# Linphone's User Manual

## Introduction

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 :

- a computer running the GNU/Linux operating system

- gtk+>=2.4, in order to use the graphical interface (highly recommended!). The console-only application (linphonec) does not need gtk but libreadline.

- a sound card correctly configured to use the ALSA linux sound system

- headphones or speakers

- a microphone

- a connection to a network (the Internet for example), using a modem, an ethernet card, a Wifi adapter or anything else

Since linphone needs to use the computer's sound system, before running linphone, please make sure that no other application is using the audio device.

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 SHOULD NOT use linphone for confidential conversations: there is NO encryption, so it is easy for any bad-intentioned person to monitor the audio streams, and thus your conversation. Note also that it is not recommended to run Linphone as root.

## Running linphone

Linphone can be run in three different ways:

- as a normal application: in the gnome menu, linphone should appear in the network sub-menu. If you are not running gnome, you can execute linphone directly by typing linphone in a terminal, for example. Please note, that when linphone is not running, you cannot receive calls.

- as a gnome applet: add the linphone applet by right-clicking on the gnome panel, linphone appears in the network menu. When linphone is running silently as a gnome panel, it is able to receive calls even if its window is not shown. If you want the main linphone window to appear, click on the applet. When somebody calls you, the main window is shown and you will hear the ring normally.

## Making a call

### Basic principles

Linphone uses the Session Initiation Protocol (SIP) to establish a connection with a remote host. In this protocol each caller or callee is identified by a SIP url: sip:user_name@host_name. A SIP url's syntax like an email address, with a “sip:“ prefix.

User_name is probably your login account on a Unix machine, and host_name is the machines fully qualified domain name (FQDN) or IP address.

Note that SIP is a new telecommunication protocol designed to be simple, and it is not compatible with H323 at all.

### When IP address are not static, or not routable.

For that purpose, you can register to a SIP provider or SIP proxy. There exist several SIP proxies on the net, and some of them are free. See, for example, http://iptel.org. You'll have to get an account on the proxy and then tell linphone to use it. In this case, the user_name will assigned to you by the VoIP provider, when you register, and host_name is the provider's host name (usually something like sip.example.com).

### Test trial: If you have no friends to call at the moment (because it is too late for example), but would like to know if linphone is really working.

Since version 0.3.0, linphone comes with a test program called 'sipomatic'. Sipomatic can answer automatically calls from linphone. To do this:

- run sipomatic from a terminal. Sipomatic does not have a graphical interface, but you don't have to interact with it, so it doesn't need one.

- Then type the following SIP url in the main window of linphone: sip:robot@127.0.0.1:5064 . 127.0.0.1 is the local address for your computer, and robot is the name to use for calling sipomatic. 5064 is the port that sipomatic is listening to. Normally you should always use 5060 (i.e the default port when no port is specified) to call somebody, but sipomatic is the exception: it runs on port 5064. The reason for this is that linphone itself already runs on 5060, and you cannot have two applications running on the same port, at the same time and on the same machine.

- Then press the call button. After one second, sipomatic should answer to your call and you should hear a short announcement.

## Call parameters

### Network

Linphone allows you to set your firewall address (see section 7) or a stun server address that might help linphone calling and receiving calls.

Linphone supports ipv6: you can enable it by toggling the “Enable ipv6” checkbox. However it can support Ipv6 and Ipv4 together.

### RTP

RTP (Real Time Protocol) is a protocol used to send media streams over networks.

- RTP port: linphone uses default port 7078 to send and receive audio streams. If you think port 7078 is used by another application, change it as you wish.

- Jitter compensation: This number represents the number of audio packets linphone is waiting for before starting to play them. If sometimes some audio packets are late, they have a greater chance to be played. Increase this parameter, if the other person's voice sounds 'chopped', in order to improve the quality of the transmission. This will however increase the delay (you will hear the remote user's talk with a few seconds delay). If, on the other hand, you are using a fast network, and you have good audio drivers, you can set this parameters down to three packets, and you will have a very small delay.

### SIP

SIP (Session Initiation Protocol) is a protocol to establish and destroy media sessions over a network. In simple words, it's responsible for controlling calls. It rings the remote user, initiates the call and terminates it when one of the two parties hangs up.

- SIP port: linphone uses default port 5060 to send and receive SIP packets. It is highly recommended by SIP's RFC to use port 5060. So, please don't change this unless you really know what you are doing.

- Use registrar: toggle this button if you need the services of a remote SIP server. See section “Registering on a remote server” for details about this.

### Codecs

Codecs are algorithms especially designed to compress voice data. For example, digitized voice in 16bit / 8000 Hz represents a data flow of 128 kbits/second. Using the GSM codec, this flow is reduced to 13 kbits/second, without significant loss of quality. Currently the best bitrate/quality compromise is achieved by using the speex codec.

- Codec choice: linphone can use several codecs. Use buttons at the bottom of the codec list to put them in order of preference. Note, that according to your network connection type, some codecs are not usable. They appear in red and they are not selectable. You can decide to use or not a usable codec (in blue) by changing its status with the enable/disable buttons at the bottom of the list.

