manual.sgml

linphone 网络电话 linphone 网络电话 linphone 网络电话

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<title>Linphone's User Manual</title>
<date>July, 24th 2004</date></articleinfo><sect1>
<title>Introduction</title>
<para>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 :</para><itemizedlist>
<listitem><para>a computer running the GNU/Linux operating system</para></listitem><listitem><para>gtk+&gt;=2.4, in order to use the graphical interface (highly recommended!). The console-only application (linphonec) does not need gtk but libreadline.</para></listitem><listitem><para>a sound card correctly configured to use the ALSA linux sound system</para></listitem><listitem><para>headphones or speakers</para></listitem><listitem><para>a microphone</para></listitem><listitem><para>a connection to a network (the Internet for example), using a modem, an ethernet card, a Wifi adapter or anything else</para></listitem></itemizedlist><para>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. </para>
<para>Linphone is free, it is released under <emphasis>GNU Public License</emphasis>.</para>
<para><emphasis>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.</emphasis></para></sect1><sect1>
<title>Running linphone</title>
<para>Linphone can be run in three different ways:</para><itemizedlist>
<listitem><para>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.</para></listitem><listitem><para>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.</para></listitem></itemizedlist></sect1><sect1>
<title>Making a call</title>
<sect2>
<title>Basic principles</title>
<para>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 &ldquo;sip:&ldquo; prefix.</para>
<para>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.</para>
<para>Note that SIP is a new telecommunication protocol designed to be simple, and it is not compatible with H323 at all.</para></sect2><sect2>
<title>When IP address are not static, or not routable.</title>
<para>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). </para></sect2><sect2>
<title>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.</title>
<para id="sipomatic" ><!-- anchor id="sipomatic" -->Since version 0.3.0, linphone comes with a test program called '<emphasis>sipomatic</emphasis>'. Sipomatic can answer automatically calls from linphone. To do this:</para><itemizedlist>
<listitem><para>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. </para></listitem><listitem><para>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.</para></listitem><listitem><para>Then press the call button. After one second, sipomatic should answer to your call and you should hear a short announcement.</para></listitem></itemizedlist></sect2></sect1><sect1 id="params" >
<title><!-- anchor id="params" -->Call parameters</title>
<sect2 id="paramnetwork" >
<title><!-- anchor id="paramnetwork" -->Network</title>
<para>Linphone allows you to set your firewall address (see section 7) or a stun server address that might help linphone calling and receiving calls.</para>
<para>Linphone supports ipv6: you can enable it by toggling the &ldquo;Enable ipv6&rdquo; checkbox. However it can support Ipv6 and Ipv4 together.</para></sect2><sect2 id="paramrtp" >
<title><!-- anchor id="paramrtp" -->RTP</title>
<para>RTP (Real Time Protocol) is a protocol used to send media streams over networks.</para><itemizedlist>
<listitem><para>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.</para></listitem><listitem><para>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.</para></listitem></itemizedlist></sect2><sect2 id="paramsip" >
<title><!-- anchor id="paramsip" -->SIP</title>
<para>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.</para><itemizedlist>
<listitem><para>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.</para></listitem><listitem><para>Use registrar: toggle this button if you need the services of a remote SIP server. See section &ldquo;Registering on a remote server&rdquo; for details about this.</para></listitem></itemizedlist></sect2><sect2 id="paramcodec" >