WebRTC (Beta)

WebRTC is a software phone that is included directly in the Daktela application, eliminating the need for a third-party SW phone.

WebRTC (beta) is available from version 6.18. It works in most browsers except Safari.

The limit of the online WebRTC devices for your domain is 50 devices.

Enable WebRTC (Beta)

Please contact us on +420 226 211 245 or at daktela@daktela.com to enable WebRTC.

Set up Your SIP Device

Go to Manage → Devices → SIP Devices. Open the device you want to use with WebRTC or create a new one.

Set up your SIP device and under Transport, select wss.

Save your SIP device.

If you change an existing SIP device into a WebRTC device while a user is logged in to the original SIP device, they may need to log out and back in for the change to take effect.

Connect Your WebRTC

Connect and Disconnect Your WebRTC

To connect or disconnect your WebRTC, click the Queue button in the static panel at the top of your screen. The Log in and out of queues and devices window will open.

When you have a WebRTC device, two checkboxes will be displayed under your list of devices:

Connected – check to connect, uncheck to disconnect
Autoconnect after reload – check to automatically connect after refreshing the page

When your WebRTC is connected, a yellow phone icon will be displayed next to your extension number in the static panel. When your WebRTC is disconnected, the icon will turn red.

Using WebRTC

When a call is ringing at your WebRTC, an icon will flash green and blue next to the Phone icon.

You can accept the call in the standard notification or by clicking the flashing icon.

When you make or receive your first call, you will need to allow your browser to access the microphone.

During your calls, you will see a connection quality metre in your Phone widget. (This feature is not available in Firefox).

Refreshing your page during a call can lead to unexpected call drops.