Click to Dial

Click-to-dial also known as click-to-call or click-to-talk, is a web-based application to initiate encrypted phone calls by a single click from a browser. This capability is often used in online audio conferencing, call center applications, and customer web hotlines. The technology is based on W3C’s WebRTC Framework implemented nowaydas by major browsers. The browser phone-calls backed by a WebRTC gateway can reach almost any type of user: other browser users, SIP telephone users, SIP trunks and PSTN phones behind them. 

The use-case shown here is as simple as a custom-made dial button which, when pressed, initiates a phone call to a SIP address. During the cloud-formation process a WebRTC gateway will be started as well as a web-page for creating a custom click-to-dial button. You can alter the called SIP URI and graphical appearance of the button. When you click the button, a call will be initiated from your browser to the specified SIP URI.

How to Use It

To start the cloud formation process visit the following link:

Once the cloud formation process completes which takes several minutes, the Outputs will show the Click-to-Dial link. Click on the link and accept self-signed certificates. In the left part of the screen you can customize the dialling button which appears on the right-hand side. The most important part is the “Call To” address — it must include a valid destination SIP URI. You may for example use sip:music@frafos.net. After you press the dialling button, you will be prompted for permission to use your audio/video equipment. After you approve, your browser will  connect to the SIP destination through the WebRTC browser. You can also copy-and-paste the HTML code bellow the button and place it in your webpage. Just keep in mind that the phone calls will work as long as your WebRTC Gateway instance is running.

What is Orchestrated

 

This template starts an instance of WebRTC Gateway in your default VPC. There is a simple pre-installed web application based on JSSIP that implements the browser telephony.  The gateway translate “browser calls” to SIP (RFC3261) calls and vice versa. 

What Else You Should Know

The web telephony is using the same security protocols as the web applications do: TLS, accompanied by DTLS+SRTP for media. That means that every browser user makes a phone call which is encrypted the same way online-banking and other critical applications are. Such a massive availability of high-quality encryption provides unique resillience against wiretapping for the first time in telephony’s history!

A great benefit of AWS is its support for multiple geographic regions. This way multiple WebRTC gateways can be placed close to their users while dramatically reducing packet latency and improving the speech quality.