LINPHONE 'S USER MANUAL
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# Call parameters

## Network

- List of network interfaces: you must choose a network interface to use with linphone. If you want to contact somebody on the internet, you should choose the network interface that connects your computer to the internet. For example, if you are using a 56k modem, it should be interface 'ppp0'. If you are not connected to any network, only the local interface called "lo" will appear in the list. The only thing you can do in this case is to call sipomatic.

- Connection type: select here the way you are connected to the network you want to use (in most case the internet). This help linphone in configuring itself according to the bandwidth of your connection type.

## RTP

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

- RTP port: linphone uses default port 7000 to send and receive audio streams. If you think port 7000 is used by another application, change it as you want.

- 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 more chance for being played. Increase this parameter if you hear 'cutted voice' to improve the quality of the transmission, but it will increase the delay (you will hear the voice of the remote user a few second later). On the other hand, if you are using a perfect network, and if you have good audio drivers, you can set this parameters down to three packets, and so you will have a short delay.

## SIP

SIP (Session Initiation Protocol) is a protocol to establish media sessions over a network. In simpler words, this is the thing that makes the ring at the remote user, starts 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. This is higly recommended by SIP 's rfc to use port 5060. So don't change this unless you really know what you are doing.

- Use registrar: toggle this button if you want to suscribe services to a remote sip server. Those services can be: redirection, proxy or outbound proxy. See section “Registering on a remote server” for details about this.

## Codecs

Codecs are algorithms especially designed to compress voice data. For example, digitalised voice in 16bit / 8000 Hz represents a data flow of 128 kbits/second. Using the GSM vocoder, this flow is reduced to 13 kbits/second, without significant loss in quality.

- Codec choice: linphone can use several codecs. Use buttons on the bottom of the codec list to put them in an order of preference. Note, that according to your network connection type (given in the network section), 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 on the bottom of the list.

## Audio parameters

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

- Drivers choice: in linux, you have two different kinds of soundcard drivers: OSS (called also kernel drivers) and ALSA. A program can use ALSA driver the same way as OSS ones, but ALSA drivers have better performance when used passing trough the ALSA library. So if you have alsa drivers (names begin by snd_*), select the ALSA mode. If you don't know, choose OSS.

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

- Auto-kill option: by toggling this option, linphone will try to stop sound daemons (esd and artsd), that may lock permanently your audio device, and so cause linphone to fail in open the audio device when it needs it. It is recommended to have this option on.

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