What Is WebRTC And How It Works? Ultimate Guide

What is WebRTC?

Google created Web Real-Time Communication (WebRTC), a streaming project. This open-source project was started to assist Google in acquiring Global IP Solutions, a video conferencing and VoIP technology company, in 2010. The WebRTC project was launched the following year.

WebRTC, an open-source project, supports real-time video conference over browsers and applications. This project is made possible by several standards and protocols.

WebRTC technology is built on the foundation of early VoIP technology. VoIP is short for Voice Over Internet Protocol. It stands for voice over the internet.

How Does WebRTC Work?

WebRTC handles two important aspects of peer-to-peer conferencing. It is responsible for media capture from your device.

WebRTC, which is the technology that tells your phone to start recording, is WebRTC. It is also responsible for transmitting data between the devices.

WebRTC’s foundation is a set of JavaScript APIs. These APIs are “getUserMedia”, “RTCPeerConnection” and “RTCDataChannel”.

WebRTC Programs

WebRTC powers many major programs you may have used in the past. These include:

• Google Meet
• Google Hangout
• Slack
• WhatsApp
• Discord
• Facebook Messenger
• GoToMeeting
• Snapchat
• Houseparty

WebRTC Video Streaming: Advantages

• Ultra-Low Latency Video Streaming. Latency is 0.5 seconds
• Platform and device independence
• High-quality voice and video
• Secure voice and video
• Scale easily
• Flexible to network conditions
• WebRTC Data Channels
• Interoperability With Other Technology

How Does WebRTC Work?

• WebRTC transmits data directly between browsers. It is also known as “P2P”.
• It can transmit audio, video, and data in real-time
• To allow browsers to connect, it must use NAT traversal mechanisms
• P2P must be sent through a relay server (TURN).
• WebRTC requires you to consider signaling and media. They are all different.

— The tech allowed peer-to-peer communication (browser-to-browser), without the need for a server. This is a significant feature that was not usually possible with signaling.

— Video calls can be held with multiple people using peer-to-peer communication. A media server reduces the number of streams that a client must send to one and can also reduce the number of streams that a client must receive.

• WebRTC Servers You’ll Need

— Signaling server

— Servers for STUN/TURN

— Media servers (It’s up to you what use it for)

WebRTC: Why You Should Choose It

• WebRTC is an open-source technology

— It can be used privately or for commercial purposes. That alone is sufficient. It is always evolving, so it will continue to meet your needs for many more years.

• It is well supported

— WebRTC is now supported by nearly all browsers and can be used in any usage scenario that you can think of.

• WebRTC is also available to mobile applications

— It is used in many mobile apps. There are no limits to the number and quality of SDKs that can be purchased.

• It’s not just for video or voice chat

— WebRTC is a peer-to-peer communication tool. However, you can also use it for group calls, to build a video conference solution, and to add the recording.

Is WebRTC secure?

Security can be compromised in many ways by a plug-in or real-time communication app. Here are some examples:

• Unencrypted media and data can be intercepted as they are being transmitted between browsers, or between a browser to a server.
• Audio and video may be recorded without the user’s consent.
• A seemingly innocent plug-in or program can become infected with malware or viruses.

How WebRTC solves these security issues?

• WebRTC uses secure protocols such as DTLS or SRTP.
• All WebRTC components must be encrypted, even signaling mechanisms.
• WebRTC does not function as a plugin. Its components are run in the browser’s sandbox, and not in separate processes. The components do not need to be installed separately and can be updated as soon as the browser is updated.
• Camera and microphone access must always be granted. This is made clear by the user interface when the microphone or camera is in use.

Conclusion -

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