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Options" on page 115
suitable for your purpose, check the Web pages mentioned in
The URL to use is determined by the type of connection you choose. If you chose to call another
party directly without any further routing by a SIP provider, you would enter a URL of the first type.
If you chose to call another party via a SIP server, you would enter a URL of the second type.
Calling in the Same Network
If you intend to call a friend or coworker belonging to the same network, you just need the correct
username and hostname to create a valid SIP URL. The same applies if this person wants to call you.
As long as there is no firewall between you and the other party, no further configuration is required.
Calling across Networks or the Internet (Static IP Setup)
If you are connected to the Internet using a static IP address, anyone who wants to call you just
needs your username and the hostname or IP address of your workstation to create a valid SIP URL,
as described in
behind a firewall that filters incoming and outgoing traffic, open the SIP port (5060) and the RTP
port (7078) on the firewall machine to enable Linphone traffic across the firewall.
Calling across Networks or the Internet (Dynamic IP Setup)
If your IP setup is not static—if you dynamically get a new IP address every time you connect to the
Internet—it is impossible for any caller to create a valid SIP URL based on your username and an IP
address. In these cases, either use the services offered by a SIP provider or use a DynDNS setup to
make sure that an external caller gets connected to the right host machine. More information about
DynDNS can be found at
Calling across Networks and Firewalls
Machines hidden behind a firewall do not reveal their IP address over the Internet. Thus, they cannot
be reached directly from anyone trying to call a user working at such a machine. Linphone supports
calling across network borders and firewalls by using a SIP proxy or relaying the calls to a SIP
provider. Refer to
necessary adjustments for using an external SIP server.
7.1.3 Configuring the Network Parameters
Most of the settings contained in the Network tab do not need any further adjustments. You should
be able to make your first call without changing them.
NAT Traversal Options
RTP Properties
114 SUSE Linux Enterprise Desktop 10 GNOME User Guide
and check the provider's registration documentation. For a list of providers
"Calling in the Same Network" on page
Wikipedia.org
"Configuring the SIP Options" on page 115
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.
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,
"For More Information" on page
114. If you or the calling party are located
(http://en.wikipedia.org/wiki/Dynamic_DNS).
for a detailed description of the
119.
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