WebRTC, an open source project started by Google in 2011, provides API-based communication between web browsers and mobile applications, including audio, video, and data streams. Eliminating the need for native plug-in and app installation makes these connections easy to use and compatible with all major browsers and mobile operating systems.
WebRTC adoption in the tech community has grown dramatically in recent years. Facebook, Amazon and Google are among the largest technology companies that implemented WebRTC Development to make their web applications faster, more reliable, and more secure.
WebRTC functions are also provided in standard solutions that can be easily integrated with other software. A good example is OpenTok, a PaaS for live communication, courtesy of our business partners at the old TokBox (now Vonage). We use it successfully in a number of solutions for our customers, including an advanced authentication service based on biometric technologies.
As already mentioned in the summary, the key feature of WebRTC is that it is a simple yet complex technology. The essence of simplicity comes down to ease of implementation. It is enough to use five to ten lines of code to Organised peer-to-peer video communication between two browsers. if it does not work correctly. Also, to get the desired result, you need to know STUN, TURN and NAT.
STUN is a Standarised set of methods, including a network protocol, for crossing network address translator (NAT) gateways in voice, video, messaging and other interactive real-time communication applications. Why do we need this?
STUN is Compulsory when we need to connect two browsers that do not have external IP addresses. They both connect to the server and find out your IP. Browsers exchange these ports through which they are related to each other.
TURN does pretty much the same thing. It automatically sends traffic. This traffic is not modified or altered in any way. This approach allows us to connect two points while operating over TCP (a more reliable protocol but slower than UDP). It should be noted that about 15% of calls cannot be made without TURN.
Now that you know what WebRTC is, let's dive into the story to understand when and how the technology appeared and in what cases it can be used. Additionally, we will discuss the pros and cons of the technology, examples of WebRTC solutions, and high-demand WebRTC applications. The biggest advantage of technology is that it saves time. Since we are ready to complete a task in less time, we will use the saved time for other important activities. With the help of technology, many activities like cooking, cleaning, working and traveling are done faster. these applications rely on peer-to-peer communication. If we need to organize group calls and live broadcasts, it is imperative to use a server that works as a protocol client.
- Saves time
- Better communication
- Developed better learning methods due to technology
- Easily spend our time
- Improves efficiency for Business
- We will communicate more efficiently due to technology.
How does WebRTC work?
The primary focus of WebRTC is to provide real-time audio and video communication between participants, who use a web browser to initiate conversations, locate each other, and bypass firewalls.
WebRTC uses JavaScript and HTML5 APIs, and integrates into a browser. The distinctive features of the WebRTC application are as follows:
Send and receive streaming audio and video.
Retrieve network configuration data, for example, IP addresses, application ports, firewalls and NAT (Network Address Translator), that are needed to send and receive data to another client using the WebRTC API.
Open/close connections and report errors.
Stream media data, for example image resolution and video codec To receive and send data stream, WebRTC provides the Various APIs that can be used in web applications.
RTC Peer Connection for audio and video streaming, encryption and bandwidth configuration RTC Data Channel for generic data transmission Media Stream to access multimedia data streams from devices such as digital cameras, webcams, microphones or shared desktops The Internet Engineering Working Group and the Web Real-time Communications Working Group are developing a set of standards for the use of WebRTC in software.
Published by Capanicus
