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This guest post about Web Real-Time Communication API (WebRTC) is written by Alexey Aylarov, CEO and co-founder of Zingaya, the Russian startup, which enables placing voice calls from a webpage. Follow him on Twitter, or Google+.

Background

Most of us nowadays still use traditional phone services, such that when you are making a call from your phone it will be connected using either PSTN (if a landline) or through a mobile network (if you’re on your cell). Of course, a lot of people have started using VoIP during the last decade, and new technologies like SIP (or services like Skype) have been major drivers of VoIP usage growth all around the world.

But it’s not only about software when it comes to VoIP. You’ll need to have a good network with sufficient bandwidth and some hardware too, since a microphone is required to capture audio and send it as IP-packets to the Internet. As networks have been evolving quickly, broadband connections became available even for mobile users with the new generation of mobile networks (3G, and now LTE). Devices were evolving too; first, we saw how laptop sales surpassed desktops and now we see the blooming smartphones and tablets market that shows new records every quarter. There’s a very important commonality amongst these devices – all of them have built-in microphones and capabilities for high speed access to the Internet. Because of the evolution in the microchip industry, the latest smartphones (like the iPhone5) has power similar to the super computers from the previous century, so encoding and decoding audio and video is a piece of cake for them, even in HD quality. All of these sets the groundwork for the next breakthrough: in the telecommunication industry.

The shift

VoIP technologies can get a bit complicated when it comes to audio and video codec implementations, the patents around them, and the standardization of all the new ideas that the industry tries to adopt. There tend to be a lot of interoperability questions too. To understand the scale of experienced VoIP developers versus experienced web developers, the ratio would be look like the population of Luxembourg for the former, and the population of China for the latter. And so it was for a long time until Google hasn’t decided to buy Global IP Solutions (GIPS) and open most of its IP for developers to start new exciting project called WebRTC.

In the VoIP world, the client application used to make and receive calls is usually referred to as the User Agent. (For example, the Skype application is a User Agent for its network.) There are a lot of other User Agents for SIP networks (since SIP is an open standard that anybody can implement as a SIP User Agent, according to SIP specification). Different companies were developing different applications for their purposes – some of them were using open protocols and standards, some weren’t. It was the same situation with audio and video codec support in different User Agents. In the end, we have a plethora of different applications, and many of them can’t work with each other because of interoperability issues (perhaps due to developers not focusing on that aspect of it, since there was no such intention from the beginning).

With WebRTC, Google has proposed a very interesting idea:  embedding the User Agent inside the web browser. The installation base of web broswers is bigger than the installation base of any other application in the world, but web browsers still lack built-in real-time audio and video communication capabilities based on open standards and open technologies. There are a lot of web browser vendors on the market, but the major players now are Google (Chrome), Mozilla (Firefox) and Microsoft (Internet Explorer). Understanding that it would be hard to ask other vendors to implement WebRTC in their browsers according to some specification created by Google only, Google created a WebRTC working group and started the standardization process of WebRTC in order to make it W3C (for the web world) and IETF (for the telco world) standard. It was a very wise decision and a lot of vendors joined the working group. Google, Mozilla, Microsoft (with Skype), Cisco, Ericsson, Opera, Huawei and many others are involved and it’s really exciting to see them working together. Look for a separate post explaining how this group works and what the decision-making process looks like.

So what will we be able to do with WebRTC when it finally becomes available on our browsers? We will be able to make calls right from the browser, without any download and installation required. This includes audio and video calls between different browsers, calls to real phone numbers, calls to different VoIP systems, etc. Imagine billion of devices automatically turned into VoIP phones – this is the near future. Operators most likely won’t be able to block or control this traffic because it will be hard to understand where it’s used for VoIP calls and where it’s used to stream movies from CDN to your device. More to that point, all audio and video data will be encrypted/decrypted by the browser. Instead of creating a native VoIP app for every platform, developers will be able to create a web application with the same functionality and let people run it in their browser on any device.  And there’s no need to be a VoIP-guru; all low-level and complex processes will be implemented by the browser’s vendor for you. How do you like that?

Adoption

There is no doubt that WebRTC will become a part of our browser (and thus our daily life) as soon as the standard is ready and implemented by browsers’ vendors.  Consumers adopt all new technologies extremely fast if they are useful, convenient and easy to use. WebRTC has appeared just in time: devices, networks and people are ready for the changes this technology can bring.

Timeframe

The ball is in the court of the WebRTC working group, but we most likely won’t see a WebRTC implementation in the release version of browsers until the standard is ready. My best guess: expect it in the 2nd quarter of 2013.

One Parting Thoughts

Although I tried to cover a lot of it, there’s a lot more to WebRTC than I’ve managed to get into this post. If you’d like to learn more please let us know and we will try to write some additional articles and explain different parts of the technology, like protocols, codecs, NAT traversal, P2P-capabilities, etc. You can also follow our blog, where we write different articles and updates about WebRTC there from time to time.

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