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<!ENTITY app "Ekiga">
<!ENTITY appversion "3.00">
<!ENTITY manrevision "3.00">
]>
<!-- Begin -->
<article id="index" lang="en_GB">
<articleinfo>
<title><application>Ekiga</application> Manual 3.00</title>

<copyright><year>2003-2008</year><holder>Damien Sandras</holder></copyright>
<copyright><year>2003-2004</year><holder>Matthias Redlich</holder></copyright>
<copyright><year>2003-2004</year><holder>Christopher Warner</holder></copyright><copyright><year>2009</year><holder>Philip Withnall (philip@tecnocode.co.uk)</holder></copyright>
<!-- translators: uncomment this:

<copyright>
<year>2002</year>
<holder>ME-THE-TRANSLATOR (Latin translation)</holder>
</copyright>

-->

<revhistory>
<revision> 
<revnumber>Ekiga Manual 3.0</revnumber> 
<date>2008-08-31</date> 
<revdescription> 
<para role="author">Damien Sandras</para> 
</revdescription> 
</revision>
<revision>
<revnumber>Ekiga Manual 2.0</revnumber> 
<date>2006-01-22</date> 
</revision>
</revhistory>

<publisher>
<publishername>Damien Sandras</publishername>
<address><email>dsandras@seconix.com</email></address>
</publisher>
<publisher>
<publishername>Christopher Warner</publishername>
<address><email>zanee@kernelcode.com</email></address>
</publisher>
<publisher>
<publishername>Matthias Redlich</publishername>
<address><email>m-redlich@t-online.de</email></address>
</publisher>

<authorgroup>
<author role="maintainer">
<firstname>Damien</firstname>
<surname>Sandras</surname>
</author>
<author>
<firstname>Christopher</firstname>
<surname>Warner</surname>
<othername>zanee</othername>
</author>
<author>
<firstname>Matthias</firstname>
<surname>Redlich</surname>
</author>
</authorgroup>

<releaseinfo>This documentation is for version 3.00 of Ekiga.</releaseinfo>

<abstract role="description">
<para><application>Ekiga</application> is an application for voice over IP, IP telephony, and video conferencing, with support for many audio and video codecs.</para>
</abstract>
</articleinfo>

<indexterm zone="index">
<primary>Ekiga</primary>
</indexterm>
<indexterm zone="index">
<primary>Ekiga</primary>
</indexterm>
<indexterm zone="index">
<primary>Ekiga</primary>
</indexterm>

<!-- INTODUCTION          -->
<!-- What is &app; -->

<section id="ekiga-introduction">
<title>Introduction</title>

<section><title>Ekiga</title>
<para><application>Ekiga</application> is a free Voice over IP, IP Telephony and Video-Conferencing application for Linux and other Unices (e.g. BSD, OpenSolaris or Mac OS X). It was written by Damien Sandras and is licenced under the GNU/GPL.</para>

<para>Ekiga is able to use modern Voice over IP protocols like SIP and H.323. It supports all major features defined by those protocols like <emphasis>call hold</emphasis>, <emphasis>call transfer</emphasis>, <emphasis>call forwarding</emphasis>, etc.. It also supports <emphasis>instant messaging</emphasis>, and <emphasis>presence</emphasis>. It also has advanced support for <emphasis>NAT traversal</emphasis>. Ekiga supports the best <emphasis>free</emphasis> audio and video codecs, and has wideband support for a superior audio quality, together with echo cancellation.</para>
</section>

<section><title>SIP and H.323</title>
<para>The Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. In November 2000, SIP was accepted as a 3GPP signalling protocol and permanent element of the IMS architecture. It is one of the leading signalling protocols for Voice over IP.</para>

<para>H.323 was originally created to provide a mechanism for transporting multimedia applications over LANs but it has rapidly evolved to address the growing needs of VoIP networks. One strength of H.323 was the relatively early availability of a set of standards, not only defining the basic call model, but in addition the supplementary services, needed to address business communication expectations. H.323 was the first VoIP standard to adopt the IETF standard RTP to transport audio and video over IP networks. H.323 is based on the ISDN Q.931 protocol and is suited for interworking scenarios between IP and ISDN, respectively between IP and QSIG. A call model, similar to the ISDN call model, eases the introduction of IP Telephony into existing networks of ISDN-based PBX systems.</para>
<para>
<graphic fileref="figures/lumi.png"/> 
</para>
</section>
</section>

<section id="ekiga-getting-started">
<title>Getting Started</title>
<para>When starting <application>Ekiga</application> for the first time the configuration assistant will show automatically. The Configuration Assistant is a step-by-step questionnaire that will guide you through all the steps involved in creating the basic configuration you will need to operate <application>Ekiga</application>. You should go through all of these steps properly, otherwise the assistant will re-appear (when it has not been completed) or <application>Ekiga</application> will not function appropriately (if some of your answers were not correct). You may run the Configuration Assistant at any time from the Edit menu.</para>

<tip><title>Tip</title><para>All settings can be changed via the preferences window at any time.</para></tip>


<section><title>Configuration Assistant Introduction</title>
<figure>
<title/>
<graphic fileref="figures/config_d1.png"/>
</figure>

