The COVID-19 pandemic has affected nearly everyone’s way of life, irrespective of their age. Connection with family and peers brings fundamental psychological and social benefits to everyone. The loss of in-person social interactions caused by the lockdown, school closures, and social distancing during COVID-19 has strained these developmental processes. Whether it’s work, school, education, or talking to your doctor, everything has been restricted due to the protracted nature of the pandemic. Communication is vital to share the right information needed to make sure businesses continue during the crisis.
During the lockdown, we have seen various platforms available to ensure virtual communication. These include SignalR, XMPP, Twilio, Discord, Jitsi, WhatsApp Business, Skype, Zoom, Hangout, Webex, etc.
Many of these are UC platforms which provide end-to-end communication solutions. Many platforms offer the facility to enable solutions on top of their existing applications. WebRTC is one such platform.
WebRTC (Web Real-Time Communication) is a technology that enables web applications and sites to capture and optionally stream audio and video media and exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprise WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.
WebRTC consists of several interrelated APIs and protocols which work together to achieve this. The documentation you’ll find here will help you understand the fundamentals of WebRTC, how to set up and use both data and media connections, and more.
Important APIs which we should know:
Media Capture and Stream API
This API provides powerful multimedia capabilities to the web, including support for audio and video conferencing, file exchange, screen sharing, identity management, and interfacing with legacy telephone systems
https://developer.mozilla.org/en-US/docs/Web/API/Media_Streams_API
Signalling and two-way calling
WebRTC is a fully peer-to-peer technology for real-time audio, video, and data exchange.
https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling
Some of the notable use cases for WebRTC based solutions are:
- Making voice and video over IP calls from browser to browser.
- Chat
- Secure file transfer
- Screen sharing
- Integration with web apps using API for contextual information sharing
- Secure peer-to-peer networking
While using WebRTC, we need to take care of:
- Code updates as per changes in browser and upgrades
- Integration with Legacy communication systems
- Limited WebRTC skilled resources
- Multi-vendor, Multi services involvement
- UI/UX and dependency on the browser
How start-ups and business can leverage WebRTC in the current scenario:
- Real-time marketing
- Real-time advertising
- Back office communications (CRM, ERP, SCM, FFM)
- HR management
- Social networking
- Dating services
- Online medical consultations
- Financial services
- Surveillance
- Multiplayer games
- Live broadcasting
- E-learning
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