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can anyone help me what is the technical difference between WebRTC communication and the VoIP communication?
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WebRTC is thus an extension of VoIP, or a manner of carrying out VoIP. Though it is a variation of VoIP that applies only to the world of Internet browsers, this is a good thing: we all tend to have easy access to an Internet browser. Many even have a browser constantly open at their work posts.
Whereas SIP is a signaling protocol which is mainly used for voice and video calling, WebRTC provides a more versatile option to the end-user which offers SDKs to build powerful mobile applications as well as web applications so the users can literally implement it anywhere.
WebRTC can be seen as an extension of VoIP, bringing the capabilities of making phone calls, video calls, text chat and even Peer 2 Peer file transfers, directly through your web browser (or mobile apps).
WebRTC was built with bidirectional, real-time communication in mind. Unlike HLS, which is built with TCP, WebRTC is UDP-based. This means that WebRTC can start without requiring any handshake between the client and the server. As a result, WebRTC is speedier but also more susceptible to network fluctuations.
The question doesn't exactly make sense because it makes the assumption that VoIP is a technical stack, but it's not - it's a concept. The concept of sending Voice (V) over (o) Internet Protocols (IP). This means that different technology stacks can be used for accessing/capturing the media, establishing connections, negotiating streams, and transmitting streams.
WebRTC is one such stack (set of APIs, methods, and standards) for VoIP.
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VOIP - Voice over Internet Protocol was a concept which came with popularity of internet. This involved using the internet to route voice telephony data, basically using existing IP infrastructure to transport audio streams without having dedicated circuit switched telephony. Over the time popular VOIP applications like Skype, Vonage and many in enterprise telephony came in. VOIP had two parts one signalling , basically controller part and other actual media. Actual media usually but not necessarily followed RTP (Real Time) protocol. RTP could carry both voice and video. Problem with RTP has been that browsers don't support it natively and it is not secure. You usually needed some sort of plugin to have VOIP work inside browser.
With WebRTC now popular browsers like FF, Chrome and Opera support a variation of RTP which is secure and can be natively invoked. Using WebRTC and browser Javascript you can send Voice, Video and Screen (it's video only) data to any other browser, which is cool.
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