A call can be routed via a SIP provider to another SIP client or to a number on the conventional telephone network.

The parameters required to log on to a SIP provider are configured for a trunk group. During logon, the SwyxLinkManager registers the SIP trunk with the SIP provider in order to have telephone communication with a SIP client directly via an IP route.

This registration is repeated regularly as long as the SIP provider does not have any other default settings.

When calling a SIP client (this can be a terminal device or a software-based telephony system), the call will be routed to the desired client via the SIP provider. As part of this process, the SwyxLinkManager is responsible for the transmission of the logon information and the voice data as well as for the creation and termination of the connection. The voice data are routed directly to the SIP client without being routed via the SIP provider.

If this SIP link calls a number in the public telephone network (PSTN), the call will first be routed to the SIP provider via the IP route and goes from there to the appropriate phone number via the public network.

The knowledge base contains a list of possible SIP providers, which are compatible with SwyxWare. This list includes additional SIP provider profiles that are not included in the basic version of SwyxWare. You can read about this in the knowledge base:

STUN is a network protocol that recognizes the existence and type of firewalls and NAT routers and takes this information into consideration. It enables the uncomplicated use of devices (e.g. SIP telephones) and programs in networks that should receive information from the Internet.

STUN helps to identify the current public IP address of the line. This is necessary in order for the opposite terminal to correctly address and return your call data.

See RFC Standard (RFC 3489).

STUN messages are sent by the SwyxLinkManager at least every 10 seconds as long as there is no other data traffic circulating via the corresponding port. This ensures that the NAT router's NAT table (masquerading) cannot be destroyed again and that changes to the external IP address of the NAT router can be transmitted. This means that SwyxWare SIP trunks can also be operated with a DSL connection that is terminated every 24 hours by the IP provider and thus receives a new IP address.

Therefore, STUN messages, SIP logon, the SIP connection creation, and the voice data are sent via the NAT router.

If a firewall exists, it must be disabled for this type of communication. The rules required are described below and must be configured in the appropriate syntax in your firewall.