<para>Throughout the entire configuration process navigation is available at the bottom of the window. You will be able to navigate through the questions using <guibutton>Back</guibutton>, <guibutton>Forward</guibutton> and <guibutton>Cancel</guibutton>. If you press <guibutton>Cancel</guibutton> during the setup <application>Ekiga</application> will not be affected by your changes and all entered information will be discarded.</para>

<para>This page welcomes you to the Configuration Assistant. There is nothing to change or edit here. Press the <guibutton>Forward</guibutton> button towards the bottom of the window to start the configuration.</para>
</section>


<section><title>Personal information</title>
<figure>
<title/>
<graphic fileref="figures/config_d2.png"/>
</figure>

<para>The Personal Information window requires you to supply personal information to use <application>Ekiga</application>. This information is displayed when connecting to other audio/video applications.</para>
</section>


<section><title>Ekiga.net Account</title>
<figure>
<title/>
<graphic fileref="figures/config_d3.png"/>
</figure>

<para>Ekiga.net is a free SIP services platform provided to <application>Ekiga</application> users. If you want to call other users and to be callable, you need a SIP address. You can get one from <ulink url="http://www.ekiga.net" type="http">http://www.ekiga.net</ulink>. Ekiga.net also offers additional services like conference rooms, voice mail or online white pages. Please see <ulink url="http://www.ekiga.net" type="http">http://www.ekiga.net</ulink> for more information.</para>

<para>Just follow the link given in the dialogue to get an account if you do not have one, then fill in your username and password. Please press <guibutton>Forward</guibutton> after having entered all required information to continue.</para>
</section>


<section><title>Ekiga Call Out Account</title>
<figure>
<title/>
<graphic fileref="figures/config_d4.png"/>
</figure>

<para><application>Ekiga</application> can be used with several Internet Telephony Service Providers. Those providers will allow calling real phones from your computer using <application>Ekiga</application> at interesting rates. We recommend you use the default <application>Ekiga</application> provider.</para>

<para>If you want to create an account and use it to call your friends and family using regular phones at interesting rates, simply create an account using the <guilabel>Get an Ekiga Call Out account</guilabel> link. Once the account has been created, you will receive a login and a password by e-mail. Simply enter them in the dialogue, and you are ready to call regular phones using <application>Ekiga</application></para> 

<para>With the default setup, you can simply use <userinput>sip:3210444555</userinput> and choose <guilabel>sip.diamondcard.us</guilabel> to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call. We encourage you to put your favourite phone numbers in the address book.</para>

<para>Just follow the link given in the dialogue to get an account if you do not have one, then fill in your username and password. Please press <guibutton>Forward</guibutton> after entering all required information to continue.</para>
</section>


<section><title>Connection Type</title>
<figure>
<title/>
<graphic fileref="figures/config_d5.png"/>
</figure>

<para><application>Ekiga</application> supports several audio and video codecs. It includes codecs with excellent quality as well as codecs with medium-to-good quality. The higher the quality of a codec, the more bandwidth it requires. Moreover, video codecs can adapt their quality to the available bandwidth. This option is necessary in the initial configuration of <application>Ekiga</application> so that it chooses the optimal codec suited to your network connection and so that it adjusts the video quality settings. If your connection type is not mentioned in the list you should select the one closest to your network connection and adjust <application>Ekiga</application> manually with the preferences window (codecs section) later on.</para>	

<para>When done, continue on with the Configuration.</para>
</section>


<section><title>Audio Devices</title>
<figure>
<title/>
<graphic fileref="figures/config_d6.png"/>
</figure>

<para><application>Ekiga</application> requires audio devices to play and record sound. The audio output device ouputs the incoming sound stream during a call. Please select the device to which that your headset or speakers are connected. The audio input device is what your microphone is connected to. These settings might be the same as the settings for the audio player if you have only one soundcard. But please note that it is also possible to record sound via another device (e.g. internal microphone in a webcam) too. This section also allows you to choose the ringing device. This device can be different from the audio output device. It allows you to hear the incoming call ringing sound event in your speakers, while having your headset connected for calls.</para>

<para>When done, continue on with the Configuration.</para>
</section>


<section><title>Video Devices</title>
<figure>
<title/>
<graphic fileref="figures/config_d7.png"/>
</figure>

<para>This step is optional and concerns users with video devices (e.g. webcams) only. If you do not have any video devices you may skip this page.</para>

<para>If you have a webcam or video device in the list you may select it here.</para>

<para>When done, continue on with the Configuration.</para>
</section>


<section><title>Configuration Complete</title>
<figure>
<title/>
<graphic fileref="figures/config_d8.png"/>
</figure>

<para>The configuration of <application>Ekiga</application> is now complete. The last window only shows a short configuration summary of the settings you have chosen. Please verify that all these settings are correct. If something is incorrect you may use the <guibutton>Back</guibutton> button in the lower-right-hand corner of the window to move to any page of the assistant and correct the mistake.</para>

<para>If everything is correct please press the <guibutton>Apply</guibutton> button to save the configuration. The assistant will be closed and the main Window of <application>Ekiga</application> will now appear. Remember, all settings can be changed via the preferences window at any time.</para>   
</section>

</section>




<section id="ekiga-basic-usage">
<title>Basic Usage</title>

<section id="ekiga-calling-and-being-called">
<title>Calling and being called</title>

<graphic fileref="figures/call_d1.png"/>

<section><title>From computer to computer (PC-to-PC)</title>
<para>If you want to call other users and to be callable, you need a SIP address. You can get a SIP address from <ulink url="http://www.ekiga.net" type="http">http://www.ekiga.net</ulink> as described above.</para>

