Web configurator

Proxy server port

Enter the number of the communication port that the SIP proxy uses to send and receive signalling data (SIP port).

Port 5060 is used by most VoIP provid- ers.

Registrar server

Enter the (fully-qualified) DNS name or the IP address of the registrar server.

The registrar is needed when the phone is registered. It assigns the public IP address/port number to your SIP address (Username@Domain) that were used by the phone at registration. With most VoIP providers, the registrar server is identical to the SIP server. Example: reg.myprovider.com.

Registrar server port

Enter the communication port used in the registrar. It is mainly port 5060 that is used.

Registration refresh time

Enter the time intervals at which the phone should repeat the registration with the VoIP server (SIP proxy) (a request will be sent to establish a ses- sion). The repeat is required so that the entry of the phone in the tables of the SIP proxy is retained and the phone can therefore be reached. The repeat will be carried out for all activated VoIP phone numbers.

The default is 180 seconds.

If you enter 0 seconds, the registration will not be repeated periodically.

Area: Network

Please note:

If you have downloaded the general settings for your VoIP provider from the Siemens configuration server (page 82), then some fields in this area will be preset with the data from this download (e.g. the settings for the STUN server and the outbound proxy).

If your phone is connected to a router with NAT (Network Address Translation) and/or a firewall, you must make some settings in this area so that your phone can be reached from the Internet (i.e. can be addressed).

Through NAT, the IP addresses of subscribers in the LAN are concealed behind the public IP address of the router.

For incoming calls

If port forwarding is activated or a DMZ is set up for the phone on the router, no special settings are required for incoming calls.

If this is not the case, an entry in the NAT routing table (in the router) is necessary in order for the phone to be reached. This entry is created when the phone is registered with the SIP service. In the interest of security, this entry is automatically deleted at certain intervals (session time- out). The phone must therefore confirm its registration at certain intervals (see NAT refresh time, page 81), so that the entry stays in the routing table.

For outgoing calls

The phone needs its public address in order to receive caller voice data.

There are two possibilities:

uThe phone requests the public address from a STUN server on the Internet (Simple Transversal of UDP over NAT). STUN can only be used with asymmet- ric NATs and non-blocking firewalls.

uThe phone does not direct the connec- tion request to the SIP proxy but to an outbound proxy on the Internet that supplies the data packets along with the public address.

The STUN server and outbound proxy are used alternately to work around the NAT/ firewall in the router.

STUN enabled

Click on Yes if you want your phone to use STUN as soon as it is used on a router with asymmetric NAT.

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