Novell LINUX ENTERPRISE DESKTOP 10 SP1 - GNOME 23-05-2007, Руководство пользователя, Страница: 162 / 280

162 SUSE Linux Enterprise Desktop 10 SP1 GNOME User Guide
no
vd
ocx
(e
n)
6 Ap
ril 20
07
NAT Traversal Options
Enable this option only if you find yourself in a private network behind a firewall and if you do
not use a SIP provider to route your calls. Select the check box and enter the IP address of the
firewall machine in dot notation, for example,
192.168.34.166
.
RTP Properties
Linphone uses the real-time transport protocol (RTP) to transmit the audio data of your calls.
The port for RTP is set to
7078
and should not be modified, unless you have another
application using this port. The jitter compensation parameter is used to control the number of
audio packages Linphone buffers before actually playing them. By increasing this parameter,
you improve the quality of transmission. The more packages buffered, the greater a chance for
“late comers” to be played back. On the other hand increasing the number of buffered packages
also increases the latency—you hear the voice of your counterpart with a certain delay. When
changing this parameter, carefully balance these two factors.
Other
If you use a combination of VoIP and landline telephony, you might want to use the dual tone
multiplexed frequency (DTMF) technology to trigger certain actions, like a remote check of
your voice mail just by punching certain keys. Linphone supports two protocols for DTMF
transmission, SIP INFO and RTP rfc2833. If you need DTMF functionality in Linphone,
choose a SIP provider that supports one of these protocols. For a comprehensive list of VoIP
providers, refer to
“For More Information” on page 167 .
Configuring the Sound Device
Once your sound card has been properly detected by Linux, Linphone automatically uses the
detected device as the default sound device. Leave the value of
Use sound device as it is. Use
Recording source to determine which recording source should be used. In most cases, this would be
a microphone (
micro
). To select a custom ring sound, use Browse to choose one and test your
choice using Listen . Click Apply to accept your changes.
Configuring the SIP Options
The SIP dialog contains all SIP configuration settings.
SIP Port
Determine on which port the SIP user agent should run. The default port for SIP is
5060
.
Leave the default setting unchanged unless you know of any other application or protocol that
needs this port.
Identity
Anyone who wants to call you directly without using a SIP proxy or a SIP provider needs to
know your valid SIP address. Linphone creates a valid SIP address for you.
Remote Services
This list holds one or more SIP service providers where you have created a user account. Server
information can be added, modified, or deleted at any time. See
“Adding a SIP Proxy and
Registering at a Remote SIP Server” on page 163 to learn about the registration procedure.