WebRTC is a new technology that has the potential to revolutionize how we interact with our web applications. In this post, I’ll cover some of the reasons why you should consider using it in your next project or application.
1) It works on all browsers and platforms. WebRTC is supported by Google Chrome since version 23, Firefox 19+, Safari 5+ and Opera 12+. Most other browser vendors have announced support for the upcoming standards.
For example, Internet Explorer 10/11 will be supporting WebRTC Peer Connection as well; however, not be until IE 11 Beta 2 was released.
2) You can use existing JavaScript libraries like jQuery, AngularJS, Backbone, etc., without having to rewrite them for WebRTC. This is because they already work with HTTP requests and sockets via XHR, which are both used when creating an RTP stream. The only thing you need to do differently is adding one line of code to enable the video element.
3) Your users don’t even know there is something called “WebRTC” inside their browser! And if you told them about it they would probably think it is just another way to play Flash videos online.
There is no mention of anything related to audio-video communication anywhere else in the user interface
4) Unlike most technologies where you have to wait years before everyone starts adopting it, WebRTC is here today. Since its adoption rate seems to grow every day, chances are high that soon enough people will start using it. We may see it become mainstream very quickly.
5) It doesn’t require any plugins or downloads from 3rd parties. All communication between two peers is done directly through the browser itself. No additional software needs to be installed.
You also don’t run into many privacy issues when working with WebRTC compared to traditional SIP-based VoIP solutions such as Skype or Vonage.
6) More than 50% of mobile devices now come equipped with cameras. So instead of relying solely on voice calls, people might eventually prefer using video chat apps over making phone calls too.
With WebRTC, you don’t need the extra hardware required to make those kinds of calls either. All you need is an internet connection and a modern device running at least Android 4.0 Ice Cream Sandwich.
7) Mobile app developers can create cross-platform native apps using HTML5 features. As long as your server supports WebSocket connections, you can easily integrate real-time communications with these apps.
They won’t have to worry about handling different protocols or complex NAT traversal techniques.
8 ) Because WebRTC uses TCP/IP packets, data rates tend to be higher than regular VOIP implementations. Also, unlike some proprietary systems, WebRTC allows for peer discovery so there isn’t much latency involved. That means lower latencies overall.
9) Finally, it’s free. WebRTC has been developed by Mozilla Research Labs, meaning it comes fully open source. Anyone who wants to build upon this technology can freely contribute back.
However, because the standardization process took longer than expected, development costs were covered mainly by companies interested in developing new applications leveraging the technology.
Therefore, while it’s completely free for anyone to implement, it still requires ongoing maintenance fees. But once a company decides to invest in building on top of the protocol, they get access to lots of potential customers and partners. In contrast, with proprietary methods, companies usually have to pay licensing fees upfront to secure their intellectual property rights.
10) Overall, I believe that WebRTC could potentially replace legacy telephony infrastructure for enterprises looking for cost savings and easier scalability.
Although we’re talking about pure IP networks, the same concepts apply regardless of whether the underlying transport mechanism is UDP or TCP. Once deployed properly, WebRTC enables organizations to leverage existing infrastructures and reduce operational expenses significantly.
Conclusion —
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