manual.lyx

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

 of 128 kbits/second.
 Using the GSM codec, this flow is reduced to 13 kbits/second, without significa
nt loss in quality.
 Currenty the best bitrate/quality compromise is achievied by using the
 speex codec.
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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.
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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.
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\begin_inset LatexCommand \label{paramaudio}

\end_inset 

Audio parameters
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In this section you will find parameters related to your sound equipment.
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Sound card choice: if you have several soundcards on your pc, you can select
 the one to be used by linphone.
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Source choice: in this combo box you can choose the recording source for
 your voice.
 In most case it will be the microphone (mic).
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Address book
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The address book let you store and recall name and sip addresses of people.
 
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When adding a new contact, a little contact box is displayed in where you
 are prompted for information about the person, mainly of course his sip
 address.
 Additionnaly you can toggle the 
\begin_inset Quotes fld
\end_inset 

send subscription
\begin_inset Quotes frd
\end_inset 

 button to ask 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.
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Using sip proxies and registrar.
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Registering on a SIP server can be usefull in two main cases:
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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 to 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 it exists
 a redirect or proxy sip server at <sip:myserver.org>.
 By registering to <sip:myserver.org>, your friends will be able to call
 you at the address <sip:[e-mail]> .
 The proxy or redirect server myserver.org will forward or redirect the calls
 from your friends to your exact location.
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You are behind a firewall.
 In order for UDP messages (SIP and RTP are on top of UDP) to go through
 the firewall correctly, in most cases a SIP proxy running on the firewall
 machine is necessary.
 You must indicate to linphone the proxy to be used, and tell it that this
 proxy is an outbound proxy (ie all messages must go through it).
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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 proxy address, route and your identity (also known
 as address of record).
 This information should be given to you by your sip provider you registered
 to.
 Route can be optionnal, so leave it empty in case you don't know what to
 put here.
 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.
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Behind a firewall
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Linphone is now able to work behind firewalls using a SIP proxy server.
 A SIP proxy server is responsible to forward calls from the private network
 to the external network, and vice versa.
 Two sip proxy based on oSIP library are being developed at 
\begin_inset LatexCommand \url[http://siproxd.sourceforge.net]{http://siproxd.sourceforge.net}

\end_inset 

 and 
\begin_inset LatexCommand \htmlurl[http://www.partysip.org]{http://www.partysip.org}

\end_inset 

.
 The setup of each of those sip proxies is outside the scope of this document.
 Please refer to their website.
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There is one case where a sip proxy is not needed: when you are in a network
 where your machine has a public address and the firewall is just here to
 filter incoming and outgoing packets from the external network.
 In this case, all you have to do is to open the SIP port and RTP port on
 the firewall machine.
 The SIP port is given in the property box, section SIP, and the RTP port
 is given in the property box, section RTP.
 Both can be changed, but it es strongly recommended you to let the SIP
 port unchanged (5060).
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Problems
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Connection problems
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Audio problems
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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.
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First rise up playback and recording level.
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If the voice is sometines cutted, you can modify parameter RTP->jitter compensat
ion in the property box to greater values to avoid this.
 But it increases the delay transmission.
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If linphone cannot open the audio device, check if it has the permission
 to open /dev/dsp, close all programs able to use audio device (xmms, kaiman...).
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Use alsa drivers (see 
\begin_inset LatexCommand \url[http://www.alsa-project.org]{http://www.alsa-project.org}

\end_inset 

).
 Most distributions still use the old oss kernel-official drivers, that
 have big latency problems and are often buggy.
 ALSA drivers are much better.
 
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Bugs reporting and suggestions
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First go to linphone's home page at 
\begin_inset LatexCommand \url[http://www.linphone.org]{http://www.linphone.org}

\end_inset 

 to check if you have the latest version if linphone.
\layout Standard

If linphone crashes, send a report to the mailing list, [e-mail].
 If linphone does not work, but does not crash, please ensure you have read
 all this manual before sending me a bug report at the above address.
 If you want to request something, don't hesitate to send me an email to
 the mailing list too.
 Note that video support, and conferencing are planned features.
 If someone is interested in doing translations for linphone, 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.
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Authors
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Simon MORLAT ([e-mail]) wrotes: 
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main library (coreapi)
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gnome interface (thanks to glade !)
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RTP library (oRTP)
\layout Itemize

audio/video framework and wrappers (mediastreamer)
\layout Standard

Aymeric Moizard ([e-mail]) wrotes the osip and eXosip stacks that is
 used by linphone.
 
\layout Standard

The speex codec 
\begin_inset LatexCommand \url[http://www.speex.org]{http://www.speex.org}

\end_inset 

 is a high quality low bitrate codec by Jean Marc Valin.
\layout Standard

The GSM library was written by : Jutta Degener and Carsten Bormann,Technische
 Universitaet Berlin.
\layout Standard

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
 [e-mail] http://www.arl.wustl.edu/~jaf/ See text files in gsmlib and
 lpc10-1.5 directories for further information.
\layout Standard

Icons by Pablo Marcelo Moia.
\layout Section

Thanks
\layout Standard

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

Thanks to Aymeric Moizard, for his famous oSIP library.
\layout Standard

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

Thanks to Jean Marc Valin, for his great speex codec.
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Thanks to the authors of LPC10-1.5 and GSM code.
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Thanks to Joel Barrios ( [e-mail] ) for his RPMS.
\layout Standard

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


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