I’m willing to bet a week and a half of steak lunches that you’ve at least heard the term WebRTC. Otherwise, why’d you click?
WebRTC is nothing new. It’s something old(ish) done in a new and exciting way.
Let me explain.
When I moved to the other side of the continent, my parents discovered Skype. With that discovery, they started asking distant friends and family, “do you have Skype?” The basis of the question is the fact that in order to hold a conversation over Skype, all parties involved need to have Skype installed on their computer and be in possession of the login credentials to a Skype account. That’s what WebRTC is here to change.
Imagine if when you clicked on the link to this article, instead of discovering this tome of text, you discovered my smiling face, live and ready to converse with you, interactively, all about WebRTC. WebRTC enables real time conversation through your browser without the need for additional software or plugins. You don’t need to ask who has Skype anymore, because it doesn’t matter as long as they’re using a WebRTC enabled browser. And it’s estimated that there are now one billion WebRTC capable devices in the world (predicted to reach three billion by 2016).
Well what’s wrong with plugins?
Plugins need updates. They don’t work with every platform. And they require the downloading of files from the Internet, which could bring destructive malware along for the ride. Plus, if the party you’re trying to reach doesn’t have the plugin, your conversation isn’t happening or at very best is delayed. Especially because in some organizations, not everyone has access to installation privileges. Or, in the case of my parents when we tried to migrate them off Skype to Facetime, not everyone knows how, and has to wait for their daughter to come over and perform the install.
OK, real time, plugin-less, browser-based communications, so what?
Live, technology agnostic (as soon as all browsers get set up) connections are easier, cheaper (freer), and more reliable than ever before. WebRTC enables instant voice and video calling through your computer based connection to the world, which is awesome for collaboration. It’s consistent across platforms, so compatibility isn’t an issue. And it doesn’t bog down your network or your applications because it’s lightweight, partly because it is meant to connect endpoints directly without routing audio through a webserver. And to top it all off, the standards consistency and architecture ensure quality audio and video transmission.
Sold? Good. But what do I mean by, “make the most of it”?
Browser to browser communication is great, assuming all parties are on a browser. But what if you want to connect with an analogue endpoint, one of those old-fashioned desk phone thingys? You need a legacy inclusion bridge. In this case, a SIP connection.
No other technology amplifies WebRTC like a SIP trunk does. Connecting your WebRTC to SIP extends your communications possibilities exponentially.
Modules being added to existing PBX technology, like FreeSWITCH, translate WebRTC coding into an audio format that can be transported over a SIP connection. And since both SIP and WebRTC use the G.711 audio codec, it’s a perfect fit.
What this means for you and your business is that the possibilities of how you communicate internally, and externally with clients, suppliers, and prospects, just got a whole lot more efficient, easier, cheaper, and extensive. In a world bustling with more and more new modes of communication seemingly daily, it pays to be able to tie them all together in a simple way until we’ve migrated completely away from the older ways in a few short decades.
The full potential of WebRTC is yet to be realized. But hoards of brilliant minds are dedicated to discovering and designing everywhere it will take us. One certainty is this, if you aren’t integrating SIP and WeRTC, you’re already behind the times, and the competition.