manual.sgml

基于osip、eXosip、speex、ffmpeg的VoIP源代码

<title><!-- anchor id="paramcodec" -->Codecs</title>
<para>Codecs are algorithms especially designed to compress voice data. For example, digitized 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 of quality. Currently the best bitrate/quality compromise is achieved by using the speex codec.</para><itemizedlist>
<listitem><para>Codec choice: linphone can use several codecs. Use buttons at the bottom of the codec list to put them in 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 at the bottom of the list.</para></listitem><listitem><para>Connection type: select how you are connected to the network you want to use (in most cases that will be the internet). This helps linphone configure itself according to the bandwidth of your connection type. For example some some high-bitrate codecs will be automatically disabled, if you select connection with a 56k modem.</para></listitem></itemizedlist></sect2><sect2 id="paramaudio" >
<title><!-- anchor id="paramaudio" -->Audio parameters</title>
<para>In this section you will find parameters related to your sound equipment.</para><itemizedlist>
<listitem><para>Sound card choice: if you have several sound cards on your PC, you can select the one to be used by linphone.</para></listitem><listitem><para>Source choice: in this combo box you can choose the recording source for your voice. In most cases it will be the microphone (mic).</para></listitem></itemizedlist></sect2></sect1><sect1>
<title>Address book</title>
<para>The address book lets you store and recall names and sip addresses of people. </para>
<para>When adding a new contact, a little contact box is displayed, where you can fill in information about the person, mainly of course his SIP address. Additionally you can toggle the &ldquo;send subscription&rdquo; button if you want 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.</para></sect1><sect1>
<title>Using SIP proxies and registrar.</title>
<para>Registering with a SIP server can be useful in two main cases:</para><itemizedlist>
<listitem><para>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 with a proxy or redirect SIP server to get a public SIP address. For example, you are &lt;sip:bob@no-host-name&gt; and let's suppose that there exists a redirect or proxy SIP server at &lt;sip:myserver.org&gt;. By registering as 'bob' with &lt;sip:myserver.org&gt;, your friends will be able to call you at the address &lt;sip:[e-mail]&gt; . Of course, the user_name assigned to you by the SIP server may be different from your login name on the local machine. It can even be a number resembling a regular (PSTN) phone number, eg. 5002000307. The proxy or redirect server myserver.org will forward or redirect the calls from your friends to your exact location.</para></listitem></itemizedlist><para>With linphone&gt;=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 a proxy address, route and your identity (also known as address of record). This information should be given to you by the SIP provider you registered with. Route can be omitted (ie. is optional), so leave it empty in case you don't know what to put there. 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.</para></sect1><sect1>
<title>Behind a firewall</title>
<para>In some cases the configuration of your network is such that linphone (or any other SIP phone program) cannot tell with certainty, how other computers on the network can talk to your computer. This is usually the case, when your machine is behind a firewall/router that uses the Network Address Translation (NAT) protocol (RFC 1631). In order to find out linphone can use the services of a "Simple Traversal of User datagram through Network address translators" (STUN) server (RFC 3489). If you are behind a NAT firewall/router put the name of your STUN server in the respective field. This information is usually provided to you by your SIP proxy/server and most times, assuming that your SIP server is 'sip.example.com', it looks like 'stun.example.com'. You may also have to specify the port your STUN server listens to (default 3478).</para></sect1><sect1>
<title>Problems</title>
<sect2>
<title>Connection problems</title>
<para>Firewalls are the main cause of problems in call routing. Check that udp ports are opened and masqueraded, and subscribe to a SIP proxy outside: most proxies are able to handle firewalls issues themselves. If not possible read section 7 (Behind a firewall).</para></sect2><sect2>
<title>Audio problems</title>
<blockquote>
<para>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.</para>
</blockquote><itemizedlist>
<listitem><para>Using your audio mixer program (eg. 'alsamixer', 'kmix', or 'aumix') make make sure the audio output is not muted and that the playback (master volume, PCM) and recording (mic) controls are set to at least their medium values.</para></listitem><listitem><para>If the voice is sometimes interrupted, you can modify parameter RTP-&gt;jitter compensation in the property box to greater values to avoid this. But this will also increase the transmission delay.</para></listitem><listitem><para>If linphone cannot open the audio device, check if the user has the right permissions to open /dev/dsp, and close all programs able to use audio device (xmms, kaiman...), as at this point linphone cannot share the audio device with other applications.</para></listitem><listitem><para>Use ALSA drivers (see <ulink url="http://www.alsa-project.org">http://www.alsa-project.org</ulink>). Most distributions still use the old OSS kernel-official drivers, that have big latency problems and are often buggy. ALSA drivers are much better. </para></listitem></itemizedlist></sect2></sect1><sect1>
<title>Bugs reporting and suggestions</title>
<para>First go to linphone's home page at <ulink url="http://www.linphone.org">http://www.linphone.org</ulink> to check if you have the latest version if linphone.</para>
<para>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 this manual in its entirety before sending a bug report at the above address. You can also send e-mail to the mailing list to request a specific feature, that you think is missing from linphone. Note that video support, and conferencing are planned features. If someone is interested in helping with the translations of linphone to other languages, s/he can 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.</para></sect1><sect1>
<title>Authors</title>
<para>Simon MORLAT ([e-mail]) wrote: </para><itemizedlist>
<listitem><para>main library (coreapi)</para></listitem><listitem><para>gnome interface (thanks to glade !)</para></listitem><listitem><para>RTP library (oRTP)</para></listitem><listitem><para>audio/video framework and wrappers (mediastreamer)</para></listitem></itemizedlist><para>Aymeric Moizard ([e-mail]) wrotes the osip and eXosip stacks that is used by linphone. </para>
<para>The speex codec <ulink url="http://www.speex.org">http://www.speex.org</ulink> is a high quality low bitrate codec by Jean Marc Valin.</para>
<para>The GSM library was written by : Jutta Degener and Carsten Bormann,Technische Universitaet Berlin.</para>
<para>The LPC10-1.5 library was written by: Andy Fingerhut Applied Research Laboratory &lt;-- 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.</para>
<para>Icons by Pablo Marcelo Moia.</para></sect1><sect1>
<title>Thanks</title>
<para>Thanks to Daemon Chaplin, for having done Glade, the gtk interface builder.</para>
<para>Thanks to Aymeric Moizard, for his famous oSIP library.</para>
<para>Thanks to Florian Winstertein, for the console interface of linphone.</para>
<para>Thanks to Jean Marc Valin, for his great speex codec.</para>
<para>Thanks to the authors of LPC10-1.5 and GSM code.</para>
<para>Thanks to Joel Barrios ( [e-mail] ) for his RPMS.</para>
<para>Thanks to Pablo Marcelo Moia for the great icons he has made for linphone.</para>
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