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FAQ

General questions about Internet telephony

What does Internet telephony or Voice over IP (VoiP) or cloud telephony mean?

The term Internet telephony and Voice over IP describe the same thing. This is about making phone calls via your Internet access using the SIP protocol specially developed for this purpose. This method of telephoning makes the old analog or ISDN technology completely obsolete and is also much more powerful.

Cloud telephony goes even further to describe the fact that a classic on-site telephone system is no longer needed, or that the “cloud” replaces it because a virtual telephone system is located there. This is especially interesting for entrepreneurs, as the classic PBX systems can cause high costs.

However, a telephone system in the cloud or cloud telephony does not mean that this service is always coupled with a dedicated system, such as a server at the cloud provider on which the telephone system then runs. At dus.net, the customer center is already the virtual telephone system and thus already a cloud-based TC ankage.

Can I completely do without my old landline with VoIP?

Yes, in principle. All you need for Internet telephony is Internet-only access. Unfortunately, it is still not possible in Germany to order just an Internet connection from all Internet providers. Unfortunately, as a customer you are almost always forced to book a telephone connection at the same time. Internet and telephone are therefore usually coupled with each other here.

Since these telephone connections are usually only very rudimentarily equipped, you can always use a dus.net telephone connection in parallel or additionally.

Do I need a specific Internet connection?

No, you can use any Internet connection with us, so a connection with dus.net is provider-independent. Only the bandwidth of the Internet connection is important. However, so little is needed here that it is possible with almost all Internet connections, even DSL-light connections. However, with low bandwidths, it is a great advantage to have enough of this available for telephony. End devices such as the FritzBox, for example, regulate this completely automatically.

What do I need for Internet telephony?

In addition to an Internet connection, you also need a SIP terminal that can handle the SIP protocol. The most well-known device is probably the Fritz!Box, although this is already much more than a pure SIP end device, such as an IP telephone. But also softphones that can be installed as software on the PC or as an app on the smartphone are SIP end devices and fully usable.

Are there any differences to the old analog or ISDN telephony?

A clear yes, and in a positive sense. Internet telephony not only offers far more functions and possibilities, as well as enormous flexibility. It offers the same ISDN voice quality and even HD telephony.

Questions about the SIP connection

What is a SIP connection and where can I find it?

A SIP connection is a virtual telephony or fax connection. Every SIP end device needs access data to be able to log in (register) with a SIP provider like dus.net on the SIP server.

SIP connections are available in conjunction with a tariff, and the number varies depending on the tariff. With dus.net it is necessary to use a separate SIP connection for each SIP end device. A SIP connection may not be used by multiple SIP end devices at the same time.

The SIP connections can be found in the Customer Center and additional SIP connections can be created there at any time until the quota corresponding to the rate is exhausted.

Why is it not possible to use a SIP connection more than once at dus.net?

With a SIP connection, your SIP terminal device logs on to our SIP server and remains registered there for a certain time. With the registration we receive an IP address and a port from the end device. Now we know how and where to find your terminal when we need to deliver an incoming call to it. If several terminals were to share a SIP connection, only the terminal that registered last would ring.

How long does a SIP connection remain registered?

The time for which a SIP connection or the SIP end device with this connection remains registered on our SIP server depends on the end device itself. This is because the terminal device tells our system when it will re-register the next time. With a FritzBox, for example, this is always every 30 minutes, i.e. the box registers a connection at 13:32, for example. Shortly before 14:02, the box reports to the server again and renews the register for another half hour.

My terminal device is offline. Why does the customer center say it's online?

As described in the answer to the question “How long does a SIP connection remain registered?”, your terminal device itself specifies the time until the next registration, i.e. with the first registration your terminal device tells our server when it will register again. In the time between these two points in time, your terminal device is considered “online” for our system, even if it may have been switched off in the meantime. Only when the device no longer reports at the second time, it is then considered “offline” for us.

The “online” “offline” display in the customer center is therefore not a real-time display.

Why can't my SIP connection register?

User data incorrect
This can be for a variety of reasons. Most of the time, it is something quite banal, such as incorrect access data. It often happens that passwords are simply mixed up or a number is forgotten in the user name. If you can exclude that everything has been entered correctly on your side, it is also possible that the register attempt of your terminal does not reach us at all due to network problems.

