Network: the key to a stable VoIP connection

1. Bandwidth requirements for reliable calls

To guarantee optimal call quality, a stable connection with the following specifications is required:

• Latency:
• < 50ms = Low ping = ideal
• 50 ms - 100 ms = Average ping = acceptable - delay possible
• > 100 ms = High ping - possible audio quality / call continuity issues
• Download speed: Minimum 1 Mbit for up to 16 simultaneous calls
• Upload speed: Minimum 1 Mbit for up to 16 simultaneous calls

Why these specifications? The Alaw Codec (G.711), commonly used for high-quality voice calls, has a bitrate of 64 kbit/s per audio stream (one direction). This means each call requires 128 kbit/s for both upload and download combined.In practice:

• 1 Mbit of bandwidth provides capacity for 8 simultaneous calls when upload and download each use 512 kbit/s.
• With a full 1 Mbit used only for upload or download, 16 simultaneous calls can be handled.

With these guidelines, you'll ensure smooth and uninterrupted communication.

2. NAT

NAT stands for Network Address Translation. What NAT does is make a translation between external (WAN) and internal (LAN) traffic. This is necessary so that, for example, our server doesn't just reach the client's router (WAN), but also the device located in the network behind the router (LAN). The separation between internal and external networks is necessary because there are more network devices than there are IP(v4) addresses. By separating an internal and external network from each other, you bypass this problem. With the new IPv6 protocol, NAT will no longer be necessary in the future because there will be more than enough IP addresses available to give each network device its own unique IP address.

So NAT is good and, as long as IPv4 is still in use, even necessary. But what exactly is Double NAT? And why is it bad for 'peer-to-peer' services like telephony and also online gaming?

2.1 Double-NAT

Double-NAT (Network Address Translation). A router translates between an internal network (LAN, your office network for example) and an external network (WAN, your internet provider's network). You then connect your network equipment (computers, printers, phones) to this router. VoIP telephony basically assumes this situation. However, sometimes VoIP phones are connected to a router that is in turn connected to another router.

An example is a WiFi router that you connect to your internet provider's router. Or your own router that is connected to a poorly configured router in a business collective building. This causes a translation from internal to external network to occur twice.

Another example is an Englishman speaking with a Dutchman who understands English. The Dutchman happens to know German and translates this conversation to a German. The Englishman's message does reach the German, but they are unable to talk directly to each other, which is what should be possible with peer-to-peer services. In practice, this often leads to one-way audio or even no audio at all with VoIP telephony.

The result of this can be that you experience one-way audio, no audio at all, or calls that fail to connect. This can be solved in several ways:

• Connect the device to the first router in your building.
• Put the first router in bridge mode. This allows you to keep the physical network setup as it is, but NAT no longer takes place on the first router, which structurally solves the problem. Usually, your internet provider can do this remotely for you.
• Contact your network administrator or the building's network administrator with a request to eliminate the Double-NAT situation for VoIP telephony.

2.2 Testing double NAT yourself (for the technical people)

You can find out through your computer if there is a double-NAT situation. It's important that this computer is in the same internal network as your fixed device.

Windows

• Go to Start on your laptop.
• Type in the search bar CMD (command prompt) and press enter.
• Enter the command: tracert ha.voys.nl [enter] This is what a connection looks like with 1 router that translates between an internal and external network. The internal IP address is censored with white:

The image above shows a single NAT, which is good. You can tell because there is only 1 internal IP address (step 1 of the trace). Internal IP addresses always start with 10.x.x.x, 192.168.x.x, or 172.16-24.x.x. The example below with the MacBook shows a double-NAT situation.

MacOS

• Open the Terminal on your macbook (find it using the search bar in the top right).
• In MacOS, enter the command: traceroute ha.voys.nl [enter]

The image above shows a single NAT, which is good. You can tell because there is only 1 internal IP address (step 1 of the trace). Internal IP addresses always start with 10.x.x.x, 192.168.x.x, or 172.16-24.x.x. The example below with the MacBook shows a double-NAT situation.

