VoIP Resources VoIP Fundamentals WebRTC Essentials

WebRTC Video

by OnSIP
⏱1 minute read

Learn about the general specifications of WebRTC video and how to use OnSIP to stream video with WebRTC.

WebRTC Video Capabilities

Real-time video is the lifeblood of communications applications such as Skype, a program in which the computer’s video camera captures visual data and sends it to another user. But programs such as Skype require the user to download software to implement this real-time video exchange. Perhaps the most impressive component of WebRTC is its ability to stream this kind of real-time video without requiring any additional plugins or add-ons for the internet browser. WebRTC video capabilities are built directly into the browser itself, and they do not require anything beyond a Chrome or Firefox browser to run.

WebRTC Video: General Specifications

WebRTC video capabilities are powered by VP8, an open source video codec maintained by Google. The software is completely free to implement, unlike Flash-based alternatives that require licensing fees to run. VP8 and the audio codecs Opus and G.711 are the only codecs WebRTC needs to run. Although the official W3C specification is a work in progress, WebRTC’s fundamental architecture (including codecs) have been agreed upon the by various groups (Google, Mozilla, et al.) involved in its drafting. This means that different types and versions of Chrome and Firefox can all handle the same WebRTC video stream without requiring the user to amass a hodgepodge of media plugins.

What kind of quality can one expect from WebRTC video streams? Limits on frame and data rates for VP8 are nonexistent. The width and height measurements come out to 14 bits each, which allows for a maximum resolution of 16384 x 16384 pixels. Its quality has been compared favorably to H.264, a proprietary video codec that requires licensing fees. But why not take a look with your own eyes? Try GetOnSIP, our WebRTC-based video chat, to test its video capabilities for quality.

Using OnSIP to Stream WebRTC Video

The ability to craft real-time video applications using built-in HTML5 browser functionality has changed the landscape of streaming media applications. But in order to harness the potential of WebRTC video capabilities on a wide level, developers must adopt a signaling solution that connects peers (i.e. browsers) to each other. OnSIP has a massively scalable, redundant, SIP-based signaling platform that has fully incorporated WebRTC functionality into its architecture. With a purposely undersubscribed platform, OnSIP offers speedy and quality delivery of WebRTC video data without the worry of lag or congestion. OnSIP's platform allows you to focus on coding your awesome in-browser video application, while we step up and take care of the heavy lifting.

Learn more about VoIP Fundamentals