Wiki binary tree traversal
This is the Ekiga manual. It is updated continuously. If some information is not enough clear for you, or if you think you have found an error or an obsolete information, feel free to contact us. Ekiga is a free Voice over Internet Protocol or VoIP IP telephony and video conferencing application for Linux, Unix-like e. BSD or OpenSolarisand Windows operating systems. You can place audio and video calls from PC-to-PC or PC-to-phone. SMS-style messaging is also supported. It supports all major VoIP features like audio and video calling, call hold, call transfer, call forwarding, and instant messaging. It supports the best free video and audio codecs with limited echo cancellation for superior call quality. See features for a detailed list of features. Currently, Ekiga does not support: Ekiga has also been used in more exotic cases, such as sending fax over IP in French. Feel free to contact us to report a bug or make a suggestion regarding the application or this manual. The Session Initiation Protocol SIP is a protocol developed traversal 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 NovemberSIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture. It is one of the leading signalling protocols for Voice over IP. H 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 was the relatively early availability of a set of standards, not only defining the basic call model, but also the supplementary services needed to address business communication expectations. H was the first VoIP standard to adopt the IETF standard RTP to transport audio and tree over IP networks. H is based on the ISDN Q 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. Ekiga is compatible with any software, device or router supporting SIP or H It includes SwissVoice, CISCO, SNOM, Polycom, IP Phones, but also software like Windows Messenger, Netmeeting, SJPhone, Eyebeam, X-Lite, or also the Asterisk popular IPBXas well traversal any other commercial or Open Source IPBX. Ekiga is not compatible with Skype and will never be as long as their protocol stays proprietary. We do not think using closed protocols for communications is a good thing. For more information about software compatibility see Ekiga Interoperability. You need a microphone audio input into your computer. A headset is recommended to better prevent echo; however, a webcam with an internal microphone, or a separate microphone will also give good results. For binary External problems which might arise when using ekiga are bugs in libraries ekiga relies on, network itself for ex. Ekiga is free software as in both free speech and free beer and is licensed under the GNU General Public License GPL either version 2 of the License, or at your option any later version with the special exception that you have permission to link or otherwise combine this program with the programs OPALOpenH and PTLIBand distribute the combination, without applying the requirements of the GNU GPL to the OPAL and PTLIB programs, as long as you do follow the requirements of the GNU GPL for all the rest of the software thus combined. The mascot was drawn by Carlos Pardo. The mascot is named Lumi. When starting Ekiga for the first time, the Configuration Assistant will launch automatically. You should go through all of these steps properly, otherwise the assistant will re-appear when it has not been completed or Ekiga will not function correctly if some of your answers have been inaccurate. You may run the Configuration Assistant at any time from the Edit menu Throughout the entire configuration process, navigation is available at the bottom of the window. You will be able to navigate through the questions using the Back, Forward and Cancel buttons. If you hit Cancel during the setup, Ekiga will not be affected by your changes, and all entered information will be discarded. This page welcomes you to the Configuration Assistant. There is nothing to change or edit here. Press the Forward button at the bottom of the window to start traversal configuration The Personal Information window is where you enter your personal details needed to use Ekiga. You should provide both your first name and last name. This information will be displayed on your partner screen when you call him. If you want to call other users and to be callable yourself, you need a SIP address. Follow the link given in the dialog to register for an account if you do not already have oneand fill in your username and password. Then press Forward to continue Ekiga can be used with several Internet Telephony Service Providers. These providers will allow calls to real phones from your computer using Ekiga at competitive rates. There is no obligation for you to use a commercial provider, but if you need such service, we recommend using the default Ekiga provider. If you want to create an account and use it to call your friends and family using regular phones at competitive rates, click the "Get an Ekiga Call Out account" link. Once the account has been created, you will receive a login and a password by e-mail. Then press Forward to continue Ekiga 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, traversal more bandwidth it requires. Moreover, video codecs can adapt their quality to the available bandwidth. If your connection type is not mentioned in the list you should select the one closest to your network connection and adjust Ekiga manually with the preferences window codecs section later on. This setting will help Ekiga select the optimal codecs to suit your available bandwidth Audio device configuration is dependent on the operating system on which Ekiga is running, and different operating systems will configure audio devices in different ways. Ekiga will attempt to automatically detect audio hardware e. These audio devices will then be listed here, allowing you to