WebRTC to SIP calling: How to Call A Desk Phone From A WebRTC-enabled Browser
One of the most revolutionary features of WebRTC is its ability to merge different mediums of communication. Voice over Internet Protocol (VoIP), which is essentially making phone calls through the internet, has become a mature business sector in its own right. But generally speaking, the hardware that powers VoIP calls are functionally identical to standard telephone interfaces (usually 12 buttons attached to a desk phone, though software phones are somewhat popular).
Employees at call centers, office spaces, and many other venues still engage in the same telephonic activity with VoIP (pick up phone, hang up phone), but WebRTC is primed to merge these standard practices with the average user’s internet browser. WebRTC to SIP calling is the first step in the process.
The OnSIP App: WebRTC to SIP calling
OnSIP has a mature SIP platform that allows developers to harness the real-time communications capabilities of WebRTC. OnSIP's WebRTC platform is built atop its mature VoIP infrastructure and offers a scalable, geographically distributed SIP architecture that we’ve harnessed to build our own WebRTC to SIP calling applications. WebRTC to SIP calling happens when a user calls a deskphone through Chrome or Firefox. For instance, WebRTC to SIP calling could be used in a scenario where a user wants to call a tech support line by using Chrome itself, as opposed to actually picking up the phone and dialing the number.
The OnSIP app is located at app.onsip.com. OnSIP users can log in with their SIP addresses and web passwords. The OnSIP app works with Mac, Windows, Chrome, and Firefox. However, WebRTC is only used with the Chrome and Firefox versions, since it's a web-based technology.
On the first inbound or outbound call, the user will be asked to allow Chrome to share his/her camera and/or microphone with the OnSIP app. Give OnSIP a ring! Dial 1-800-801-3381 on the OnSIP app for your first WebRTC to SIP calling experience.
WebRTC to SIP Calling - How Does It Work?
WebRTC to SIP calling is an eminent possibility for any developer who utilizes the WebRTC APIs. When implemented on a mature SIP platform like OnSIP's, WebRTC applications can essentially operate as phones within the browser. Our engineers have completely integrated WebRTC into our platform, which enables browsers to make actual phone calls to support personnel and sales representatives who manage their operations with desk phones. In fact, we’ve managed to convert the entire process of calling a business into a single mouse click with our WebRTC-based sayso.
OnSIP's WebRTC platform is built for developers who want to blur the boundaries of traditional telephony. WebRTC to SIP calling merges the internet’s primary interface (i.e. browsers) with the IP traffic of VoIP platforms. The prospect of WebRTC to SIP calling also leads directly to WebRTC to Public Switch Telephone Network (PSTN) calls. OnSIP operates SIP-to-PSTN gateways that use SIP as an intermediary for browsers to call Plain Old Telephone Service (POTS) lines. This means that developers can use WebRTC and OnSIP to build applications capable of calling any phone directly from an internet browser.