WebRTC World Feature Article

April 28, 2014

OnSIP Rolls Out InstaCall, WebRTC-based Solution based on The OnSIP Network

OnSIP is taking its business phone service to the next level with The OnSIP Network and WebRTC-based InstaCall solution. We first met InstaCall back in February, when the company announced the beta release for sales and support teams to embed video calling functionality into their websites or applications. Today, the company is officially rolling out InstaCall, which runs on The OnSIP Network, a signaling Platform as a Service (PaaS) for developers to build scalable WebRTC solutions.

The company is also working on an open-source API SIP.js so developers can access The OnSIP Network in their applications. We caught up with John Riordan, OnSIP CTO, to discuss the solutions, the challenges they help developers overcome and how this aligns with OnSIP’s role in the industry.

With InstaCall, developers can add WebRTC functionality to a website or app in less than five minutes. It works as part of OnSIP’s hosted PBX offering, enabling developers to place buttons on existing websites and field calls from their customers.

“We try to make it as easy as possible to create,” Riordan explained. The solution also uses the hosted PBX admin panel, which was also designed with ease of use in mind and a “what you see if what you get” type of interface.

InstaCall is targeted at people who are operating websites and want to get buttons on their site quickly and easily. This application is just one example of what is possible with the underlying platform – The OnSIP Network. The OnSIP Network supports whole signaling infrastructure to help developers set up all kinds of real-time communications and embed that into their existing applications. Riordan explains that leveraging SIP will enable WebRTC endpoints to talk to each other.

OnSIP focuses on Internet-based peer-to-peer (P2P) communication, especially as more and more communication between businesses and customers happens over the Internet as opposed to traditional phone networks.

Riordan explained that WebRTC and HTML5 are helping the future move along in a rapid way. He also explained that one key piece that has not been included in WebRTC standards is handling signaling. He says it’s definitely possible for developers to build their own, but it’s a lot of work and there are issues when it comes to solidifying a demo-proof concept that can be distributed reliably.

To help with this challenge, OnSIP is supporting an open-source project SIP.js, which is giving a signaling library for free that can be used with OnSIP’s platform or others. It’s not required to use with OnSIP’s platform, but it helps developers get going, Riordan said.

The company also has a patent, “System and Method for Geographic SIP Scaling,” which falls in line with how the company is distributing SIP calling. 

Riordan will be discussing WebRTC in more detail at the WebRTC Conference & Expo in Atlanta, Ga., happening June 17-19, in two sessions: “Deploying WebRTC Successfully – The Big Issues” and “Using WebRTC to Deliver Enterprise Services.” 

Edited by Maurice Nagle


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