Written by . Last updated: January 18, 2021

Network and VoIP

A great advantage of VoIP telephony is that a separate network doesn’t need to be built. You can connect a VoIP telephone (or a VoIP server) directly to your established internet connection. It doesn’t matter which internet provider you receive the connection from.

In the event that you are going to use a telephony system with five telephones or fewer, then there isn’t really anything you need to do to your network. Just make sure to pay attention to the following:

  • Having a telephone conversation takes about 0.1 Mb/s upload and download speed.
  • A VoIP telephone is connected directly to a modem/router. If you want to connect multiple devices, you need to check to see if there are enough free ports. If not, you can buy a switch.
  • When the telephone is connected to an internet connection that is shared by computers that use a large about of internet traffic, it can affect the conversation quality. You can choose to modify the router settings in order to reserve a portion of the internet speed for VoIP (QOS). Not all routers support this.

When you want to connect multiple phones, read the following directions that are the foundation to a successful VoIP implementation.

The foundation of a successful VoIP implementation

Before you set up a VoIP environment, its important to know that you can solve most problems by following the guidelines below. It usually turns out that almost all malfunctions happen because of incorrectly set up infrastructure and/or hardware. The guidelines below are meant as a general guide and should be followed regardless of which hardware you use.


  • Avoid NAT behind NAT at all times.
  • Always turn off SIP functions (for example SIP application Layer Gateway) in the router.
  • Don’t use STUN, TURN, UPnP or ICE.
  • Turn on keep-a-live (10 seconds is a good value).
  • Port Forwarding is not only unnecessary, but also discouraged.
  • It is preferable to use an extra PVC (xDSL) or VLAN (glass fiber) to not be bothered by other data traffic, preferably without overbooking (1 to 1).
  • Dimension the data line on the basis of 0.1 Mb per conversation (up and down) when using the G. 711 A-law (PCMA).
  • Make sure that the MTU for the data line in use is properly set in the modem/router.
  • Use a separate network for voice and data (vlan).
  • Apply [[Quality of Service (QoS)]|QOS] in switches and routers to guarantee bandwidth and prioritize voice traffic.
  • Always use a host name for registration. IP addresses can change.
  • The proxy domain supports SRV records.
  • SIP/UDP has priority over SIP/TCP.
  • Its preferable to use G.711 A-law (PCMA). This has the best sound quality. If bandwidth is a problem, you can also consider using G729a.
  • DTMF support is based on RFC2833.

Hosted telephony

  • It is preferable to us the G.711 A-law codec with a packet size of 20 milliseconds.
  • Use a unique local listen port for each SIP device (5061, 5062, etc.).
  • A good NAT router is important for hosted VoIP telephony.

Personal IP PBX

  • Ideally, don’t place the IP PBX behind NAT. If you do want to put the PBX behind NAT, use 1:1 NAT with an unused public IP address.
  • Prevent dynamic NAT. This often causes problems.
  • Provide the IP PBX with a firewall that only allows traffic to and from a limited IP range (see below).
  • Have the IP PBX authenticate / match by account_id and not host name. Incoming traffic can come from multiple hosts and would be blocked.
  • A static trunk works more reliably than a dynamic trunk with registration.
  • DID/DDI is set up for the whole number (including country code with a + before). This can be changed in the admin environment.
  • Caller ID to the outside also has the same format. (+31501234567)

IP Range

The IP range of our platform is:  and

The next ones in the series are: / / /

The IPv6 range, which is only used for webhooks: 2a06:2a80::/29


We do different things to avoid misuse of your trunk and VoIP accounts.

  • Firstly, we built a misuse detection system. When misuse is detected, a one-time charge of € 44.00 for the detection, analytics, and blocking by an engineer, will be added to your bill on top of the conversation costs.
  • Secondly, we blocked the most used end destinations that are used for misuse.
  • The third defense is in our web interface. There you have: real time insight in your conversation details, the ability to make IP restrictions, the option to block specific types of destinations, and the ability to change your VoIP account or VoIP trunk passwords in real time.

We don’t stop here. We also always busy fighting fraud. This is how we keep up with operator’s blacklists of end destinations that are used for misuse.

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