Linphone is a simple web-phone. It allows you to make two party-calls using an IP network like the internet. What you need to run Linphone is :
Linphone is free, it is released under GNU Public License.
WARNING:
This software is provided with NO WARRANTY see file COPYING for details.
This means you MUST NOT use linphone for confidential conversations: there is NO encryption,
so it is easy for any bad-intentionned person to catch audio streams.
Note also that it is not recommended to run Linphone as root.
Linphone can be run as three different ways:
Linphone uses the Session Initiation Protocol to establish the connection with a remote host.
To call somebody, you must provide to linphone a SIP URL:
user_name@host_name(or IP address).
Note that Sip is a new telecommunication protocol designed to be simple, and it is not compatible with H323 at all.
The user_name is a login name in a linux machine. host_name is the name of a machine, if known by name servers. If not use IP address directly. If you are using a temporary connection to the internet, for example if you are using a modem, it is important that you run linphone after being connected, since linphone needs to find your IP address. If you are connected to multiple networks, linphone will warn you, and you will be recommended to check in the property box if it is using the good network. The person you want to phone must have linphone running in order to receive the call. As SIP is an open protocol, Linphone should work with other sip-compatible web phones.
Here is a "simple" way to proceed:
Requirements:
Since version 0.3.0, linphone comes with a test program called 'sipomatic'. Sipomatic can answer automatically
to calls from linphone. To do this:
Linphone has several parameters and options you can change, in order to change or improve its performance:
You can store and recall sip address using the address book (in the connection menu).
A sip address is in the form: user@domain_name. You can also add a display name, that is used only for your information.
To recall a sip address, select it in the address list, and click on the "select" button. Then you will see the address you have selected in the main window.
Just press the call button to call the person.
When you connects to the internet via a dialup connection, there is a serious problem for your friends to know your IP address in order to phone you.
For this reason you can register a fixed sip address on a SIP server called registrar or redirect server.
Your sip address will become sip:my_name@myregistrar. Your friends will be able to phone you using this address. The principle is very simple: when linphone startups, it tells to the registrar the correspondance between your sip address sip:my_name@myregistrar and your IP address. Then when your friends call you at the above sip address, the registrar gives them your exact Ip address, and the call is redirected to your exact location.
To accomplish this, go to the property box, section SIP and toggle the button in front of « use registrar ». Type the url of the registrar, and choose a user name that is not too frequent to avoid name conflicts. For example it is sure that sip:jack@my_registrar is very usefull. Prefer something like
sip:jack.nicholson@my_registrar.
You may choose a password for your registration: it depends on the registrar server. Follow the instructions given on the registrar web site.
Finnaly click the OK button of the property box. Linphone will close it and immediately talk to the registrar server to inform it of your exact location. When linphone shutdowns, it will take a few seconds to unregister your location from the registrar.
A list of public registrar servers can be found at http://www.cs.columbia.edu/~hgs/sip/servers.html.
Unfortunately, many of these servers don't work anymore, maybe caused by the recent crisis in telecommunications. Some requires authentification methods that are currntly not supported by linphone.
In order for you not to lost your time, a list of working public sip servers usable with linphone is availlable on linphone web site at http://simon.morlat.free.fr/english/servers.html .
Some requires you to fill a registration form, but don't worry all servers on this list are totally free of charge.
Complete firewall support is planned for near future in linphone.
For the moment linphone can work behind a firewall in the following situation:
Since version 0.4.0, linphone uses the osip stack to implement the SIP protocol. Some responses or requests are still unsupported, but others should work fine (INVITE, REGISTER, BYE, ACK, CANCEL, 200, 486, 606, 404).
You may notice that some menus or boxes do nothing. That is normal, it will be done in furure versions.
Linphone-0.4.0 breaks compatibility with older versions. This means that you can't use a 0.4.0 version to call a 0.3.0 one and vice-versa.
Here is a list of things you can check before saying "That 's shit.":
See file BUGS in the sources for other noticed strange things.
NOTE: Linphone cannot work in local loopback: don't try to call yourself or run several linphones on the same host: it does not work. Only the call to sipomatic works in local loopback.
First go to linphone's home page at http://simon.morlat.free.fr/english/linphone.html to check if you have the latest version if linphone.
If linphone crashes, send me directly a report at
linphonebugs@free.fr
If linphone does not work, but does not crash, please ensure you have read all this manual before sending me a bug report at the
above address.
If you want to request something, don't hesitate to send me an email at the same above address. Note that video support, proxy support
and conferencing are planned features.
If someone is interested in doing translations for linphone, send me a xx.po file based on the po/linphone.pot file of the distribution. You can also translate this user manual in other languages. In any case, please contact me if you want more details.
Simon MORLAT (simon.morlat@free.fr) wrotes:
Aymeric Moizard (jack@atosc.org) wrotes the osip stack that is used by linphone.
The GSM library was written by :
Jutta Degener and Carsten Bormann,Technische Universitaet Berlin.
The LPC10-1.5 library was written by:
Andy Fingerhut
Applied Research Laboratory <-- this line is optional if
Washington University, Campus Box 1045/Bryan 509 you have limited space
One Brookings Drive
Saint Louis, MO 63130-4899
jaf@arl.wustl.edu
http://www.arl.wustl.edu/~jaf/
See text files in gsmlib and lpc10-1.5 directories for further information.
Icons by Pablo Marcelo Moia.
Translations in english by Delphine PERREAU.
Thanks to Daemon Chaplin, for having done Glade, the gtk interface builder.
Thanks to the authors of LPC10-1.5 and GSM code.
Thanks to Joel Barrios ( jbarrios@linuxparatodos.com ) for his RPMS.
Thanks to Pablo Marcelo Moia
Thanks to BlueFish developpement team, for having made bluefish, the wonderful html editor that I used to make this page.