You will now see all the 'hops' that a small packet makes from your computer to our server (ha.voys.nl / ha.voys.be). In a network setup suitable for VoIP telephony, you will encounter only 1 hop with an internal IP address. You can recognize an internal IP address if it starts with the following number sequences: - 192.168.x.x - 172.16.x.x through 172.31.x.x - 10.x.x.x If you encounter multiple consecutive hops with an internal IP address, there is a double-NAT situation. However, there is an exception. Some providers in the Netherlands also use the 10.x.x.x range as an external (WAN) network. If you encounter a 10.x.x.x network as the second hop, this usually does not indicate double-NAT. Any time-outs in a traceroute do not indicate network issues and can be ignored.

3. High ping times - latency

Stuttering audio, or audio that 'suddenly' drops out? Good chance that the ping times from your device to the outside world are high. A ping time is the time it takes for certain network devices to respond to each other and is (hopefully) displayed in milliseconds. Network device A (your computer) sends a small packet to network device B (for example, our server). The time it takes for A to get a response back from B is the ping time. If these ping times are constant and consistently <150ms? Then you can have a good quality conversation without audio stuttering. Are the ping times variable and sometimes above 150ms? Then you may experience audio issues. This can be caused by quite a few factors (from corrosion to cables that are too long). Initially, it's important to determine where these high ping times occur:

3.1 Testing high latency yourself (for the technical people)

Open 3 different command prompts on your computer. In macOS, you can open them with 'terminal'; in Windows, click on the Windows logo and type CMD [enter].

1. In the first command prompt in macOS (terminal), enter the command ping ha.voys.nl. In Windows, use the command ping ha.voys.nl -t. This sends small packets to our server. The time in ms indicates how long it takes before you receive an answer to this packet. Timeout? No answer is given within one and a half seconds = audio issue. This is what a stable connection looks like. All response times are consistently ~10ms.
2. In the second command-prompt, use the same commands, but instead of ha.voys.nl / ha.voys.be enter the IP address of your device.
3. In the third command-prompt, use the same commands, but instead of ha.voys.nl enter the IP address of your router.

This is what a stable connection looks like. All response times are consistently ~10ms.

If high ping times occur only in step 2 (ping to ha.voys.nl), there is a problem with your external network connection. Try resetting your router in this case. If the problem is not resolved, contact your internet provider or network administrator with these findings.

If the high ping times also occur in step 3 or 4, then the problems are occurring within your own network. This can have multiple causes (and solutions!):

• If the high ping times (or time-outs) only occur in step 3, then there is an issue with the connection between your device and your router. Try connecting your device in a different location, or replace the cabling. If this doesn't work, contact your network administrator.
• If the ping times occur on steps 3 and 4 (which should never happen on just 4), there may be an issue with a network switch, the router itself, or the cabling. Reset your network equipment. If this doesn't work, contact your network administrator.

If all ping times are good, but you still experience stuttering audio, 'packet loss' may be the cause of the audio issues.

3.2 Causes of VoIP latency and solutions

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?

1. Check and replace equipment: Make sure modems, cables, and devices are up-to-date and in good condition.
1. 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.
2. Disconnect unused devices: Remove inactive devices from the network to free up more bandwidth.
3. Use a secondary internet connection: A backup network can help during network outages or high latency.
4. Prioritize network traffic (QoS): Give VoIP traffic priority with a VoIP router and optimize the use of data-intensive applications.
5. Optimize data routes: Have your network tested and adjust routing for faster connections.
6. 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.

3. Packet loss

As the name suggests, small network packets can get lost on the internet. When you're browsing, the page might load a fraction of a second slower. Streaming services like Netflix buffer their videos, so you won't notice this either. But VoIP telephony is live and cannot buffer or deliver audio to the other end later. If a packet is lost, your conversation stutters. You may also experience this with online gaming.

We can investigate for you whether packet loss is occurring between our servers and your router. For this, please contact one of our technical colleagues. Unfortunately, we cannot solve these problems for you, but we can make a report that will help your internet provider resolve these issues for you.

4. Additional tips

Testing. 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. 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.

WiFi. 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. If you still want to use WiFi, be sure to read our article about the Ideal wireless internet connection. Full of useful tips and tricks. Because to measure is to know!

4G-routers. 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. 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.

WiFi-bridges and radio links. 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.