OnSIP

At OnSIP, we are passionate about Real Time Communications (RTC) for business. We like to blog about technology, tools, and tips to help small and medium sized businesses succeed. Connect with us on Twitter and Facebook!

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WebRTC and HTML5: An Overview

WebRTC is comprised of three HTML5 APIs - getUserMedia, RTCPeerConnection, and RTCDataChannel - that are built into Chrome and Firefox. In this sense, WebRTC and HTML5 are inseparably linked. But WebRTC developers will actually spend a majority of their time coding in JavaScript. The WebRTC APIs essentially allow JavaScript to generate real-time video and voice streams that can be treated like standard video and audio objects in HTML5. This gives developers a chance to incorporate streaming media into their applications using nothing more than HTML5 capabilities that are already built into Chrome and Firefox browsers.

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Topics: WebRTC

WebRTC Development

WebRTC development, like any other form of software development, is shaped by the technical possibilities of a given technology. In the case of WebRTC, the constraints are relatively simple. The code is in JavaScript, the functionality is powered by three APIs built into Chrome and Firefox, and the mechanism that connects WebRTC peers (i.e. browsers) is generally powered by a pre-built signaling platform such as OnSIP's. Coding is of course handled by the developer. But WebRTC’s built-in browser functionality and OnSIP’s pre-existing signaling network take care of the other two components of WebRTC development.

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getUserMedia - WebRTC Explained

Out of WebRTC’s three APIs - getUserMedia, RTCPeerConnection, RTCDataChannel - getUserMedia is ultimately the one that allows browsers to capture streaming voice and video in a few simple JavaScript commands. The real-time data that users generate with a computer’s microphone and webcam are retained by getUserMedia via a high-level JavaScript command. This lets the browser take care of the tricky backend procedures (such as retention, rendering, codec management) on its own.

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WebRTC, Google, and the W3C

WebRTC offers developers unprecedented capabilities for streaming voice and video data through the browser. The three WebRTC APIs - getUserMedia, RTCDataChannel, RTCPeerConnection - work together to capture, relay, and render the input that comes from a computer’s webcam and microphone from one browser to another. From its inception, WebRTC has been backed by some of the web’s leading development teams. But Google, more than any other company, has paved the way for WebRTC’s success. From its inception in the Chrome team’s lab to its current status as a game-changing technology, WebRTC has relied on Google’s backing to sustain its growth. Google and WebRTC, for better or worse, are inseparable entities at this point. This of course has ramifications for WebRTC’s codification at the W3C.

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Chrome and WebRTC

45.22% of all internet users employ Google Chrome, which makes it the most used browser in the world.1 This figure is over twice as much as Internet Explorer’s 21.43% market share (a number that has dropped nearly 10% in the past year), and more than Firefox, Safari, and Opera’s combined share (29.8%). WebRTC and Chrome have become inseparable since the Chrome team released its WebRTC API in 2011. But besides its closeness, WebRTC is also beholden to Chrome because the browser represents WebRTC’s largest adoption opportunity.

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Android and WebRTC

One of the most exciting prospects for WebRTC adoption continues to be the mobile market. WebRTC on Android has already made serious headway into this field. As of now, WebRTC is supported by the Android versions of Chrome, Firefox and Opera. Together, these browsers currently account for more than 31% of all mobile browser usage today. Safari alone accounts for 40%. This is why Apple’s support for WebRTC is being sought, even though they are a fairly minor consideration when it comes to desktop browsers.

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Topics: WebRTC

WebRTC Server

WebRTC applications are executed entirely “in-browser” and do not require the end user to install any software to run. But this does not mean that developers will deal with just the browser when they are building WebRTC-based applications. WebRTC server-side solutions must be implemented by developers to get peers (i.e. browsers) to communicate with each other. In this sense, WebRTC-server side considerations must deal not just with connecting peers across the internet, but with ensuring that packets containing crucial metadata are able to traverse the NATs and firewalls that a given endpoint might be behind. This is where existing SIP signaling platforms, such as OnSIP's, come in handy for WebRTC developers.

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OnSIP for WebRTC Developers

WebRTC: The Next Innovation in Communication

WebRTC is a browser technology, backed by Google, Mozilla and Microsoft, which enables real-time communication between users. Since it's built into the browsers (no downloads or plug-ins), users can make voice and video calls, send messages and transfer data, all without having to worry about compatibility. This is nice for catching up with your mother, but even better for customer-facing businesses and application developers. Many businesses have already adapted WebRTC, such as Amazon's Mayday button and SnapChat's video chat feature.

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Topics: WebRTC

Study: SIP Trunking Shot Up 50% In 2013

Infonetics is an international market research and consulting firm that helps businesses plan, strategize, and compete in the communications market. It recently released the VoIP and UC Services and Subscribers report for 2014, a comprehensive overview of service providers' revenue and subscriber bases. The study documented, most pertinently for OnSIP, a remarkable 50% increase in SIP trunking for 2013, with the highest growth taking place in North America.

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Creative Uses for VOIP

VOIP isn't just for serious people trying start their own telephony service. There are a lot of fun and creative ways that you can include voice, video, and text messaging in your application.

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