Is your network ready for VoIP? Let’s make sure.
On this page, you'll find all the essential checks and recommendations to prepare your network for high-quality VoIP calling. From bandwidth requirements and Wi-Fi optimization to avoiding double NAT and fixing latency or audio issues—everything you need is here.
If you’ve gone through these steps, your network is well on its way to delivering stable, reliable VoIP calls. With practical tips, clear explanations, and proven solutions, we help you get it right—before the first call even starts.
1. Bandwidth requirements for reliable calls
To ensure high-quality and uninterrupted calls, a stable internet connection with the following characteristics is recommended:
Latency
- < 50 ms – Low ping: ideal performance
- 50–100 ms – Average ping: acceptable, though minor delays may occur
- > 100 ms – High ping: risk of audio quality issues and call disruptions
What causes VoIP latency?
- Overloaded network: Too many devices (computers, smartphones, etc.) on the same network can lead to delays.
- Wi-Fi usage: Making calls via Wi-Fi, especially while on the move, can cause delays.
- Outdated hardware: Damaged cables, old modems, or devices that are not compatible with updates cause problems.
- Incorrect codecs: The codec used by your VoIP provider can affect latency, but this is beyond your control.
How to reduce VoIP latency?
- Check and replace equipment: Make sure modems, cables, and devices are up-to-date and in good condition.
- When experiencing problems, it's advisable to restart all network equipment, leave it unplugged for 1 minute, and then plug it back in. This clears the cache memory, and the equipment starts with a clean slate.
- Disconnect unused devices: Remove inactive devices from the network to free up more bandwidth.
- Use a secondary internet connection: A backup network can help during network outages or high latency.
- Prioritize network traffic (QoS): Give VoIP traffic priority with a VoIP router and optimize the use of data-intensive applications.
- Optimize data routes: Have your network tested and adjust routing for faster connections.
- Choose a good internet provider: A fast and stable internet connection is essential for reliable VoIP services.
By following these steps, you can minimize latency and improve the quality of your VoIP calls.
Bandwidth
- Download speed: At least 1 Mbit/s
- Upload speed: At least 1 Mbit/s
These minimum speeds are sufficient for up to 16 simultaneous calls, provided that 1 Mbit/s is available in each direction (upload and download).
Why These Requirements?
Voice calls using the G.711 (Alaw) codec require 64 kbit/s per audio stream. Since each call involves two streams (incoming and outgoing), this totals 128 kbit/s per call.
In practice:
- 1 Mbit/s total bandwidth (512 kbit/s up + 512 kbit/s down) supports 8 simultaneous calls.
- 1 Mbit/s exclusively available for upload or download can handle 16 simultaneous calls in one direction.
Tips for Reliable Call Quality
- Run a speed test to verify your actual upload and download speeds.
- Pay close attention to upload speed, as it's often more limited.
- If your current connection can't meet these requirements, consider a dedicated internet line for your VoIP phones.
- Limit bandwidth for cloud-based services. Limit the bandwidth of cloud-based services such as Dropbox and WeTransfer. These can cause your internet connection to become congested, resulting in lower call quality.
- Enable QaS.
In shared office buildings, internet is often shared. Ask the network administrator about available bandwidth, VoIP support, and whether there's a single NAT router—multiple routers can cause issues.
2. Connect devices directly
For the most stable and reliable VoIP performance, always connect phones directly to the modem/router provided by your internet service provider, or use a network switch that is connected to this main router.
Avoid placing an additional router between your phones and the main router. This can create a separate internal network, resulting in Double NAT, which often causes problems like one-way audio, failed calls, or poor connection quality.
3. Understanding NAT and its impact on VoIP
NAT (Network Address Translation) is a method used to manage and route traffic between external (WAN) and internal (LAN) networks. It allows multiple devices on a local network to share a single public IP address, which is crucial due to the limited availability of IPv4 addresses. Sometimes this can also cause troubles within your network. Learn more about it 
4. Allow Voys through your firewall
Strictly configured firewalls can cause audio and connection problems. In this case, ensure that the IP range of Voys is allowed for the VoIP phones. Simply opening ports is often not sufficient. Read more about it in 
5. Quality of Service (QoS)
Quality of Service (QoS) instructs your network to prioritize audio packets, sending them immediately without delay and ensuring that no packets are skipped. Voice and video traffic is then given priority over other types of traffic.
With heavy use of your network, such as heavy downloads or uploads, ping can increase significantly. This is because large data streams consume the available bandwidth, causing small VoIP data packets to be “queued.” Without prioritization via QoS (Quality of Service), for example, the network treats all data equally, which can lead to congestion.
Enabling QoS instructs your network to prioritize voice and video traffic, send packets immediately without delay, and prevent packet loss. QoS is set up on your router.
There is a big difference how to set up QoS on a router, so we can’t give you the exact steps. We can give you the IP Range that we use, you can find them in this page:
Tips
Never test audio problems with each other in the same room. This is because when you can hear each other 'live' as well, it seems like there's a huge delay on the line. When you can't hear each other live, you won't experience this delay, which is always present with telephony, as bothersome at all.
HD Audio codecs deliver beautiful sound, but are sensitive to network issues. Additionally, a crystal-clear sound on a phone is often perceived as strange and doesn't feel like a telephone conversation. On fixed devices, there isn't much value in enabling this.
Professionally installed WiFi-bridges or radio links set up by reputable companies work well in principle (for example, a WiFi bridge to your warehouse that's thirty meters from your office building). But WiFi-bridges that are self-purchased and installed for indoor use are not recommended for the same reason as connecting your device via WiFi. Try to always connect your telephony via cable in these situations.
- Avoid using WiFi bridges and Powerline adapters. Avoid using wireless WiFi bridges and Powerline adapters to extend your network. These can cause reduced voice signal quality. Always use CAT 5/6 network cables.
Some fixed devices can be connected via WiFi. However, WiFi is not a suitable medium for VoIP calls. Ping times are higher, the signal is more unstable, and there are more factors that can cause audio issues. Connect your fixed device always via cable.
In some rural areas or new business parks, the fixed internet connection may be so poor that you can't have a decent VoIP conversation over it. However, if you have good 4G coverage, you can purchase a 4G router to connect your devices. We have good experiences with this. VoIP doesn't use much data, so you can make and receive calls perfectly well with a small data plan.
Powerline adapters offer a simple way to distribute internet via the power network, but they have important limitations. They increase latency, limit available bandwidth, and can be a weak link in your network, especially for live applications like VoIP telephony, where a stable connection is crucial. While they may be useful in specific situations, they are generally less suitable for those who need a reliable and future-proof network. Using Cat5e or Cat6 UTP cables is the best choice for optimal performance and stability.