What was a stuffy world of phone number dialing, slow call routing, and low-quality sound over the Public Switched Telephone Network is becoming a world of instant and rich communications over the Internet.
New York, New York (PRWEB) October 16, 2013
OnSIP is the first hosted PBX provider to combine SIP and WebRTC to deliver instant, free, browser-based video calling to anyone seeking a convenient and secure way to communicate over the Internet. The company’s relaunch of GetOnSIP.com is a demonstration of integrating WebRTC into an already-mature communications platform as well as a starting point for developers to leverage browser-to-browser and browser-to-SIP-phone calling for their own applications.
With over 40,000 users, GetOnSIP is a free service from OnSIP that has historically allowed VoIP enthusiasts to register their standard SIP phones for calling. With the relaunch, GetOnSIP has made it easy for anyone to benefit from secure, high-definition calling with a video phone in the browser, no downloading required. GetOnSIP also boasts easy-to-remember call addresses (SIP addresses) and instant “Call Now” links that GetOnSIP users can share for an instant, high-quality voice and video session with unregistered users.
“This is only the beginning,” said Robert Wolpov, President and Cofounder of OnSIP. “In early advancements of WebRTC, we saw the potential for breakthroughs in telecommunications. What was a stuffy world of phone number dialing, slow call routing, and low-quality sound over the Public Switched Telephone Network is becoming a world of instant and rich communications over the Internet. Our team has been working on integrating WebRTC with our SIP platform, and GetOnSIP was a good place to start sharing the benefits of this technology. However, we’re not finished there. Soon we’ll be offering OnSIP customers instant calling capabilities that will deliver cost savings, productivity enhancement, and improved CRM.”
Headquartered in New York City, OnSIP is a leading provider of hosted VoIP services to over 20,000 businesses. Since it’s beginning in 2004, the company has focused on redefining traditional telephony services with cutting-edge, open standards-based real-time communications.
“Now that most PCs are equipped with a webcam, the ability to send a clickable link to someone to initiate a video call has considerably lowered the bar for how individuals can connect,” said Randolph Resnick, Producer of VoIP Users Conference. “GetOnSIP delivers a stable, peer-to-peer audio and video conference without the need to download any plugins or applications. I have successfully conducted WebRTC to SIP (audio) phone calls and am impressed by the interoperability and stability of this free offering. We are looking forward to hosting the OnSIP Team on VoIP Users Conference next month (November 8th, 12 pm ET) to discuss this milestone and what's next.”
OnSIP is among the leading technology-communication providers developing applications with Web Real-Time Communication (WebRTC), a project open-sourced by Google in 2011 to enable applications with browser-to-browser voice calling, video chat, and P2P file sharing without plugins. The standard is currently supported on Firefox and Chrome browsers. Visitors to http://www.getonsip.com will see links to developer tools for any business or service provider who’d like to incorporate web-based calling in their applications.
Having helped over 20,000 customers, OnSIP is a leading provider of business VoIP services to businesses. OnSIP provides commercial-grade real-time communications over a geographically redundant and scalable (pat-pending) platform. OnSIP offers enterprise pricing and tools to leverage their SIP platform for large-scale deployments, including APIs and smart proxy, registrar and location services. For more information, visit http://www.onsip.com/developer.