# Network requirements

> Set up your firewall and router so phones stay registered and calls have clear two-way audio.

- URL: https://kb.voipcloud.online/getting-started/network-requirements/
- Section: Getting started
- Type: guide
- Applies to: All countries
- Last reviewed: 2026-11-09

Your phones and apps need to reach the SIP and audio servers through your network. Most home and office routers work without changes. Use this page if you have a strict firewall, or if calls have one-way audio, drop out or devices go offline.

## Before you start

- You need admin access to your firewall or router.
- Find the server your devices use: go to **PBX** › **Devices**, open a SIP device and copy the **Domain** under **SIP credentials**. It shows the server name and port. **Transport protocol** shows **TCP**, or **TLS** when TLS encryption is turned on in **PBX** › **Settings** › **General**.

## 1. Allow SIP signalling

SIP signalling registers your devices and sets up each call. If your firewall is stateless, allow this traffic both inbound and outbound.

The server addresses and ports aren't published here. [Raise a support ticket](https://kb.voipcloud.online/support/support-tickets/) and ask the support team for the SIP addresses and ports to allow for your account.

Always set up devices with the domain name from **SIP credentials**, never with an IP address. The servers are load balanced, so a device won't always connect to the same IP address.

If you use a SIP trunk that connects by IP address (SIP peering), the support team gives you the address and port to allow when they set up the trunk.

## 2. Allow call audio (RTP)

Call audio travels separately from signalling, over RTP. Allow this traffic **both inbound and outbound**. If only one direction is allowed, you'll get one-way audio.

Ask the support team for the audio (RTP) addresses and port range to allow, at the same time as the SIP addresses.

These addresses can change. If you allow traffic by IP address, check with the support team from time to time.

## 3. Turn off SIP ALG

Many routers have a feature called SIP ALG that rewrites SIP traffic. It often causes registration problems, one-way audio and dropped calls. Turn it off in your router's settings. Where it's found depends on the router, so check its manual if you can't see it.

## 4. Prioritise voice traffic

If your internet connection is shared with downloads, backups or video, give voice traffic priority with your router's QoS (quality of service) settings. For example, prioritise your phones by device, or prioritise the RTP port range the support team gives you.


## Check it works

1. Go to **PBX** › **Devices** and check the device shows **Online**.
2. Open the device. Under **Connection information** you'll see its **Status**, public **IP address**, **User agent** and **Transport protocol used**.
3. Click **Show more** to open **Advanced connection information**. **Local IP** shows the device's address on your local network. It only shows correctly when SIP ALG is turned off on the router.
4. Make a test call in each direction and check that both people can hear each other.

**Advanced connection information** also lists **Device type**, **SBC IP**, **AOR** and **Expiry time** (in seconds). The support team may ask for these values when helping you troubleshoot.

## Next steps

- [Device offline](https://kb.voipcloud.online/troubleshooting/device-offline/)
- [One-way or no audio](https://kb.voipcloud.online/troubleshooting/one-way-or-no-audio/)
- [Poor call quality](https://kb.voipcloud.online/troubleshooting/poor-call-quality/)
- [Calls drop after a few seconds](https://kb.voipcloud.online/troubleshooting/calls-drop-after-a-few-seconds/)
- [Devices](https://kb.voipcloud.online/pbx/settings/devices/)