Click to Conference

This is a convenient variation of the click-to-call use-case: you can create a button that instantly connects to a personal audio conferencing room. Making a phone call takes a single click in the visited webpage. The network architecture is entirely identical, all it takes is a single WebRTC Server. That’s thanks to the fact that this server includes built-in conferencing. 

It is also important to realize that use of Web encryption standards in WebRTC conferencing makes it the most secure publicly available telephony system in telephony’s history. Signaling is using the same TLS standard used for communication between browsers and secured banking, retail and other sites. Voice and video are using similarly strong cryptography to make the conversations resilient against wiretappers. 

In this use-case, the cloud formation process creates a WebRTC Gateway and a web-page with a single click-to-confence button. If you click the button, you will be placed in an audio-conference. The conference room can also be reached through a SIP URI by a SIP phone or PSTN via Twilio Elastic Trunking.

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 conferencing link. Click on the link and accept self-signed certificates, a simple webpage with the button for joining the conference will appear.

Thanks to the WebRTC Gateway, the same conferencing room is reachable for SIP callers. They just need to use the SIP address that also appears in the Cloud Formation Outputs and has a  form “sip:*DIGITS@server-address”, such as



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 processes conferencing calls locally.

What Else You Should Know

You can also have your audio conferencing room reachable from PSTN. You can use any PSTN/SIP service that offers PSTN origination, such as twilio Elastic SIP Trunking. With twilio, you need to set up your account, an Elastic SIP Trunk, and a PSTN number — see Quick Start Guide and  Getting Started. Eventually you need to link twilio’s PSTN service to your conferencing service. Visit the “Elastic SIP Trunking” section in the twilio management GUI, and add the SIP URI shown in Cloud Formation template ouputs as Origination URI. Then, calls to the twilio PSTN phone number will be routed directly to your conferencing room.