How to use WebRTC to make a video chat application?

Due to the pandemic, there are many more professionals working from home than before, and businesses are increasingly embracing online conferencing as their main channel for connecting with clients and staff.

Due to the social distancing measures enacted over much of the world and the general state of lockdown, these video conferencing systems are currently utilised by almost everyone for events and corporate meetings.

Building real-time video chat applications is now simpler than ever thanks to WebRTC and the growing ability of browsers to support peer-to-peer connections in real time. We’ll look at SimpleWebRTC in this blog and see how we can leverage the platform to build WebRTC technologies.

Web Real-Time Communication is known as WebRTC. It allows peer-to-peer communication without the use of a server and allows the transfer of data, audio, and video between the connected peers. With WebRTC, the server’s function is restricted to simply facilitating the establishment of a direct connection between the two peers.

No third-party software or plugins are required for this technology. It is open-sourced, and www.webrtc.org offers access to its source code without charge.

Signalling in video chat apps is considered as one of the most crucial stages, in which before communication the two peers must assure the information about one other to connect.

This information may include:

• The update about the availability of any other peer for connection
• Information such as peer’s IP address and port
• Control messages – being used to start and end up established communication
• Error messages
• Important data that is required for secure connections

All the above mentioned information’s are called metadata that are the must have for any direct connection to be placed. For signaling, server availability is a must-have requirement.

The process of signaling is used to begin initial communication between the two browsers. This process is done so that they can find other present peers and send the information that is required to create a direct connection among them. This signaling process continues until the direct connection is established.

Based on codecs, WebRTC functions. These methods are employed in the compression and decompression of data, audio, and video. You can communicate audio and video content with low latency using a variety of codecs using WebRTC. SRTP, the safe and encrypted variant of RTP, is used by WebRTC to process and send media via the network using well-known VoIP mechanisms.

The steps listed below must be followed in order to generate a replica of the WebRTC video chat application:

• Create the client application.
• Create a basic chat application with AWS Websockets
• Now, run an Ubuntu AWS EC2 instance as the Stun/Turn Server.
• Utilize the server less framework
• You’re ready to launch your web application for video chat!

Contact E-Alphabits if you want to construct a WebRTC-based clone of the video chat web app. With more than 13 years of expertise, we think that by adding value, we can produce things that are incredibly useful. By integrating WebRTC into your video chat application, we ensure the success of your product.

Get in touch with us at E-alphabits or sales@ealphabits.com to discuss developing your own video chat application.