OnSIP has launched the OnSIP Network today, a signaling Platform-as-a-Service (PaaS) that provides API-enabled access to platform infrastructure that can manage WebRTC applications. ProgrammableWeb spoke with John Riordan, CTO of OnSIP prior to today’s launch.
OnSIP has released a signaling platform that, they say, does for web video calls what the PBX system did for business telephone systems. Using an open source API — written in SIP.js — developers can build WebRTC applications without having to manage their own signaling platform.
To offer web video chat directly from their websites, John Riordan, CTO of OnSIP, says developers “need to solve the signaling issue for their application. We have always been oriented towards delivering peer to peer connections. So, we have tens of thousands of users, making use of hundreds of thousands of endpoints: so we have done a lot of the heavy lifting to create that signaling platform and are now expecting that webRTC developers will see it as a good option instead of building their own.”
Accessing OnSIP via API
The OnSIP platform can be accessed via an open source API. Riordan explains:
For developers, there is a tab on our home page which will take you down a different pathway, with documented getting started guides. You can use any SIP Endpoint with our platform. We can even do SIP over web sockets.”
Alongside the OnSIP network access API, an admin API has also been released to let customers monitor usage of their WebRTC applications. “We have an admin API that lets you do call routing, billing information, access our hosted PBX services, all that stuff is available from an admin-based API,” says Riordan. “
While the OnSIP Network is currently free to access (with the SIP.js set to remain an open source project), OnSIP does plan to introduce industry-level pricing later in the year. In the meantime, the Admin API can help developers and business customers monitor their usage rates to assess the future return on investment of providing web video chats using a volume-based charging model based on minutes and number of endpoints.
Use cases for WebRTC: From sales and support to emergency medical advice
The launch today has also included the announcement of the InstaCall feature, which lets developers embed an instant video chat button on their websites. “There are three steps in signing up, and after that you can start making calls and receiving calls. If that’s our ‘hello world’, thats how long it will take,” Riordan says.
The service is already being used by sales and support services. In their press release for today’s launch, OnSIP points to a business management software company that supports the floor covering industry. They are using OnSIP in a similar manner to the ‘Amazon Mayday’ functionality, noting a “huge uptick in customer satisfaction.”
“That is an easy one for us: hosting video sales and support calls and augmenting that experience for their customers through this feature. It is an easy, straightforward approach for us, and our customers can add an 800-number to their website,” Riordan says. “But we have also started working with customers developing on our platform. In the medical field, for example, the OnSIP platform WebRTC applications allow field emergency responders to communicate with a burn center so before they move or touch a burn victim, they can get medical staff advice on how to proceed with supporting that burn victim.”
The growth of WebRTC
Developer interest in WebRTC technologies has been increasing over the past year, with options from PubNub and TokBox released in the past six months. Riordan credits the ease with which video chat can be viewed on people’s computers as a main cause. “Certainly the more recent releases of Chrome and Firefox have been a real boon for creating working applications,” Riordan explained. “Even though the standards are not really standards yet, the components are out there and interest seems to be accelerating.
“We’ve been trying to get software-based solutions for some time. We used Java at one point, and certainly think developers at Google and Mozilla have been great at implementing the low level video and audio to work well in their browsers. And people’s laptops and processing power has also helped.”
“We’re really excited to expose our platform to the public and to the WebRTC community in particular,” Riordan said.
By Mark Boyd. Mark is a freelance writer focusing on how we use technology to connect and interact. He writes regularly about API business models, open data, smart cities, Quantified Self and e-commerce. He can be contacted via email, on Twitter, or on Google+.