<para>The SIP address can be used by other users to call you. Similarly, you can use the SIP address of your friends and family to call them. You can for example use <userinput>sip:dsandras@ekiga.net</userinput> to call the author of <application>Ekiga</application>.</para>

<para>You can use the online address book of <application>Ekiga</application> to find the SIP addresses of other <application>Ekiga</application> users. It is of course possible to call users who are using another provider than ekiga.net. You can actually call any user using SIP software or hardware, and registered to any public SIP provider</para>

<para>If you know the URI address of the party that you wish to call, you may enter that URI into the <guilabel>sip:</guilabel> input box at the top of the screen and press the <guibutton>Connect</guibutton> button; e.g.: <userinput>sip:foo@ekiga.net</userinput> and pressing the <guibutton>Connect</guibutton> button would call the user at that address.</para>

<para>It is also possible to call contacts using the address book, the call history or the roster. You can add contacts you call frequently to your roster, and watch their presence information in order to know when they are available. Please refer to the appropriate section of the manual for full explanations.</para>

<tip><title>Tip</title><para><application>Ekiga</application> also supports H.323 and as such can call any H.323 software or hardware. Please refer to the section related to URIs to learn more about the various types of URIs that can be used to call remote H.323 and SIP users.</para></tip>
</section>

<section><title>From computer to real phones (PC-to-Phone)</title>
<para><application>Ekiga</application> can be used with several Internet Telephony Service Providers. Those providers will allow calling real phones from your computer using <application>Ekiga</application> at interesting rates. We recommend you use the default <application>Ekiga</application> provider. You can get an account using the links in the configuration assistant as described above.</para>

<para>With the default setup, you can simply use <userinput>sip:3210444555</userinput> and select <guilabel>sip.diamondcard.us</guilabel> in the list to call the real phone number +3210444555, 32 is the country code, 10444555 is the number to call.</para>

<para>You can also dial real phone numbers from the address book. If the phone number of the contact you want to call is stored in the address book, simply select <menuchoice><guimenu>Action</guimenu><guimenuitem>Call [Ekiga Call Out]</guimenuitem></menuchoice> when the contact is highlighted. It will dial the phone number of the contact using the Ekiga Call Out account.</para>

<tip><title>Tip</title><para><application>Ekiga</application> also supports connecting to H.323 and SIP PBX systems. If the PBX at your office supports those protocols, you will be able to call real phones and be called from real phones after having connected to the PBX. Please ask for the settings from your administrator.</para></tip>
</section>

<section><title>From real phone to computer (Phone-to-PC)</title>
<para><application>Ekiga</application> can be used to receive incoming calls from regular phones. To allow this, you can simply log into your PC-to-Phone account using the <guimenu>Tools</guimenu> menu as described above, and buy a phone number in the country of your choice. <application>Ekiga</application> will ring when people will call that phone number.</para>

<tip><title>Tip</title><para>You can actually use any H.323 or SIP ITSP provider, including your own PBX at work. However we recommend using the integrated provider.</para></tip>
</section>

</section>


<section id="ekiga-manage-contacts">
<title>Managing Contacts</title>
<section><title>Adding contacts to the roster</title>

<graphic fileref="figures/roster.png"/>

<para><application>Ekiga</application> allows you to add the contacts you dial the most in the roster. It allows to call them or start a chat conversation with your friends without having to remember their URI. If supported by the service, <application>Ekiga</application> will display <emphasis>extended presence information</emphasis> about your friends. Ekiga.net supports publishing presence information for its users. Software PBX systems like <ulink url="http://www.asterisk.org" type="http">Asterisk</ulink> can report if a user is on the phone or not, and <application>Ekiga</application> will display that information in its roster.</para>

<para>You can thus use <application>Ekiga</application> to monitor lines on your PBX.</para>

<para><application>Ekiga</application> is also able to detect other <application>Ekiga</application> users on the LAN using the Bonjour technology popularised by Apple (tm) and to display them in the roster. That supposes you have a local mDNSResponder daemon running on your computer. On Linux systems this service will usually be provided by <application>Avahi</application>.</para>

<para>To add a contact to the roster, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Add Contact</guimenuitem></menuchoice>, and fill in the required fields. If the service managing the URI you entered for the contact is able to publish presence status, <application>Ekiga</application> will automatically display it.</para>

<para>If you do not know the VoIP URI of a contact, you might try searching for them using the Ekiga.net online directory. To do so, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Address Book</guimenuitem></menuchoice>, and start searching using the <guilabel>Search Filter</guilabel> feature.</para>

<tip><title>Tip</title><para>You can organise your contacts in groups in the roster.</para></tip>

<section><title>Managing contacts</title>

<graphic fileref="figures/addressbook_d1.png"/>

<para><application>Ekiga</application> allows you to look for contacts using various sources like the <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink> address book, an LDAP directory or the Ekiga.net contact directory. You can use the result of your search to start a chat, call the contact, or simply add them to your roster if you have frequent calls with them. To start looking for contacts, select <menuchoice><guimenu>Chat</guimenu><guimenuitem>Address Book</guimenuitem></menuchoice> in the menu. To your left there will be a list dialogue showing the LDAP directories as well as a list of local Address Books. The defaults are the <application>Ekiga</application> white pages, and the personal address book from <ulink url="http://www.novell.com/products/evolution" type="http">Novell Evolution</ulink>. Support for more contact sources is possible.</para>

