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Announcing The OnSIP Network: We've Slain Those Signaling Dragons for WebRTC Developers

by Kevin Bartley

The OnSIP Network to developers is a Platform as a Service offering that allows WebRTC developers to add the vital signaling layer to their apps.

Published: April 28, 2014

You may have noticed some changes on our website— We've added a whole new section For Developers. That's because OnSIP is happy to announce The OnSIP Network, a Platform as a Service that enables developers to add the vital signaling layer to their WebRTC applications in a matter of minutes.

The OnSIP Network locates and negotiates communications between WebRTC application peers, solving a similar problem that the Public Switched Telephone Network once did for telephones.

WebRTC developers can easily access The OnSIP Network using SIP.js, an open source JavaScript SIP stack by OnSIP. Watch this quick 60 second video to learn more:

Why The New Offering

Launching The OnSIP Network was a logical next step for us. Known for our Hosted PBX service, OnSIP has always been a bit different under the hood. We've leveraged a variety of open source projects (OpenSIPS, FreeSWITCH, and Cassandra, to name a few) to build a geographically distributed signaling network based on open standard SIP. Our Hosted PBX applications and customers' phones are distributed on top of that network— In essence, we look a lot more like a highly-available, real-time platform (think social media networks and robust e-commerce platforms) than a collection of PBX servers "in the cloud" to which businesses are tethered to.

Now supporting hundreds of thousands of SIP and WebRTC endpoints, we are excited to help the developer community and businesses create applications that push the limits and change the way people communicate every day, without worrying about the infrastructure costs associated to building and maintaining a signaling network.

The Importance of a Signaling Network

On the forefront of Web Real-Time Communications (WebRTC) development, WebRTC has been a recurring topic in our blogs, newsletters, and products. WebRTC has been a breakthrough open source project by Google. It enables secure, real-time voice, video, messaging, and IM in the browser without downloads or plugins. However, whenever businesses dream of real-time communications applications over the web, they quickly face a massive hurdle: a signaling solution.

“While WebRTC defines a set of protocols that handle flow of real-time communications between browser peers, it does not define a signaling protocol to help browser peers locate others and initiate communications,” said John Riordan, OnSIP CTO in today's press release. “Session Initiation Protocol (SIP), is a time-tested open standard, but building a network to route signaling over the Internet still requires significant investment in infrastructure and expertise.

Our service helps solve that issue by handling the heavy lifting of a signaling network and offering SIP.js, an open source JavaScript SIP stack, to access it. Leveraging these, the developer community and businesses building with WebRTC will benefit from faster and more affordable product development and maintenance.”

We have always been avid open source fans. As such, our team is especially excited about our first open source library contribution, ever, SIP.js (a fork of JsSIP). "SIP.js is more SIP-centered than other JavaScript libraries," said OnSIP Software Engineer James Criscuolo. "By changing both the code structure and the coding terminology of the prior library, we were able to make SIP.js more SIP-compliant. We were also able to separate the original code between client and server, and simplify event emissions, which streamlined the process of incorporating projects such as GetOnSIP and InstaCall into our mature SIP platform."

OnSIP remains committed to offering the transformative power of in-browser solutions to developers, existing customers, and the broader WebRTC community.

 

Illustration credit: Sococo. Thanks for the blog title inspiration!