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 | Enter the signaling port to send and receive the SIP message |
Port | for building a session. The default value is 5060 and this must |
| match with the peer Registrar when making VoIP calls. |
RTP Start | Specifies the start port for the RTP stream. The default value is |
| 40000. |
RTP End | Specifies the end port for the RTP stream. The default value is |
| 50000. |
STUN | Enable or disable the STUN server feature. |
Server | Enter the STUN server IP address if you enabled the STUN |
Address | server feature. |
NAT Keep | Check the checkbox and enter the time period for the NAT to |
Alive Time | keep alive. |
Rport | 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. |
T.38 Fax | Check the checkbox if the remote end also supports FAX |
Protocol | function. |
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