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7.1 Configuring Linphone
Before you start using Linphone there are some basic decisions
to make and some configuration tasks to complete. First, determine
and configure the run mode of Linphone, determine the connection
type to use, then start the Linphone configuration ()
to make the necessary adjustments.
7.1.1 Determining the Run
Mode of Linphone
Linphone can be run in two different modes, depending on the
type of desktop you run and on its configuration.
- Normal Application
- After the Linphone software has been installed,
it can be started via the GNOME and KDE application menus or via
the command line. When Linphone is not running, incoming calls cannot
be received.
- GNOME Panel Applet
- Linphone can be added to the GNOME panel. Right-click
an empty area in the panel, select ,
and select Linphone. Linphone is then permanently added to the panel
and automatically started on login. As long as you do not receive
any incoming calls, it runs in the background. As soon as you get
an incoming call, the main window opens and you can receive the
call. To open the main window to call someone, just click the applet
icon.
7.1.2 Determining the
Connection Type
There are several different ways to make a call in Linphone.
How you make a call and how you reach the other party is determined
by the way you are connected to the network or the Internet.
Linphone uses the session initiation protocol (SIP) to establish
a connection with a remote host. In SIP, each party is identified
by a SIP URL: sip:username@hostname
username is your login on your
Linux machine and hostname the name of
the computer you are using. If you use a SIP provider, the URL would
look like the following example: sip:username@sipserver
username is the username chosen
when registering at a SIP server. sipserver is
the address of the SIP server or your SIP provider. For details
on the registration procedure, refer to Configuring the SIP
Options and
check the provider's registration documentation. For a list of providers
suitable for your purpose, check the Web pages mentioned in For More Information.
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 Calling
in the Same Network. If you or the calling party are
located 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 Wikipedia.org.
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 Configuring the SIP
Options for
a detailed description of the necessary adjustments for using an
external SIP server.
7.1.3 Configuring the Network
Parameters
Most of the settings contained in the tab
do not need any further adjustments. You should be able to make
your first call without changing them.
- 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.
7.1.4 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 as
it is. Use to determine which
recording source should be used. In most cases, this would be a
microphone (micro). To select a custom ring sound,
use to choose one and test your choice
using . Click to
accept your changes.
7.1.5 Configuring the SIP
Options
The 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 to learn about the registration
procedure.
- Authentication Information
- To register at a remote SIP server, provide certain
authentication data, such as a password and username. Linphone stores
this data once provided. To discard this data for security reasons, click .
The list can be filled
with several addresses of remote SIP proxies or service providers.
Adding
a SIP Proxy and Registering at a Remote SIP Server
-
Choose a suitable
SIP provider and create a user account there.
-
Start Linphone.
-
Go to .
-
Click to open a registration form.
-
Fill in the
appropriate values for , , and .
If working from behind a firewall, always select and
enter an appropriate value for .
This resends the original registration data after a given time to
keep the firewall open at the ports needed by Linphone. Otherwise,
these ports would automatically be closed if the firewall did not
receive any more packages of this type. Resending the registration data
is also needed to keep the SIP server informed about the current
status of the connection and the location of the caller. For , enter the SIP URL that should be used for local calls.
To use this server also as a SIP proxy, enter the same data for . Finally, add an optional route, if needed, and
leave the dialog with .
7.1.6 Configuring the
Audio Codecs
Linphone supports a several codecs for the transmission of
voice data. Set your connection type and choose your preferred codecs
from the list window. Codecs not suitable for your current connection type
are red and cannot be selected.
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