manual.lyx

基于osip、eXosip、speex、ffmpeg的VoIP源代码

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Codecs
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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 significa
nt loss of quality.
 Currently the best bitrate/quality compromise is achieved by using the
 speex codec.
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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.
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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.
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\begin_inset LatexCommand \label{paramaudio}

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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 sound cards 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 cases it will be the microphone (mic).
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\begin_layout Section
Address book
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\begin_layout Standard
The address book lets you store and recall names and sip addresses of people.
 
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\begin_layout Standard
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 
\begin_inset Quotes fld
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send subscription
\begin_inset Quotes frd
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 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.
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Using SIP proxies and registrar.
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\begin_layout Standard
Registering with a SIP server can be useful 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 with 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 that there
 exists a redirect or proxy SIP server at <sip:myserver.org>.
 By registering as 'bob' with <sip:myserver.org>, your friends will be able
 to call you at the address <sip:[e-mail]> .
 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.
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\begin_layout Standard
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 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.
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Behind a firewall
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\begin_layout Standard
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).
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\begin_layout Section
Problems
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\begin_layout Subsection
Connection problems
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\begin_layout Standard
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).
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\begin_layout Subsection
Audio problems
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\begin_layout Quotation
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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\begin_layout Itemize
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.
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\begin_layout Itemize
If the voice is sometimes interrupted, you can modify parameter RTP->jitter
 compensation in the property box to greater values to avoid this.
 But this will also increase the transmission delay.
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\begin_layout Itemize
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.
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Use ALSA drivers (see 
\begin_inset LatexCommand \url[http://www.alsa-project.org]{http://www.alsa-project.org}

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).
 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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\begin_layout Section
Bugs reporting and suggestions
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\begin_layout Standard
First go to linphone's home page at 
\begin_inset LatexCommand \url[http://www.linphone.org]{http://www.linphone.org}

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 to check if you have the latest version if linphone.
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\begin_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
 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.
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\begin_layout Section
Authors
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\begin_layout Standard
Simon MORLAT ([e-mail]) wrote: 
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\begin_layout Itemize
main library (coreapi)
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\begin_layout Itemize
gnome interface (thanks to glade !)
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RTP library (oRTP)
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audio/video framework and wrappers (mediastreamer)
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\begin_layout Standard
Aymeric Moizard ([e-mail]) wrotes the osip and eXosip stacks that is
 used by linphone.
 
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\begin_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.
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\begin_layout Standard
The GSM library was written by : Jutta Degener and Carsten Bormann,Technische
 Universitaet Berlin.
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\begin_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.
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Icons by Pablo Marcelo Moia.
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\begin_layout Section
Thanks
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\begin_layout Standard
Thanks to Daemon Chaplin, for having done Glade, the gtk interface builder.
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\begin_layout Standard
Thanks to Aymeric Moizard, for his famous oSIP library.
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Thanks to Florian Winstertein, for the console interface of linphone.
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\begin_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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\begin_layout Standard
Thanks to Joel Barrios ( [e-mail] ) for his RPMS.
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Thanks to Pablo Marcelo Moia for the great icons he has made for linphone.
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