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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
session). 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 default is 180 seconds.
If you enter 0 seconds, the registration
will not be repeated periodically.
Area:
Listen ports
Specify the phone's local ports for VoIP
telephony here. The ports must not be
used by any other subscriber in the LAN.
SIP port
Specify the local communication port
that the phone should use to send and
receive signalling data. Specify a
number between 1024 and 49152.
The default port number for SIP signal-
ling is 5060.
Please note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.
RTP port
Specify the local communication port
that the phone should use to send and
receive voice data. Enter an even
number between 1024 and 49152.
The port number must not be the same
as the port number in the
If you enter an odd number, the even
number just below it will be set (e.g. if
you enter 5003, 5002 is set). The
default port number for voice transmis-
sion is 5004.
Please note:
Ports 0 to 1023 should not be used,
because these are often used by standard
applications.
Use random ports
Area:
If your phone is connected to a router with
NAT (Network Address Translation) and/or
Firewall, you must make a few 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
field.
SIP port
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
Click on the option Yes, if you do not
want the phone to use fixed ports for
SIP port
and
RTP
port, but rather to use
any free ports.
The use of random ports makes sense if
you want several phones to be oper-
ated on the same router with NAT. The
phones must then use different ports
so that the router's NAT is only able to
forward incoming calls and voice data
to one (the intended) phone.
If you click on No, the phone will use
the ports specified in
port.
Network
Please note:
If you have downloaded the general settings
for your VoIP provider from the Siemens
configuration server (page 60), 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).
Web configurator
SIP port
and
RTP
57
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