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.

WebRTC Development - JavaScript and HTML5

The construction of WebRTC applications is powered by JavaScript. This makes the process of coding fairly straightforward for most developers who acquainted with standard web languages. The APIs involved in WebRTC development - getUserMedia(), RTCDataChannel, RTCPeerConnection - are built into the Chrome and Firefox browsers. The commands that these APIs utilize are generally commonsensical [ex: stream.getVideoTracks(), mediaStreamSource.connect()] and are referenced by reputable documentation all across the internet. In this sense, the learning curve for the average web developer is slight. WebRTC was never designed to be an experts-only technology. It’s meant to be used widely by developers of all stripes to solve real-time communications problems in their apps.

In terms of required software, WebRTC development demands nothing more than programs that can read and code HTML5. VIMs and eMacs, or any standard text editor, can be used to write HTML, JavaScript, and CSS. Then Chrome or Firefox can open the code to run the program, while debuggers such as Chrome’s Dev Tools can be utilized to pick apart errors. In other words, WebRTC development requires nothing more than the free, standard programs that most web developers use on a daily basis.

WebRTC Development - Built in Browser Functionality

WebRTC is an open source project spearheaded by the Google Chrome Team. WebRTC’s three APIs are currently built into Chrome and Firefox. By a conservative estimate, 63% of the world’s browsers are WebRTC enabled. There are no fees or licensing requirements for any component of WebRTC development. The W3C specification for WebRTC is pending, but there are no disputes about the overarching design of the APIs. This means that every Chrome and Firefox browser is equipped to stream real-time media without requiring developers to build complex APIs, license or construct codecs, and negotiate communication between different browsers.

Signaling Platform - Putting WebRTC Development Altogether

You’ve built your WebRTC-based application. You have the Chrome and Firefox browser to run the app. Now you just need a mechanism that can get peers (i.e. browsers) to communicate with each other and share media. This is where a signaling platform, such as OnSIP's, enters WebRTC development. OnSIP's scalable, SIP-based signaling architecture, built atop a business VoIP platform services over 20,000 small and medium sized businesses. WebRTC capabilities have been fully integrated into its core structure, allowing developers to use our pre-existing, geographically distributed network to scale applications, bridge compatibility gaps between endpoints, broker connections behind firewalls, and track communications information.

Topics: WebRTC