<para><application>Ekiga</application> is able to browse any LDAP directory and use any attribute as the calling URI. For example, you could have an LDAP directory in your company, with a specific attribute containing the local extensions of all your colleagues. <application>Ekiga</application> is able to use such an LDAP directory. Simply select in <menuchoice><guimenu>Address Book</guimenu><guimenuitem>Add an LDAP Address Book</guimenuitem></menuchoice>, and fill in the required details. You can then right-click on the contact and call them using the call attribute as VoIP URI.</para>

<para>The LDAP Address Book supports a range of settings to allow it to work with any LDAPv3 directory. It allows you to choose the attribute to use for displaying a contact's name in the address book as well as a list of attributes for callng info. E.g., if the directory uses the LDAP <literal>inetOrgPerson</literal> schema you can configure the Address Book to retrieve the <literal>homePhone</literal>, <literal>mobile</literal>, and <literal>pager</literal> attributes and make those values available for calling or messaging. You can also customise a <guilabel>Filter Template</guilabel> for the default LDAP search filter, and override the default filter at any time if you need to perform a more specialised search. The browser also supports all security options for LDAP including ldaps:// (LDAP over SSL), StartTLS, and SASL authentication.</para>

<para>To refresh the list of users for a specific address book, simply click the <guibutton>Find</guibutton> button. It will search for all users in that address book. You can contact people by double-clicking on their highlighted field. You can also message them by right-clicking or by choosing the appropriate action in the <guimenu>Action</guimenu> menu of the window.</para>

<para>In certain cases you will want to search specifically for a person's name, or their call URI in the <application>Ekiga</application> white pages. The address book window allows you to apply filters when searching for contacts. When searching an LDAP directory, entering a simple name in the <guilabel>Search Filter</guilabel> field will perform an LDAP Substring search using the configured <guilabel>Filter Template</guilabel>. If you need to perform a more specialised search you can enter a complete LDAP Filter string, and it will be used instead of the configured <guilabel>Filter Template</guilabel>.</para>

<tip><title>Tip</title><para>The <application>Ekiga</application> white pages will allow you to look for users in your region. It returns a limited number of results corresponding to your search. You can then add them to your personal roster to call them later.</para></tip>
</section>

<section><title>Editing contacts</title>

<graphic fileref="figures/addressbook_d2.png"/>

<para>Local address books provided by <application>Novell Evolution</application> allow you to add new contacts, or to edit existing contacts. Each different address book allows a different set of features depending on what makes sense for the address book in question. To discover what features are possible, simply select the address book and consult the <guimenu>Action</guimenu> menu.</para>

<para>To add a contact to one of your local address books, simply select the address book you wish to add the contact to and select <menuchoice><guimenu>Action</guimenu><guimenuitem>New Contact</guimenuitem></menuchoice>. The option of adding a <guilabel>New Contact</guilabel> will appear and you may now enter their name and VoIP URI as well as other settings. When done, press <guibutton>OK</guibutton> and now your contact has been added. You can only add contacts to local address books. The contact parameters can be changed at any time by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Properties</guimenuitem></menuchoice> when the contact is highlighted. They can also be deleted by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Remove</guimenuitem></menuchoice>.</para>

<para>You can also add a contact from the white pages (or any other local or remote address book) to the roster by selecting <menuchoice><guimenu>Action</guimenu><guimenuitem>Add to local roster</guimenuitem></menuchoice> when the contact is highlighted.</para>

<para>Finally, you can edit the groups your users belong to using the <menuchoice><guimenu>Action</guimenu><guimenuitem>Properties</guimenuitem></menuchoice> dialogue when the contact is highlighted.</para>
</section>

</section>

<section id="ekiga-sending-instant-messages">
<title>Sending instant messages</title>

<graphic fileref="figures/chat_d1.png"/>

<para><application>Ekiga</application> allows you to send instant messages to remote users provided that you know their URI.</para>

<para>You can send instant messages from the roster, from the call history or from the address book. From the roster or from the call history, simply select <menuchoice><guimenu>Contact</guimenu><guimenuitem>Message</guimenuitem></menuchoice> in the main window when a contact is highlighted. From the address book window, simply select <menuchoice><guimenu>Action</guimenu><guimenuitem>Message</guimenuitem></menuchoice> when the contact is highlighted. A window is displayed: enter your text message, and press <keycap>Enter</keycap>.</para>

<tip><title>Tip</title><para>You can not exchange text messages with all protocols. <application>Ekiga</application> will only display the <guimenuitem>Message</guimenuitem> menu item when the protocol associated with the user permits it.</para></tip>
</section>


<section id="ekiga-changing-status">
<title>Updating your own status</title>

<graphic fileref="figures/status.png"/>

<para><application>Ekiga</application> allows you to publish your status to other users.</para>

<para>There are three categories of status messages: online, away and do not disturb. Each of them allows you to specify a more complete status information. Simply select <guilabel>Custom message</guilabel> in the status menu at the bottom of the main window. You can then define your extended status message that will be published using all available protocols supporting it.</para>

