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Advanced
SIP Setting
This page allows you to configure the settings related to the SIP service
provider. You need to configure your VoIP Adapter to communicate with the
devices that will send and receive telephone calls over the Internet.
If you are behind a NAT modem or router, you may have to use a STUN server.
Signaling
Port
RTP Start
RTP End
STUN
Server
Address
NAT Keep
Alive Time
Rport
T.38 Fax
Protocol
Enter the signaling port to send and receive the SIP message
for building a session. The default value is 5060 and this must
match with the peer Registrar when making VoIP calls.
Specifies the start port for the RTP stream. The default value is
40000.
Specifies the end port for the RTP stream. The default value is
50000.
Enable or disable the STUN server feature.
Enter the STUN server IP address if you enabled the STUN
server feature.
Check the checkbox and enter the time period for the NAT to
keep alive.
Check this checkbox to use rport. Rport, or symmetric
signaling, is critical for operations behind some classifications
of NATs. If enabled, the SIP stack will send outbound SIP
messages on the same port that it listens on. Unless you are
familiar with single SIP port, it is better to use rport.
Check the checkbox if the remote end also supports FAX
function.
DVG-2101S User Manual
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