If everything else is ruled out, it is also possible in the end that your IP address has ended up on a blacklist with us. This usually happens only if your terminal device has tried to register too many times with incorrect user data. After a certain time, the lock is automatically removed again. If not or if it takes too long, you can write to us by eMail and have your IP address unlocked again.

no real IPv4
You may no longer have a real IPv4 address on your Internet connection. Some ISPs no longer assign real IPv4 addresses on their customers’ connections (scarcity), but only internal IPv4 addresses. The problem now, internal IP addresses are not reachable from outside. Therefore, on the provider side, these internal IPv4 addresses are converted to external IPv4 addresses. However, since savings have to be made on IP addresses, many customers now receive identical IPv4 addresses to the outside world, with the only difference being the ports.

Therefore you should try to use “proxy3.dus.net” (DS-lite) or “proxyv6.dus.net (pure IPv6 connection) as registrar instead of “proxy.dus.net”. In the FritzBox, however, you must also allow the connection via IPv6 in the access data, which is actually already specified by default.

Questions about the SIP call

Why are there always audio dropouts during the call?

Dropouts during the call indicate that RTP packets are lost. This usually happens regularly and, in principle, does not matter. However, if there are too many packets that do not find their destination, this becomes noticeable with dropouts.
The reasons for this are manifold and can be caused in your own network or on the side of your ISP or routing. The latter is the path of the packets through the Internet, whereby between your Internet provider and the dus.net there are other providers through which the traffic also runs. So there are many possible causes of errors that you can try to determine with the help of programs on the PC. These free utilities determine the individual “hops”, i.e. stations over which the packets run in the network and possibly show where these packets “hang”. The data can allow conclusions to be drawn, but cannot always be used as a cause.

Why can't I hear the other side or can't he hear me?

If after a call setup the other party or you yourself do not hear any audio, it sounds like a firewall problem. The structure of a conversation and the conversation itself take place on different levels. The SIP packets are responsible for the setup and teardown, the call itself then runs as an RTP stream via other IP addresses or servers. If the firewall blocks the other IPs, it also blocks the stream and thus the audio.

As a rule, the firewall does not need to be configured because mechanisms allow nothing to be blocked here. However, if you have deliberately “sharpened” your firewall, it is best to unblock the dus.net IP network as a whole.

Why does my conversation stop after barely 30 seconds?

FritzBox behind router
If, for example, a FritzBox is not operated directly on a DSL connection, but behind another router/modem or a provider that works with wireless technology, for example, it can happen that incoming calls are simply disconnected after approx. 30 seconds. This has to do with the firewall of the upstream router at your end or that of the provider. In such an environment, the FritzBox does not register with its default port 5060 because the box is in an internal network. The FritzBox must therefore be configured to behave like a client in the internal network. So you have to configure the Internet access on the FritzBox as a <<Connection to external modem or router>> set up (or with a cable provider as <<Connection to a cable modem>>).

no real IPv4
You may no longer have a real IPv4 address on your Internet connection. Some ISPs no longer assign real IPv4 addresses on their customers’ connections (scarcity), but only internal IPv4 addresses. The problem now, internal IP addresses are not reachable from outside. Therefore, on the provider side, these internal IPv4 addresses are converted to external IPv4 addresses. However, since savings have to be made on IP addresses, many customers now receive identical IPv4 addresses to the outside world, with the only difference being the ports.

Therefore you should try to use “proxy3.dus.net” (DS-lite) or “proxyv6.dus.net (pure IPv6 connection) as registrar instead of “proxy.dus.net”. In the FritzBox, however, you must also allow the connection via IPv6 in the access data, which is actually already specified by default.

Questions about the tariff

Where can I find the DUStel trunk tariff?

The DUStel trunk rate no longer exists in its old form. Instead, we offer you an excellent alternative with the DUStel business flex rate.

Where can I find the DUStel 60 tariff?

We have integrated the DUStel 60 tariff into the DUStel starter. In the starter you can now select whether you want to have the seconds or the minutes clocking.

Where can I find the DUStel complete 6 and DUStel complete 12 tariff?

Now only the DUStel complete is available from EUR 7.90/month with a minimum contract term of 12 months. It already includes a flat rate to the German fixed network for one line. An additional line can be booked, as well as flat rates for the German mobile network. The DUStel one is omitted completely.