<tip><title>Tip</title><para>Many servers will not support relaying your extended presence information. To make sure that this feature is available with the server you are using or with the PBX you are connected to, please ask your administrator. Please note that Ekiga.net will publish your presence information.</para></tip>
</section>


<section id="ekiga-manage-calls">
<title>Managing Calls</title>

<section><title>Forwarding incoming calls</title>
<para><application>Ekiga</application> supports different policies for unanswered incoming calls. By default it displays a popup window which allows you to decide whether you want to refuse or accept the request for an incoming call. If you do not answer the call in the required time, if you are busy, or if you do not want to receive any call, <application>Ekiga</application> can forward the call to another party.</para>

<para>Notice that you need to specify an URI to which to forward calls in the preferences to be able to activate that option. Open the preferences window by choosing <menuchoice><guimenu>Edit</guimenu><guimenuitem>Preferences</guimenuitem></menuchoice> in the main window and select <guilabel>Call Options</guilabel> on the left. You will now see the appropriate section. It contains three checkboxes for the three cases described above. The URI of the party the calls shall be forwarded to can be configured separately in SIP Settings for SIP and accordingly in H.323 Settings for H.323.</para>
</section>

<section><title>Controlling the call</title>
<para><application>Ekiga</application> supports several actions which can be performed when in a call. These actions enable you to control active sessions.</para>

<itemizedlist>
<listitem>
<para>Ending a call: The communication to the remote user can be ended by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Hang up</guimenuitem></menuchoice>.</para>
</listitem>

<listitem>
<para>Holding a call: You can hold a remote party call by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Hold Call</guimenuitem></menuchoice>. This effectively pauses Video and Audio transmission; to continue transmission again you select <menuchoice><guimenu>Chat</guimenu>Retrieve Call</menuchoice></para>
</listitem>

<listitem>
<para>Suspend Audio: This effectively prevents all Audio communication to your respective party when selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Suspend Audio</guimenuitem></menuchoice>.</para>
</listitem>

<listitem>
<para>Suspend Video: This effectively prevents all Video transmission to your respective party when selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Suspend Video</guimenuitem></menuchoice>.</para>
</listitem>

<listitem>
<para>Transferring the remote party: You can transfer the remote user to another user by selecting <menuchoice><guimenu>Chat</guimenu><guimenuitem>Transfer Call</guimenuitem></menuchoice>. It is also possible to transfer an active call by right-clicking and choosing the transfer action when a contact is highlighted in the roster, in the address book or in the call history. Double-clicking or selecting the <guimenu>Contact</guimenu> menu in the main window or the <guimenu>Action</guimenu> menu in the Address Book window and choosing the transfer action will also work.</para>
</listitem>
</itemizedlist>

<tip><title>Tip</title><para>All URIs supported by <application>Ekiga</application> can be used for call transfer if the protocol supports it.</para></tip>
</section>

<section><title>Adjusting the audio and video settings</title>
<para>Your audio and video settings can be adjusted through the call panel while you are in a call. If you want to change the audio or video settings during a call, simply show the Call Panel by select <menuchoice><guimenu>View</guimenu><guimenuitem>Show Call Panel</guimenuitem></menuchoice> in the menu. The audio volume, but also the brightness, whiteness, colour and contrast of your video input device can be changed to achieve the best quality.</para>

<para>You can also change your audio and video devices during a call. Simply go in the preferences window by selecting <menuchoice><guimenu>Edit</guimenu><guimenuitem>Preferences</guimenuitem></menuchoice> in the menu, and adjust your devices in the appropriate section.</para>
</section>

<section><title>Checking the call history</title>

<graphic fileref="figures/call_history.png"/>

<para>The Call History stores information (date, duration, URI, Remote user) about all outgoing and incoming calls. They are divided into three groups — received calls, placed calls and missed calls. You can consult the call history by selecting <menuchoice><guimenu>View</guimenu><guimenuitem>Call History</guimenuitem></menuchoice> in the menu. <itemizedlist>
<listitem>
<para>Received calls contains all incoming calls which were accepted by <application>Ekiga</application></para>
</listitem>

<listitem>
<para>Placed calls keeps track of all attempts – successful or not – to call another user.</para>
</listitem>

<listitem>
<para>Missed calls shows incoming calls which timed out.</para>
</listitem>
</itemizedlist></para>

<tip><title>Tip</title><para>Double-clicking on a row in the Calls History will call back the selected user or transfer any active call to that user. Notice that you can also add the contact to your roster by selecting <menuchoice><guimenu>Chat</guimenu><guisubmenu>Contact</guisubmenu><guimenuitem>Add to local roster</guimenuitem></menuchoice> in the main menu when the call is highlighted.</para></tip>
</section>

</section>
</section>
</section>

<section id="ekiga-advanced-usage">
<title>Advanced Usage</title>

<section id="ekiga-registering-additional-accounts">
<title>Registering Additional Accounts</title>

<section><title>The accounts window</title>

<graphic fileref="figures/accounts_d1.png"/>

<para>You can open the accounts window by selecting <menuchoice><guimenu>Edit</guimenu><guimenuitem>Accounts</guimenuitem></menuchoice>. This will open the Accounts Window. The Accounts Window will allow you to add Ekiga.net, Ekiga Call Out, SIP and H.323 accounts and to register to them. An account describes the user login and password parameters to register to SIP and H.323 services. Those <emphasis>services</emphasis> can be an Internet Telephony Service provider (like ekiga.net), or an IPBX (like CISCO, Nortel, or Asterisk).</para>
</section>