select them for your calls. The audio ringing device allowing you to hear a ringtone for incoming calls is usually set to the internal sound card, under the name Default. The audio output device outputs the incoming sound stream during a call. You would typically select the device that your headphones or speakers are connected to. Default is a good choice for your internal sound card. The audio input device is where your microphone is connected to. Default is also a good choice for your internal sound card. It can be Video4Linux2 to manage webcams, or any other choice depending on the operating system on which Ekiga is running. This step is optional and is relevant for users with video devices e. If you do not have any video devices you may tree this page: Ekiga works fine for audio calls only Configuration of Ekiga is now complete. The last window displays a summary of the settings you have chosen. Verify that your settings are correct; if something is incorrect, you may use the Back button in the lower right hand corner of the window to go to any page of wiki assistant to correct the mistake. If all is well, press the Apply button to save the configuration. The assistant will close, and the main window of Ekiga will now appear It is generally recommended that you test your setup after having completed the Configuration Assistant. Click on the video camera icon on upper left of Ekiga window. A new window will appear: if you see your face then the camera works well with Ekiga. Otherwise make sure that your camera is working by testing it with another program which uses camera, and check Troubleshooting page. In this mode everything you say will be repeated back to you just as soon as is it received. The purpose tree this test is to traversal you an audible sense of the latency between you and the machine that is running the echo test application. You may end the test by hanging up or pressing the pound key". By speaking into your mic, you should hear your voice repeated back to you after a short delay. If the test was successful, you can continue to the next page of this manual. If the test was unsuccessful e. Note that the echo test accepts only some video and audio codecs, cf. You might also be interested by free fun numbers. An account to a VoIP service provider is not mandatory for using Ekiga you can use the IP address to call, or to use network neighbours, as shown later in the manualbut having at least wiki account will greatly enhance the Ekiga softphone experience. A SIP address is a way to be reachable and to reach people. You can compare it to an email address. This will give you a unique SIP address that you can give to your friends so that they can contact you. This window allows you to add and register with SIP, H and other types of accounts or VoIP services. An account contains the user login and password details to register with to the account or VoIP service. A dialog will appear and allows you to enter several parameters: Ekiga will do a best guess in determining the identity that will be used when calling out. Sometimes, you will need to force that identity. You can do this by specifying the identity in the User field. A dialog will appear and allow you to enter several parameters. The meaning of the parameters are similar to SIP account, see previous section. A dialog will appear and allow you to enter the connection parameters. For this to work, Ekiga should have been built with XMPP support enable-loudmouthwhich by default wiki is disabled. Usually, when calling someone, you type the SIP address which contains the registrar of one of your VoIP account, e. Ekiga will choose which account to use depending on the end of the SIP address if it is the same as one of the registrar you have in your account list to place the call. The default account is the one used when you type or dial a SIP address without the end of the SIP address what is on the right ofor the account used if you call a sip address which do not refer to any of your registrar in your account list. Wiki is your own address at this default account which will appear on the SIP client of your friend as the source of the call. Contacts can also be added from a tree menu by right-clicking in the roster, and selecting New contact. If you do not precise the host part, e. This is sometimes useful, as shown in the following. Like a long distance company, your VoIP provider can change often, and you can have more than one. For instance, I currently use one company for US calls, and ekiga dialout for all other countries. Furthermore, there may be traversal future global addressing schemes that could be used with multiple providers. For instance, at some point Skype will have to give in and at least allow gateways to SIP. Then the Skype address would be useful without a specific domain, in order to allow to chose among gateway providers. The contact is removed from your roster immediately. Contacts can also be removed from a contextual menu by right-clicking on the relevant contact in the roster. You can sort your contacts visible in the local roster into pre-defined groups, such as Friends or Work ; this allows for example to make visible only one of the groups e. Work when you are at work. You can also create custom group names to sort your contacts into, for example First Level Tech Support. To change the group a contact belongs to, first right-click on the contact as binary in the local roster, then click on Edit from the contextual menu that appears and tick the group you wish the contact to be sorted into. If you wish to add a custom group, enter your custom group name here, and click Add. Click OK to finish. Network neighbours is a special group in the roster, called Neighbours, which automatically discovers and shows all ekiga users from your LAN. It is updated each time a users connects or