Siemens Gigaset S675IP User Manual page 100

Proxy server port
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.
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 pub-
lic 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 con-
figuration server (page 101), then some fields
in this area will be preset with the data from
the download (e.g. the settings for the STUN
server and 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 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
NAT refresh time, page 100), 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.
STUN enabled
Click if you want your phone to use
Yes
STUN as soon as it is used on a router
with asymmetric NAT.
Web configurator
99