SIP Information & Settings

Below you can see basic or default settings that must be entered in each SIP device, in addition to the user information (user name and password). You can also find information about this in the customer center under the respective help. A little further down you will also find general information about TCP and encryption.

Please note that we already officially support IPv6 since 2014. This means that you no longer need any NAT or tunnel services if you have a native IPv6 connection (e.g. Deutsche Glasfaser customers) and use the corresponding FQDN’s of the servers.

The methods for determining the IP addresses of our SIP servers are DNS A Record, DNS NAPTR Record and DNS SRV Record.

What type of IP address do I have?

General setup info

Firewall info

SIP over TCP and UDP

The Session Initiation Protocol(SIP) is a network protocol for establishing, controlling and terminating a communication session between two or more participants (source: Wikipedia). The signaling of SIP packets of a telephone call is usually handled via UDP (User Datagram Protocol). Data is divided into packets with a theoretical maximum size of 1500 bytes. However, this size is further reduced by additional and further logical information to a usable size of about 1300 bytes. If such a package is sent, it contains both the sender’s and the recipient’s address. Unfortunately, there is no feedback to the sender whether the package has actually reached the recipient. As a result, when using UDP, additional care and control must be taken to ensure that a packet has actually reached the recipient. This happens in the SIP protocol by means of acknowledgement responses. An “INVITE” (call from A to B) is followed in this case by a “TRYING”, then a “PROGRESS” and so on. If an acknowledgement is missing, the last package is sent again at intervals according to defined timers until an acknowledgement is received. If this continues to fail, the session runs into a predefined timeout and is aborted.

With SIP over TCP (Transmission Control Protocol), the transport behaves fundamentally differently. Comparable to a “registered mail with return receipt”. TCP is more “robust” than UDP in many respects due to its concept, but unlike UDP it also has a certain amount of additional data (overhead) and requires more bandwidth than UDP for the same amount of user data. However, this is largely negligible in the case of SIP. The advantages of TCP is the architecture of the protocol itself. It contains built-in mechanisms to detect and resend “lost packets” autonomously. Likewise, a TCP connection is established in dedicated form. This means that before data is transmitted, the recipient already knows that it is coming. There are more advantages of TCP which are also easier to handle for NAT (Network Address Translation) and accordingly for firewalls than compared to UDP. Which is the “better” protocol does not really matter. It must meet the requirements for you as a customer. dus.net GmbH offers you the possibility to use either UDP or TCP as transmission protocol for SIP. The settings for this must be made exclusively in your respective terminal device. No setting is required on the part of dus.net.

By the way…for the audio data, i.e. the actual phone call, UDP is generally used. The reason for this is, among other things, the significantly lower “overhead”, i.e. the large proportion of user data and the less time-intensive checks. A possible resending of packets makes rather less sense for a real-time application.