disconnets from your LAN, or changes his status or message. Neighbours group is visible, at the bottom of the roster, only if there is at least one ekiga user in your LAN. It is available only if ekiga has been built with support for it. This feature uses the zeroconf technology called sometimes Bonjour or avahi or rendezvous. To your left there will be a list dialog showing the Servers you have added to the list as well as a list of local Address Books. In the left-hand pane there will be a list of local address books, as well as any online LDAP address books. You can also place a cell phone call to users in the local address book, if the user information was originally entered using Novell Evolution. Note that you can only add contacts to local address books in Ekiga: adding contacts to LDAP address books from within Ekiga is not supported. The contact can also be removed by right-clicking on the contact, then selecting Remove from the contextual menu that appears. Note that you can only remove contacts from local address books in Ekiga: removing contacts from LDAP address books from within Ekiga is not supported Ekiga is able to connect to online address books, allowing you to search for contacts in a remote directory. The most common online address book type is the LDAP directory. Ekiga is able to search an LDAP directory and use a specific attribute as a calling address i. For example, you could connect to a an LDAP directory provided from within your own company, and use a specific attribute containing the local extensions of all your colleagues. You can use the Search Filter field to search for contact names and call addresses in an LDAP directory. A limited number of results corresponding to your search are returned. You can also right-click on the address book you wish to remove, and select Remove addressbook from the contextual menu that appears. You can use the Search Filter field to search for contact names and call addresses, and a limited number of results corresponding to your search are returned. You can also right-click on the contact, and select Add wiki local roster from the contextual menu that appears. You can set a presence status message in Ekiga that advertises your presence to your contacts. Ekiga features three preset status messages, and within each one of these presets, custom messages can be created. Simply click on the drop-down status message list to select a preset status. Note that you may be called in any of these statuses; they are used simply to inform the other users about your wish. Technical note: Ekiga binary SIMPLE PIDF Presence Information Data Format Ekiga v5 uses RPID Rich PID. To define a custom message, click on the drop-down status message list in the main Ekiga window, and select " Custom message. In the pop-up window that appears, enter your custom message. To delete a custom message, click on the drop-down status message list in the main Ekiga window, select Clearand select the custom message you wish to delete. When a call arrives, Ekiga rings and displays a notification window in the systray allowing you to accept or reject the call. If notifications are not available in your system which is the case for Windows systemsthen the call window is shown instead allowing you to accept or reject the call. Some systems have broken notifications, for example they do not show the whole text in case of long text, or they inform that they support notifications but they do not. In this case, when a call arrives, you have to manually open the call window and accept or reject the call from there. Wiki can be activated from the the REPLACE. These actions enable you to control active sessions. If you want to call other users and to be callable, you usually need a SIP address. Binary 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 use the online address book of Ekiga to find the SIP addresses of other Ekiga users. You can actually call any user using SIP software or traversal, and registered to any public SIP provider. Ekiga does not necessarily require a SIP account or STUN network detection in order to work. Ekiga allows you to call real phones. You simply have to register an account to traversal Internet Telephony Service Provider. You do not need any specific hardware to be able to do PC-to-Phone calls: a simple soundcard is enough. Using a headset to avoid echo problems is however highly recommended. Ekiga can be used with several Internet Telephony Service Providers. Those providers will allow calling real phones from your computer using Ekiga at interesting rates. We are recommending you to use the default Ekiga provider. The default provider is Diamondcard Worldwide Communication Servicewhich offers these rates. When signing up for a DiamondCard account, a small amount of the subscription is given back to the Ekiga project. If you want to create such an account and use it to call your friends and family using regular phones at interesting rates, go in the Edit menu, and select Configuration Assistant menu item. Afterwards, press Continue until the Ekiga Call Out Account Once the account has been created, you will receive a login and a password by e-mail. Simply enter them in the dialog, resume the Configuration Assistant until the last step, press Apply, and you are ready to call regular phones using Ekiga. To dial a number, simply add "00" followed by the country code, and by your number. For example, sip to call tree in Belgium. See List of PC to phone providers for a list of alternative providers. To allow this, you can simply login to your PC-To-Phone account using the Tools menu as described above, and buy a phone number in the country of your choice. Ekiga will ring when people will call that phone number. You can actually use any H or SIP ITSP provider, including your own PBX at work. However we recommend using the