manual.lyx

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

#LyX 1.3 created this file. For more info see http://www.lyx.org/
\lyxformat 221
\textclass docbook
\language english
\inputencoding default
\fontscheme default
\graphics default
\paperfontsize default
\spacing single 
\papersize Default
\paperpackage a4
\use_geometry 0
\use_amsmath 0
\use_natbib 0
\use_numerical_citations 0
\paperorientation portrait
\secnumdepth 3
\tocdepth 3
\paragraph_separation indent
\defskip medskip
\quotes_language french
\quotes_times 2
\papercolumns 1
\papersides 1
\paperpagestyle default

\layout Title

LINPHONE 'S USER MANUAL
\layout Author

Simon Morlat
\layout Date

July, 24th 2004
\layout Section

Introduction
\layout Standard

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 :
\layout Itemize

Linux 
\layout Itemize

Gnome >=2.4 with gtk+>=2.4, to use the graphical interface (higly recommended!)
\layout Itemize

a sound card correctly configured enabled by alsa drivers.
\layout Itemize

headphones or speakers.
\layout Itemize

a microphone
\layout Itemize

a connection to a network (the Internet for example), using a modem, an
 ethernet card or anything.
\layout Standard

You should close any application that is using the audio device before running
 linphone.
 
\layout Standard

Linphone is free, it is released under 
\emph on 
GNU Public License
\emph default 
.
\layout Standard


\emph on 
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.
\layout Section

Running linphone
\layout Standard

Linphone can be run as three different ways:
\layout Itemize

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.
\layout Itemize

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.
\layout Section

Making a call
\layout Subsection

Basic principles
\layout Standard

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_nam
e@host_name.
 A sip url is very closed in syntax to an email address, excepted the 
\begin_inset Quotes sld
\end_inset 

sip:
\begin_inset Quotes sld
\end_inset 

 prefix.
\layout Standard

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.
\layout Standard

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

When IP address are not static
\layout Standard

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.
 
\layout Subsection

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
\layout Standard


\begin_inset LatexCommand \label{sipomatic}

\end_inset 

Since version 0.3.0, linphone comes with a test program called '
\emph on 
sipomatic
\emph default 
'.
 Sipomatic can answer automatically to calls from linphone.
 To do this:
\layout Itemize

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.
 
\layout Itemize

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.
\layout Itemize

Then press the call button.
 After one second, sipomatic should answer to your call and you should hear
 a short annoncement.
\layout Section


\begin_inset LatexCommand \label{params}

\end_inset 

Call parameters
\layout Subsection


\begin_inset LatexCommand \label{paramnetwork}

\end_inset 

Network
\layout Subsection


\begin_inset LatexCommand \label{paramrtp}

\end_inset 

RTP
\layout Standard

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

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.
\layout Itemize

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.
\layout Subsection


\begin_inset LatexCommand \label{paramsip}

\end_inset 

SIP
\layout Standard

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.
\layout Itemize

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.
\layout Itemize

Use registrar: toggle this button if you need the services of a remote SIP
 server.
 See section 
\begin_inset Quotes eld
\end_inset 

Registering on a remote server
\begin_inset Quotes erd
\end_inset 

 for details about this.
\layout Subsection


\begin_inset LatexCommand \label{paramcodec}

\end_inset 

Codecs
\layout Standard

Codecs are algorithms especially designed to compress voice data.
 For example, digitalised voice in 16bit / 8000 Hz represents a data flow