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Configuring telephone connections
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 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 96), 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 NAT (Network Address Translation) and/or the firewall are activated on your base station
router, you may have to make some settings in this area so that your base station phone
can be reached (i.e. can be addressed) from the Internet.
Through NAT, the IP addresses of subscribers in the LAN are concealed behind the public IP
address of the router.
For incoming calls
If on the router port forwarding for SIP and RTP port (page 76) is activated to the phone,
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 is automatically created when the phone is registered
with the SIP service. In the interest of security, this entry is deleted at certain intervals
(session timeout). The phone must therefore confirm its registration at certain intervals
(see
time, page 94), so that the entry stays in the routing table.
NAT refresh
For outgoing calls
The phone requires its public IP address so that it can receive the voice data for the other
caller and you can hear the other caller. If NAT is activated, the phone does not know the
public IP address.
If the NAT for your router (symmetric NAT) is the only NAT between the phone and the SIP
server, setting a value between 5056 and 5071 for your phone's SIP port will be enough
(page 102). The phone's STUN server must be deactivated. Set
below).
If the NAT for the router is not the only one between the phone and the SIP server (e.g. if
your router is connected to the Internet via another LAN and the router for this LAN has an
asymmetric NAT), there are two options:
u
Your VoIP provider makes an outbound proxy available: the phone directs its request to
establish a connection to the outbound proxy (instead of to the SIP proxy). This provides
the data packet with the public IP address.
to
(see
STUN enabled
No
93
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