<section><title>Adding an Ekiga.net account</title>

<graphic fileref="figures/accounts_ekiga_net.png"/>

<para>To add an Ekiga.net account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an Ekiga.net Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <itemizedlist>
<listitem><para><guilabel>User:</guilabel> You can enter your login.</para></listitem>
<listitem><para><guilabel>Password:</guilabel> You can enter your password.</para></listitem>
</itemizedlist></para>

<para>Ekiga.net is a free SIP services platform provided to <application>Ekiga</application> users. If you want to call other users and to be callable, you need a SIP address. You can get one from <ulink url="http://www.ekiga.net" type="http">http://www.ekiga.net</ulink>. Ekiga.net also offers additional services like conference rooms, voice mail and online white pages. Please see <ulink url="http://www.ekiga.net" type="http">http://www.ekiga.net</ulink> for more information.</para>
</section>

<section><title>Adding an Ekiga Call Out account</title>

<graphic fileref="figures/accounts_ekiga_call_out.png"/>

<para>To add an Ekiga Call Out account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an Ekiga Call Out Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <itemizedlist>
<listitem><para><guilabel>Account ID:</guilabel> You can enter your account ID.</para></listitem>
<listitem><para><guilabel>PIN Code:</guilabel> You can enter your PIN code.</para></listitem>
</itemizedlist></para>

<para>If you do not have an Ekiga Call Out account yet, you can subscribe for one using the <guilabel>Get an Ekiga.net Call Out account</guilabel> link in the dialogue. As described above, this service will allow you to call normal phones worldwide at interesting rates. Once the account has been added, you can recharge it, consult the balance history or the call history by selecting the appropriate menu item in the <guimenu>Account</guimenu> menu of the window when the account is highlighted.</para>
</section>

<section><title>Adding a SIP account</title>

<graphic fileref="figures/accounts_sip.png"/>

<para>To add a SIP account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add a SIP Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <itemizedlist>
<listitem><para><guilabel>Name:</guilabel> You can enter the account name.</para></listitem>
<listitem><para><guilabel>Registrar:</guilabel> The registrar with which you want to register. This is usually an IP address or an host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to a SIP IPBX.</para></listitem>
<listitem><para><guilabel>User:</guilabel> You can enter your login.</para></listitem>
<listitem><para><guilabel>Authentication User:</guilabel> If it is different from the user parameter you provided above. In that case, the user field will be used to control the outgoing identity for the account you are adding, while the login will be used during the authentication phase.</para></listitem>
<listitem><para><guilabel>Password:</guilabel> You can enter your password.</para></listitem>
<listitem><para><guilabel>Timeout:</guilabel> The timeout after which the registration should be refreshed.</para></listitem>
</itemizedlist></para>

</section>

<section><title>Adding an H.323 account</title>

<graphic fileref="figures/accounts_h323.png"/>

<para>To add an H.323 account, simply select <menuchoice><guimenu>Account</guimenu><guimenuitem>Add an H.323 Account</guimenuitem></menuchoice> in the menu. A dialogue will appear and allow you to enter several parameters: <itemizedlist>
<listitem><para><guilabel>Name:</guilabel> You can enter the account name.</para></listitem>
<listitem><para><guilabel>Gatekeeper:</guilabel> The gatekeeper with which you want to register. This is usually an IP address or a host name that will be given to you by your Internet Telephony Service Provider, or by your administrator if you are trying to register to an H.323 IPBX.</para></listitem>
<listitem><para><guilabel>User:</guilabel> You can enter your login.</para></listitem>
<listitem><para><guilabel>Authentication User:</guilabel> If it is different from the user parameter you provided above. In that case, the user field will be used to control the outgoing identity for the account you are adding, while the login will be used during the authentication phase.</para></listitem>
<listitem><para><guilabel>Password:</guilabel> You can enter your password.</para></listitem>
<listitem><para><guilabel>Registration Timeout:</guilabel> The timeout after which the registration should be updated.</para></listitem>
</itemizedlist></para>

</section>

</section>

<section id="ekiga-uris">
<title>Understanding URIs</title>

<section><title>SIP URIs</title>

<para>SIP URIs are formatted as such <literal>sip:user@[host[:port]]</literal></para>

<para>This permits you to call the given user or extension on the specified SIP proxy: <userinput>sip:jonita@ekiga.net</userinput></para>
</section>


<section><title>H.323 URIs</title>

<para>H.323 URIs are formatted as such <literal>h323:[user@][host[:port]]</literal></para>

<para>This permits you to: <itemizedlist>
<listitem><para>Call a given host on a port different from the default port which is 1720: <userinput>h323:seconix.com:1740</userinput></para></listitem>
<listitem><para>Call a given user using their respective alias if registered to a gatekeeper: <userinput>h323:jonita</userinput></para></listitem>
<listitem><para>Call a given phone number if you are registered to a gatekeeper for a PC-to-Phone provider, or if that user has an ENUM record associated to an H.323 URI: <userinput>h323:003210111222</userinput></para></listitem>
<listitem><para>Call a given user using their alias through a specific gateway or proxy: <userinput>h323:jonita@gateway.seconix.com</userinput></para></listitem>
<listitem><para>Call an MCU and join a specific room: <userinput>h323:myfriendsroom@mcu.seconix.com</userinput></para></listitem>
</itemizedlist></para>
</section>