integrated provider. This effectively pauses video and audio transmission. Video and audio transmission will then resume. Audio will cease to be sent, but video transmission will continue. Video will cease to be sent, but audio transmission will continue. Starting with versionekiga can auto-answer calls. Note that tree security reasons Ekiga does not yet support call auto-answering when a specific SIP tag appears in the incoming packet. Call Forwarding can be configured through the preferences window. Notice that you need to specify an URI where to forward calls in the preferences to be able to activate that option. You will now see the appropriate section. It contains three checkboxes for the three cases described above. This is similar to a secretary who receives a call, and after some discussion transfers the call to the right person in the organisation. Note that Ekiga does not support to ask permission of the target address to accept the call transfer. The Call history window stores information date, duration, URI, Remote user about all outgoing and incoming calls. They are divided into three groups "Received calls", traversal calls" and "Unanswered calls". Received wiki contains all incoming binary which were accepted wiki Ekiga. Double-clicking on a row in the Calls History will call back the selected user or transfer any active call to that user The status bar visible at the bottom of the main Ekiga window shows information about audio and video bandwidth, and the video FPS frames per second achieved during an active call. To view network traffic statistics during an active call, you can hover the mouse pointer over the status bar to display a pop-up window Ekiga traversal you to send instant messages to your Binary contacts. Chat messages are not stored on any server, so in order for chat to work your partner must be connected when chatting. The chat window will appear and allow you to perform a text conversation with the selected remote user. To send the message simply type Return. To insert a new line inside the message, type Shift-Return. You can also open the chat window from a contextual menu by right-clicking on the user in your roster, and selecting Message. To do this, simply click on the new tab icon, and a new tab will automatically be created allowing a conversation with the user you are on a call with. To be checked: H text chat however only works from Ekiga to Ekiga. Ekiga allows to do some simple formatting of the text italic, bold, underline and use some emoticons, accessible from within an active chat window itself. Ekiga supports several audio and video codecs. It includes codecs with excellent quality, as well as codecs with medium to good quality. Generally, the higher the quality of a codec, the more wiki or CPU power it requires. The audio Ringing Device allowing you to hear a ringtone for incoming calls is usually set to the internal sound card, under the name Default. The audio Output device outputs the incoming sound stream during a call. The reason to have different output and ringing devices is that you might be interested to have headphones for output and external speakers for ringing, so that you hear incoming calls binary when you are far from the computer. The audio Tree device is where your microphone is connected to. Each codec has strong and weak points. Note that there are two versions of Speexone with 16kHz and another one with 8kHz, the former is more accurate but consumes more network bandwidth. When you reorder codecs, you are reordering the local capabilities table. During a communication, only one audio codec is used, for both directions. The audio codec used is the first active codec on receiver which is active on sender. You can force the use of a specific codec by selecting it and disabling all other codecs, but this will result in failed calls if the remote user has not enabled that specific codec. The best approach is to put your preferred codecs at the top of the list, and to disable the codecs that you do not want to use for sending and receiving audio. You can the maximum delay to wait before playing the sound buffers that you have received using the jitter buffer tree. If there is too much packets 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 binary receiving will be of bad quality. A solution to that problem would be to increase the maximum limit of the jitter buffer to say one second, resulting in a big delay but an improved voice quality. In Ekiga v5, setting maximum buffer size was removed. To tree up, Ekiga uses a buffer for incoming packets jitter buffer with dynamic size. If packet jitter is small, jitter buffer is made small too, and audio packets will play soon after their reception. If packet jitter is big, jitter buffer is automatically increased, and audio packets will play late after their reception, without however exceeding a predefined value ms as of Ekiga version Echo cancellation will only work if you use the same sound device for in and out audio, e. Echo cancellation is based on Speex software. Ekiga will attempt to automatically detect video hardware plugged into your computer. These video devices will then be listed here, allowing you to select them for your calls. You can also adjust other settings for the selected video input device, such as the size of the transmitted video display, the regional video format, and the video source channel to use. Then, click on the camera icon to display a real-time local feed from your video device. Finally, click on the contrast icon to the left of the camera icon, and use the sliders to adjust the brightness, whiteness, colour, and contrast settings for your video feed The video codecs table in the Ekiga preferences window allows you to enable, disable, and reorder video codecs. For example, h is an older codec available on the wiki variety of video