- Connection type: select how you are connected to the network you want to use (in most cases that will be the internet). This helps linphone configure itself according to the bandwidth of your connection type. For example some some high-bitrate codecs will be automatically disabled, if you select connection with a 56k modem.

### Audio parameters

In this section you will find parameters related to your sound equipment.

- Sound card choice: if you have several sound cards on your PC, you can select the one to be used by linphone.

- Source choice: in this combo box you can choose the recording source for your voice. In most cases it will be the microphone (mic).

## Address book

The address book lets you store and recall names and sip addresses of people.

When adding a new contact, a little contact box is displayed, where you can fill in information about the person, mainly of course his SIP address. Additionally you can toggle the “send subscription” button if you want the person to keep you informed of his online status (ready, busy, gone...). You can also choose to reject subscription from this person, meaning that he will not be informed of your online status.

## Using SIP proxies and registrar.

Registering with a SIP server can be useful in two main cases:

- Your machine does not have a public domain name, which prevents other users to call you as they can't guess your IP address. In this case, you can register with a proxy or redirect SIP server to get a public SIP address. For example, you are <sip:bob@no-host-name> and let's suppose that there exists a redirect or proxy SIP server at <sip:myserver.org>. By registering as 'bob' with <sip:myserver.org>, your friends will be able to call you at the address <sip:bob@myserver.org> . Of course, the user_name assigned to you by the SIP server may be different from your login name on the local machine. It can even be a number resembling a regular (PSTN) phone number, eg. 5002000307. The proxy or redirect server myserver.org will forward or redirect the calls from your friends to your exact location.

With linphone>=1.0.0 you can choose to use several proxies simultaneously. Go to the property box, section sip, and click on add proxy. You'll be prompted for a proxy address, route and your identity (also known as address of record). This information should be given to you by the SIP provider you registered with. Route can be omitted (ie. is optional), so leave it empty in case you don't know what to put there. The identity is the SIP address you are known by the proxy. Other users on the network are supposed to always be able to find you at this SIP address.

## Behind a firewall

In some cases the configuration of your network is such that linphone (or any other SIP phone program) cannot tell with certainty, how other computers on the network can talk to your computer. This is usually the case, when your machine is behind a firewall/router that uses the Network Address Translation (NAT) protocol (RFC 1631). In order to find out linphone can use the services of a "Simple Traversal of User datagram through Network address translators" (STUN) server (RFC 3489). If you are behind a NAT firewall/router put the name of your STUN server in the respective field. This information is usually provided to you by your SIP proxy/server and most times, assuming that your SIP server is 'sip.example.com', it looks like 'stun.example.com'. You may also have to specify the port your STUN server listens to (default 3478).

## Problems

### Connection problems

Firewalls are the main cause of problems in call routing. Check that udp ports are opened and masqueraded, and subscribe to a SIP proxy outside: most proxies are able to handle firewalls issues themselves. If not possible read section 7 (Behind a firewall).

### Audio problems

> Linphone seems to connect to the remote SIP url, it rings, but when the callee answers, nothing happens and we can't hear each other.

- Using your audio mixer program (eg. 'alsamixer', 'kmix', or 'aumix') make make sure the audio output is not muted and that the playback (master volume, PCM) and recording (mic) controls are set to at least their medium values.

- If the voice is sometimes interrupted, you can modify parameter RTP->jitter compensation in the property box to greater values to avoid this. But this will also increase the transmission delay.

- If linphone cannot open the audio device, check if the user has the right permissions to open /dev/dsp, and close all programs able to use audio device (xmms, kaiman...), as at this point linphone cannot share the audio device with other applications.

- Use ALSA drivers (see http://www.alsa-project.org). Most distributions still use the old OSS kernel-official drivers, that have big latency problems and are often buggy. ALSA drivers are much better.

## Bugs reporting and suggestions

First go to linphone's home page at http://www.linphone.org to check if you have the latest version if linphone.

If linphone crashes, send a report to the mailing list, linphone-users@nongnu.org. If linphone does not work, but does not crash, please ensure you have read this manual in its entirety before sending a bug report at the above address. You can also send e-mail to the mailing list to request a specific feature, that you think is missing from linphone. Note that video support, and conferencing are planned features. If someone is interested in helping with the translations of linphone to other languages, s/he can 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.

## Authors

Simon MORLAT (simon.morlat@linphone.org) wrote:

- main library (coreapi)

- gnome interface (thanks to glade !)

- RTP library (oRTP)

- audio/video framework and wrappers (mediastreamer)

Aymeric Moizard (jack@atosc.org) wrotes the osip and eXosip stacks that is used by linphone.

The speex codec http://www.speex.org is a high quality low bitrate codec by Jean Marc Valin.

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.

## Thanks

Thanks to Daemon Chaplin, for having done Glade, the gtk interface builder.

Thanks to Aymeric Moizard, for his famous oSIP library.

Thanks to Florian Winstertein, for the console interface of linphone.

Thanks to Jean Marc Valin, for his great speex codec.

Thanks to the authors of LPC10-1.5 and GSM code.

Thanks to Joel Barrios ( jbarrios@-NO-SPAM-linuxparatodos.com ) for his RPMS.

Thanks to Pablo Marcelo Moia for the great icons he has made for linphone.