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linphone源码-1.3.5.tar.gz,linphone源码-1.3.5.tar.gz

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  <articleinfo>
   <title>
LINPHONE 'S USER MANUAL
  </title>
  <author>
Simon Morlat
  </author>
  <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>
Linux 
   </para>

  </listitem>
   <listitem>
   <para>
Gnome &gt;=2.4 with gtk+&gt;=2.4, to use the graphical interface (higly recommended!)
   </para>

  </listitem>
   <listitem>
   <para>
a sound card correctly configured enabled by alsa drivers.
   </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 or anything.
   </para>

  </listitem>

  </itemizedlist>
  <para>
You should close any application that is using the audio device before running linphone. 
  </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-intentionned person to catch audio streams. 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 as three different ways:
  </para>
  <itemizedlist>
   <listitem>
   <para>
as normal application: in the gnome menu, linphone should appear in network sub-menu. If you are not running gnome, you can execute linphone by typing linphone in a terminal, for example. When linphone is not running, you cannot receive calls.
   </para>

  </listitem>
   <listitem>
   <para>
as a gnome applet: by clicking on the gnome panel with the right button, add the applet. Linphone appears in the network menu. By running silently in the panel, linphone is able to receive calls even if its window is not shown. If you want the main window to appear, click on the applet. When a call arrives, 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 to establish the 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 is very closed in syntax to an email address, excepted the &ldquo;sip:&ldquo; prefix.
   </para>
   <para>
User_name is like a login name on an Unix machine, and host_name is the name or the IP address of the machine the user can be joined.
   </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
   </title>
   <para>
For that purpose, you can register to a sip proxy. There exists several sip proxies on the net, for example see http://iptel.org, it's free. You'll just need to get an account on the proxy and then tell to linphone to use it. 
   </para>

   </sect2>

   <sect2>
    <title>
test trial: you have no friends to call at the moment (because it is too late for example), but would like to know if linphone really works
   </title>
   <para>
<anchor id="sipomatic">Since version 0.3.0, linphone comes with a test program called '<emphasis>sipomatic</emphasis>'. Sipomatic can answer automatically to calls from linphone. To do this:
   </para>
   <itemizedlist>
    <listitem>
    <para>
run sipomatic from a terminal. Dont't be surprised, sipomatic does not have a graphical interface, but you don't have to interact with it. 
    </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 a local address for your computer, and robot is the name to use for calling sipomatic. 5064 is the port where sipomatic can be joined. 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 already runs on 5060, and you cannot have two applications running on the same port, in 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 annoncement.
    </para>

   </listitem>

   </itemizedlist>

   </sect2>


  </sect1>

  <sect1 id="params">
   <title>
Call parameters
  </title>
   <sect2 id="paramnetwork">
    <title>
Network
   </title>

   </sect2>

   <sect2 id="paramrtp">
    <title>
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 want.
    </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 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.
    </para>

   </listitem>

   </itemizedlist>

   </sect2>

   <sect2 id="paramsip">
    <title>
SIP
   </title>
   <para>
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.
   </para>
   <itemizedlist>
    <listitem>
    <para>
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.
    </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">
    <title>
Codecs
   </title>
   <para>
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 codec, this flow is reduced to 13 kbits/second, without significant loss in quality. Currenty the best bitrate/quality compromise is achievied by using the speex codec.
   </para>
   <itemizedlist>
    <listitem>
    <para>
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, 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.
    </para>

   </listitem>
    <listitem>
    <para>
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. For example some some high-bitrate codecs will be automatically disabled if you select 56k modem.
    </para>