WebRTC: Exploring its capabilities

WebRTC

In May 2011, Google released an open-source project for browser-based real-time communication known as WebRTC. This has been followed by ongoing work to standardize the Communication protocol and browser APIs in The World Wide Web Consortium (W3C).

WebRTC includes a set of web APIs and communication protocols for peer-to-peer bidirectional, continuous audio/video communication between online browsers which eliminates the need for an intermediary server.

WebRTC Can be used with HTTPS only.

WebRTC is typically used in combination with WebSocket. WebSocket’s primary purpose is to exchange SDP (Session Description Protocol) and network information between the client and the server. Websocket has nothing to do with actual communication data and is simply used to let WebRTC connect.

WebSocket

The WebSocket API is an advanced technology that makes it possible to open a two-way interactive communication session between the user’s browser and a server. With this API, you can send messages to a server and receive event-driven responses without having to poll the server for a reply.

What is the purpose of WebRTC?

WebRTC’s purpose is to make real-time P2P communication over the internet easier. WebRTC has a variety of applications, including the following:

• WebRTC is utilized in video calling systems like Zoom, Microsoft Teams, Slack, and Google Meet for video conversations and meetings.
• For real-time media file transfer, WebRTC is widely used.

WebRTC and Why Is It So Important?

• Most major web browsers, including Google Chrome for desktop and Android, Mozilla Firefox for desktop and Android, and Safari, support it
• it can adjust communication quality, bandwidth, and traffic flow whenever network conditions change
• Simplifies peer to peer communication
• Low bandwidth consumption: Peer-to-peer communication results in low bandwidth consumption and reduced latency that users may face while sharing files.
• Low latency broadcasting

Popular Projects Based on WebRTC

• Jitsi Meet: Open Source Real-time Video Conference
• Snap Drop: Open Source Local Network File transfer using WebRTC (iDrop Clone)
• OvenMediaEngine: open-source real-time streaming server
• OpenVidu : facilitate the addition of video calls in your web or mobile application
• JSCommunicator