</section>


<section id="ekiga-video-bandwidth">
<title>Controlling the Video Bandwidth</title>

<para><application>Ekiga</application> uses a best-effort algorithm to maintain a low bandwidth when transmitting video. You can adjust the video quality settings depending on whether you prefer to have a good frame rate, or a good picture quality. It will permit <application>Ekiga</application> to dynamically adjust the video bandwidth and the number of transmitted images per second during a call while trying to respect the requested video bandwidth.</para>

<para>Notice that the algorithm is a best-effort algorithm, which means that if your video bandwidth settings are too low, it can be impossible to respect them. However, if the video bandwidth permits to transmit with a better quality, or faster than the requested values, then <application>Ekiga</application> will dynamically increase them so that the quality and the framerate are always the best possible.</para>

<para>Choosing a higher framerate and a lower quality will have the same result in terms of video bandwidth as choosing a higher quality with a lower framerate. The choice depends on if you prefer using your bandwidth to transmit more lower-quality images or fewer high-quality images.</para>
</section>


<section id="ekiga-monitoring-lines">
<title>Monitoring lines</title>

<graphic fileref="figures/monitoring_lines.png"/>

<para><application>Ekiga</application> can connect to PBX systems supporting the SIP protocol. In that case, it is able to indicate if the line associated with a user is in use or not. Please refer to the documentation of your PBX to enable that feature.</para>

<para>To enable that feature on <application>Ekiga</application>, simply add the contact with their URI in the roster. If the server supports publishing presence information, <application>Ekiga</application> will automatically publish your own presence information and display the presence of contacts in your roster.</para>
</section>

<section id="ekiga-audio-codecs">
<title>Managing Codecs</title>

<section><title>Audio Codecs</title>

<graphic fileref="figures/audio_codecs.png"/>

<para>The <application>Ekiga</application> audio codecs table in the preferences permits you to change the codec order as well as disabling the codecs you don't want to use. Each codec has strong and weak points. For example, G.711 will give the best voice quality but will use the most bandwidth while SPEEX will give an average voice quality but requiring a very low bandwidth usage. Notice that there are two versions of SPEEX; one of them is SPEEX WideBand, which has a 16 kHz clock rate.</para>
</section>

<section><title>Video Codecs</title>

<graphic fileref="figures/video_codecs.png"/>

<para>The <application>Ekiga</application> video codecs table in the preferences permits you to change the codec order as well as disabling the codecs you don't want to use. <application>Ekiga</application> supports codecs like H.261, H.263+, H.264, MPEG-4 or Theora.</para>
</section>


<section><title>Reordering the codecs</title>
<para>When you reorder the codecs, you are reordering the local capabilities table, i.e. the codecs you will use for sending. You will always transmit audio and video using the first codec in the corresponding table that is in common with the remote user. The remote user will transmit audio and video using the first codec in his table that is common with you.</para>
</section>

<section><title>Forcing the use of a specific codec</title>
<para>You can force the use of a specific codec by disabling all other codecs, but it will result in failed calls if the remote user doesn't support that specific codec. The best approach is to put your preferred codecs at the top of the list so that you always transmit with them if the remote user allows it and to disable the codecs that you don't want to use for transmission and reception.</para>
</section>

<section><title>Adjusting the jitter buffer</title>
<para>You can adjust the delay to wait before playing the sound buffers that you have received using the jitter buffer adjustment. If there is too much packet loss, the delay required to have received all packets could be so important that it will exceed the jitter buffer. In such a case, the sound you are receiving will be of bad quality. A solution to that problem would be to increase the maximum limit of the jitter buffer to a few seconds, resulting in a big delay but in an improved voice quality. Notice that the jitter buffer will readapt itself to the lowest delay allowing for optimum transmission, and that a bad voice quality in reception is not due to a too low jitter buffer value, but to bad Internet connection quality.</para>
</section>

</section>


<section id="ekiga-defining-ports">
<title>Changing Ports</title>

<section><title>The listen ports</title>
<para>The main port used to listen for incoming connections in <application>Ekiga</application> for SIP is port 5060 (UDP), while 1720 (TCP) is used by H.323. To change those ports you need to load <command>gconf-editor</command>. Open <command>gconf-editor</command>, select <literal>apps</literal> from the left-hand side menu and then select <literal><application>Ekiga</application></literal>, <literal>Protocols</literal>. Then select <literal>sip</literal> or <literal>h323</literal>; it should give you a list in the corresponding window to your right. Select <literal>listen_port</literal> and change it to your desired value. You can also change the UDP/RTP port ranges.</para>
</section>

<section><title>Explanation of the port ranges</title>

<para>1. The <literal>listen_port</literal> value is the port <application>Ekiga</application> will use to listen for incoming connections. It is different for SIP and H.323.</para>

<para>2. The <literal>udp_port_range</literal> value is the range of UDP ports that <application>Ekiga</application> will use for SIP signalling or when registering to H.323 gatekeepers. It is also used for RTP (audio and video communication channels).</para>

<para>3. The <literal>tcp_port_range</literal> value is the range of TCP ports beside the <literal>listen_port</literal> that <application>Ekiga</application> will use for the H.245 channel with the H.323 protocol. That port range is not used by SIP. It is also not used when H.245 Tunnelling is enabled, which is generally the case, except when calling old H.323 implementations like <application>Netmeeting</application>.</para>
</section>
</section>


