need to agree on the same in order to communicate.
- SRTP – It stands for Secure Real-time Transport Protocol. It tends to encrypt RTP packets.
The last step is communication. WebRTC video, audio, or data sharing focuses on quality over latency, end-to-end encryption, SDP coordination, and lesser bandwidth cost. Now, WebRTC enables unlimited data sharing over the web.
A quick glimpse into the rich history of WebRTC
Well, the history of WebRTC moves back to 1999. It was first developed by Global IP Solutions, popularly known as GIPS. However, in 2011, Google acquired GIPS. Since then, WebRTC has been consistently rising and gaining popularity. In 2014, the WebRTC technology was integrated with Google Hangouts. Many leading browsers such as Chrome, Opera, and Firefox showed their trust in RTC technology.
What does WebRTC have for us today?
Thanks to WebRTC, we are communicating smoothly over the web! You can build a secure audio call and video chat app using WebRTC protocols. The best part is that WebRTC does not require any plugins or external installations to conduct real-time communication. All you need is a decent internet connection and you’re good to go. As it is an open-source project, it is extremely simple to use and modify.
WebRTC technology is here to stay!
As WebRTC has a comprehensive suite of benefits to offer, its demand is constantly rising. Given the current pandemic situation, businesses have been focusing on building a robust communication infrastructure that supports real-time interactions. This is exactly where WebRTC has a role to offer. It has, certainly, made real-time communication smoother. Be it the telecom, trade, gaming, or any other relevant industry – WebRTC has great potential to revolutionize the ways communication is conducted.
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We hope our WebRTC tutorial-like article helped you understand all the important aspects revolving around the concept of WebRTC. In case you’re looking to build a communication application, opt for WebRTC enabled chat SDKs and APIs. Now, wait no more. Solidify your business interactions with WebRTC technologies today! You have our best wishes.
Frequently Asked Questions (FAQ)
The thing that makes WebRTC unique is that it enables person-to-person communication. This means that WebRTC handles all the details of directly connecting two devices and transmitting the audio and video data in real-time. This might seem simple, given the ubiquity of phone calls and video chat.
WebRTC (Web Real-Time Communication) is an awesome new tech for video/audio chat directly inside your browser or mobile app. The best of its kind. But like many new technologies, it has some nasty pitfalls even for experienced developers. This is the story of our WebRTC chat app.
WebRTC leverages three HTML5 APIs enabling browsers to capture, encode, and transmit live streams. While streaming workflows can often require an IP camera, encoder, and streaming software, the most basic WebRTC use-cases can manage the whole enchilada with just a webcam and browser.
Availability. WebRTC today is available in all modern browsers. Google Chrome, Mozilla Firefox, Apple Safari and Microsoft Edge support it. You can also “take” it and integrate it into an application or an embedded device without the need of a browser at all.
While both are part of the HTML5 specification, WebSockets are meant to enable bidirectional communication between a browser and a web server and WebRTC is meant to offer real time communication between browsers (predominantly voice and video communications).