Siemens S450 IP 사용자 설명서

Gigaset S450 IP LBA / SGP / A31008-M1713-Y221-1-7619 / web_server.fm / 30.8.07

Ve

rs

ion 4,

 16

.09.

2005

Enter the number of the communica-

tion port that the SIP proxy uses to send 

and receive signalling data (SIP port). 
Port 5060 is used by most VoIP provid-

ers.

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 pub-

lic IP address/port number to your SIP 

address (

) 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.

Enter the communication port used in 

the registrar. It is mainly port 5060 that 

is used.

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. 

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 subscrib-

ers 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 spe-

cial 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 regis-

tered 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 

, 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: 
u The 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. 

u The 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. 

Click on 

 if you want your phone to 

use STUN as soon as it is used on a 

router with asymmetric NAT.

If you have downloaded the general settings 

for your VoIP provider from the Siemens con-

figuration 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).