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Corporate Firewall Configuration Requirements

To ensure your network can communicate with Askable correctly the included information should be used

Updated yesterday

Audience: Client IT and networking teams.


The below is the info you will need to pass on to your IT or networking teams if your organisation tightly controls internet traffic through firewalls and the like.

Askable uses WebSocket and WebRTC connections through our provider LiveKit to transmit data and media content. All transmissions are encrypted using TLS and DTLS protocols.


To ensure proper connectivity, please configure your corporate firewall to allow
outbound traffic to the following addresses and ports:

Required Firewall Rules:

  • Host: *.livekit.cloud | Port: TCP 443 | Purpose: Secure WebSocket signaling
    connection

  • Host: *.turn.livekit.cloud | Port: TCP 443 | Purpose: TURN over TLS (fallback
    when UDP unavailable)

  • Host: *.host.livekit.cloud | Port: UDP 3478 | Purpose: TURN/UDP servers for
    connectivity establishment

  • Host: All hosts (optional) | Port: UDP 50000-60000 | Purpose: Direct UDP
    connections for WebRTC media

  • Host: All hosts (optional) | Port: TCP 7881 | Purpose: TCP connections for
    WebRTC media

Optimization Recommendations:

For optimal audio and video quality, we strongly recommend allowing access to the
UDP port ranges listed above. Additionally, please ensure UDP hole-punching is enabled (or disable symmetric NAT if applicable).

This configuration allows devices behind your firewall to establish direct connections to our media servers, resulting in improved performance and reduced latency.

Advanced Troubleshooting

Steps

  1. Navigate to Sessions https://sessions.askable.com
    ​

  2. Click Sign In on top right corner and sign in with an askable account

  3. When redirected back to sessions right click on the page and select inspect

  4. When the dev tools open select the network tab

  5. Inside the network tab select the WS filter
    ​

  6. Click on new call to start a new session
    ​

  7. Click on Join Call to start call
    ​

  8. Find the item starting with rtc in the network view and click on it
    ​

  9. In the request url section copy the url up to /rtc e.g. wss://sessions-6d4cid41.livekit.cloud in the image below
    ​

  10. Select the payload tab and copy the item marked access token
    ​

  11. Navigate to https://livekit.io/connection-test and enter the URL and access token and click start test
    ​

  12. A successful test should appear as the image below
    ​

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