WebRTC or Web Real-Time Communications is the latest technology that has piqued the interest of businesses and corporate across the world. The technology, supported by W3C (World Wide Web Consortium), enables real time communications like voice, data, video and instant messaging through web browsers that follows the open standard. Majority of the new technologies project money-saving as their USP but Web RTC helps the developers to create customized communication systems through integration of the technology with other applications. This automatically cuts down the need for other communication software. WebRTC is supported by the popular browsers like Mozilla and Chrome whereas Internet Explorer and Safari do not. Apple and Microsoft are presently developing browsers that would actually support Web RTC in the near future. Interested in some other areas that are related to the future of VoIP? Check these out:
WebRTC and Voice Calls
Web RTC is basically an application programming interface that the vendors of communication systems must provide to make them really useful for their users. There are many services that helps the users connect two browsers for making a call. One of the most popular users of WebRTC is WhatsApp as it helps to connect two users working on different browsers. The web users will be able to connect each other with the help of rich video calls.
3CX is another application that has used it from the time of its inception and the company still invests in the developments. The users of 3CX do not need any other software to make and receive calls. This is far better and efficient than the other VoIP services where the users need to have a software or application like Skype, Viber etc. The users will be able to make calls online by using the application like a web page. Video and web conferencing technology is fast developing but with the user Web RTC they will be able to bypass the additional software and hardware requirements and could even do away with the proprietary standards. There is a revolution in waiting with the Web RTC becoming a standard among users who would like to have a one point solution to their calling needs.
Does this ring the death bells for VoIP?
Well, it would be a little premature to say that Web RTC is going to eat away the technological space created by VoIP. This technology is still in its nascent stage and it is very difficult to say whether it actually can have any effect at all on the VoIP market share. There are numerous issues like jitter, call quality, security and latency that might or might not turn up to be a challenge. All these factors will be dependent on how secure, fast or efficient the user’s internet connection is.
The voice codecs supported by Web RTC is different from the ones that the SIP or Session Initiation protocol supports. H264 is the codec that is commonly used for video whereas VP8 is used by Web RTC. There are also differences in the signaling protocol of WebRTC as it is not yet defined but there are some protocols that are common: media transport uses Skinny Real-Time Transport Protocol (RTP) and RTP Control Protocol; for security it uses Secure RTP; for NAT it uses Session Traversal Utilities; for net it uses Interactive Connectivity Establishment and Relay NAT.
It cannot be said for sure that it will replace VoIP but there is certainly going to be some jostling for space in the coming years. The retailers and vendors will definitely speak against the use of Web RTC but the truth is that the day is not far when people will seek solution like international VoIP blocking which can only be provided by Web RTC.
About The Author
Michelle Patterson is excited with the new technologies that are threatening to change the way we stay in touch and communicate, particular in business. She works with companies that are introducing these technologies to make understanding them easy for regular people.
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