<section id="controlling-sip-h323-settings">
<title>Controlling the SIP and H.323 Settings</title>
<section id="ekiga-sip">
<title>Controlling the SIP Settings</title>

<section><title>Miscellaneous Settings</title>
<para><emphasis>Outbound Proxy</emphasis></para>
<para>The outbound proxy is the SIP proxy that will relay your calls. The behaviour of a SIP proxy is similar to the behaviour of an HTTP proxy, i.e. some entity that issues the requests on your behalf and proxies the streams.</para>

<para><emphasis>Forward URI</emphasis></para>
<para>The URI to which SIP incoming calls should be forwarded if configured in the preferences.</para>

</section>
</section>

<section id="ekiga-h323">
<title>Controlling the H.323 Settings</title>

<section><title>Miscellaneous Settings</title>
<para><emphasis>Forward URI</emphasis></para>
<para>The URI to which H.323 incoming calls should be forwarded if configured in the preferences.</para>

</section>

<section><title>Advanced Settings</title>
<para><application>Ekiga</application> permits a fine control of the H.323 settings in the <guilabel>Advanced H.323 Settings</guilabel> section of the preferences. You can enable H.245 Tunnelling, Early H.245 and Fast Start.</para>

<para><emphasis>H.245 Tunnelling</emphasis></para>

<para>H.245 Tunnelling is the encapsulation of H.245 messages within H.225/Q.931 messages (H.245 Tunnelling). If you have a firewall and enable H.245 Tunnelling, there is one fewer TCP port that you need to allow for incoming connections.</para>

<para><emphasis>Early H.245</emphasis></para>

<para>This enables H.245 early in the setup and permits achieving faster call initiation.</para>

<para><emphasis>Fast Start</emphasis></para>

<para>Fast Connect is a new method of call setup that bypasses some usual steps in order to make it faster. In addition to the speed improvement, Fast Connect allows the media channels to be operational before the <literal>CONNECT</literal> message is sent, which is a requirement for certain billing procedures. It was introduced in H.323 version 2.</para>
</section>

</section>
</section>
</section>

<section id="ekiga-about">
<title>About <application>Ekiga</application></title>
<para><application>Ekiga</application> is written by Damien Sandras (<email>dsandras@seconix.com</email>). To find more information about <application>Ekiga</application>, please visit the <ulink url="http://www.ekiga.org" type="http"><application>Ekiga</application> Home Page</ulink>.</para>

<para>To report a bug or make a suggestion regarding this application or this manual, follow the directions in <ulink url="ghelp:user-guide?feedback" type="help">this document</ulink>.</para>

<para>This program is distributed under the terms of the GNU General Public licence as published by the Free Software Foundation; either version 2 of the Licence, or (at your option) any later version. A copy of this licence can be found at this <ulink url="ghelp:gpl" type="help">link</ulink>, or in the file COPYING included with the source code of this program.</para>
</section>



<section id="ekiga-appendix">
<title>Appendix</title>

<section id="ekiga-related">
<title>Related Software</title>

<para><emphasis>IPBX</emphasis></para>
<itemizedlist>
<listitem><para>Asterisk PBX: <ulink url="http://asterisk.org" type="http">http://asterisk.org</ulink></para></listitem>
</itemizedlist>

<para><emphasis>SIP</emphasis></para>
<itemizedlist>
<listitem><para>SIP Express Router: <ulink url="http://www.iptel.org/ser" type="http">http://www.iptel.org/ser</ulink></para></listitem>
</itemizedlist>

<para><emphasis>H.323</emphasis></para>
<itemizedlist>
<listitem><para>OpenH323 Gatekeeper: <ulink url="http://www.openh323.org" type="http">http://www.openh323.org</ulink></para></listitem>
<listitem><para>GNU Gatekeeper: <ulink url="http://www.gnugk.org" type="http">http://www.gnugk.org</ulink></para></listitem>
<listitem><para>OpenH323 Proxy: <ulink url="http://openh323.sourceforge.net" type="http">http://openh323.sourceforge.net</ulink></para></listitem>
<listitem><para>H323 - ISDN Gateway: <ulink url="http://www.telos.de/linux/H323/" type="http">http://www.telos.de/linux/H323/</ulink></para></listitem>
</itemizedlist>

<para><emphasis>Conferencing/VoIP Software</emphasis></para>

<itemizedlist>
<listitem><para>OpenMCU: <ulink url="http://www.openh323.org" type="http">http://www.openh323.org</ulink></para></listitem>
</itemizedlist>

<para><emphasis>Similar Clients</emphasis></para>

<itemizedlist>
<listitem><para>XTen: <ulink url="http://www.xten.com" type="http">http://www.xten.com</ulink></para></listitem>
<listitem><para>SJPhone: <ulink url="http://www.sjlabs.com/" type="http">http://www.sjlabs.com/</ulink></para></listitem>
<listitem><para>OpenPhone: <ulink url="http://www.openh323.org" type="http">http://www.openh323.org</ulink></para></listitem>
<listitem><para>Netmeeting: <ulink url="http://www.microsoft.com" type="http">http://www.microsoft.com</ulink></para></listitem>
</itemizedlist>

</section>
</section>

</article>
<!-- END -->
