To ensure the proper functioning of WebRTC in most browsers, a trusted SSL certificate is required. We recommend using the additional module Let's Encrypt Module. For the module to work, the PBX must be accessible via a public IP address.
Create a new employee account.
Option 1. For each internal number, set it to work only over the RTC protocol. In the "Advanced Settings" section - "Extra options", add the following options:
Click "Save". From this moment, this internal number will work only using the WebRTC protocol.
Option 2. Set up all internal numbers to work both via PJSIP and WebRTC protocols. To do this, go to System → General Settings → SIP and enable the "Use WebRTC" toggle switch.
In MikoPBX, go to Network & Firewall → Firewall, and add the subnet 0.0.0.0 with a mask of 0.0.0.0. Enable access via AJAM.
Go to System → General Settings → AMI & AJAM. Ensure the "AJAM Port with Encryption" is set to 8089.
In the General Settings section, specify the STUN server address. For example, stun.sipnet.ru.
Open the link in your browser: https://PBX_ADDRESS:8089/asterisk/ws. Use Chrome, as other browsers may have issues. If the certificate is self-signed, you might see a warning "Connection is not secure", ignore it and click "Proceed to the website."
You should see the following message:
If Asterisk responds, the setup was successful.
Go to the website https://sipml5.org. You will be redirected to https://www.doubango.org/sipml5/. Follow the link "Enjoy our live demo".
Set up the WebRTC client:
In the "Public Identity" field, enter the following template:
When you enable the Use WebRTC option
Click on the "Expert mode?" button and proceed with the additional configuration:
In the "WebSocket Server URL" field, enter the following format:
Click Login. You can now start making calls.