conferencing systems, but has average quality. During a communication, only one video codec is used. The video codec used is the first active codec on receiver which is active on sender. Tip: Video codecs can adapt their quality to the available bandwidth. This setting is made when the Configuration Assistant is run during initial setup of Ekiga, tree that an optimal quality codec is selected based on your network connection. You can test your webcam by clicking on the video camera icon on upper left of Ekiga window and check that it shows anything. See test your camera section in this manual for more information. Ekiga uses a best-effort algorithm to maintain low bandwidth usage when transmitting video. You may if you like adjust video quality settings favouring a good frame rate, or, a good picture quality. Ekiga will dynamically adjust binary video bandwidth and the number of transmitted images per second during a call, while trying to respect the requested video bandwidth. A best-effort algorithm means that it may be impossible for Ekiga to respect a bandwidth setting if it is set too low. However, if the video bandwidth allows transmission with a higher picture quality or a faster framerate than the set value, then Ekiga will dynamically adjust this so that the quality and the framerate are always the traversal possible. Choosing a higher framerate and a lower picture quality will have the same result in terms of video bandwidth as choosing higher picture quality with a lower framerate. It depends if you prefer using your bandwidth to transmit more lower quality images, or fewer better quality images. In the event that your video input device is unavailable or if you prefer not to send video, you can instead send a moving logo animation. This can be selected from the Ekiga preferences as follows The outbound proxy is the SIP proxy that will relay your calls. The behavior of a SIP proxy is similar wiki the behavior of an HTTP proxy, ie some entity that issues the requests on your behalve and proxies the streams. Ekiga permits a fine control of the H settings in the Advanced H Settings section of the preferences. H tunneling is the encapsulation of H messages within H Q messages H tunneling. If you have a firewall and enable H tunneling, there is one less TCP port that you need to allow for incoming connections. 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 CONNECT message is sent, which is a requirement for certain billing procedures. It was introduced in H version 2. Ekiga has initial support for H handling multiple video streams, such as video and presentation slides. In ekiga except for Windows, the video stream to be shown can also be chosen during the communication, through the View menu. Ekiga adds support for this on Windows too. In ekiga the support is complete as receiver. This means that you can see both video streams simultaneously, in two windows. The extended stream window appears when a frame of the extended video is processed, and it is hidden when the communication ends. There is no support as sender for a second video stream. If you are behind a router, Ekiga has advanced NAT support thanks to STUN. You need first to find out the precise error, i. T then your NAT type is symmetric NAT. For H you will have to manually tree ports to to your internal machine. If you want to be reachable from the outside, you will wiki to forward TCP port from your router to the internal machine running Ekiga. If you want to be able to do calls with Netmeeting users, you will need to forward TCP ports to from the NAT router to the machine running Ekiga. This last step is only necessary when calling H software that does not support H Tunnelling the H part of Ekiga supports H Tunnelling, forwarding that TCP port range is thus not required when calling them using H The default ports are defined by the standards SIP, H and STUN. You should tree change those ports except for complex configurations or rare cases. The default ports used by Ekiga are the following: To change those ports, open gconf-editorselect apps from the left hand binary menu and then select Ekiga. Then select "sip" or "h323", it should give you a list in the corresponding window to your right. However, this port cannot be changed. To understand the problems with symmetric NAT, read technical details. This type of NAT cannot be served by a VoIP program without a "trick". Skype for example works in this case too because it uses the other users of traversal as proxies, otherwise said the other binary give are forced to give resources CPU and network bandwidth for Symmetric NAT skipe users. Another solution is to have a public proxy for all the symmetric NAT users. This can work for audio traffic, but for the much higher bandwidth consuming video traffic this is overkill. Finally, the simplest solution, if it works, is to change the NAT from Symmetric to another type. For that, if Ekiga shows a dialog box about that on startup and you have access to your router configuration, traversal may try the following. Typical SOHO routers incorporate a Network Address Translator NAT : local network addresses are hidden behind translated into a single address presented to the WAN, which in the domestic case is normally that provided by an ISP. Normally incoming packets those from the WAN are silently dropped unless they match a previous outgoing packet. Apart from reasons of security, the router has no means of determining the destination address on the LAN if there is no match with a previous outward bound packet. Such routers normally have facilities for port forwarding : incoming packets for distinguished ports can be directed to a particular machine on the LAN wiki the NAT. The limitation is that only the choosen machine can engage in voip calls. The STUN protocol rfc is designed to solve this problem, but it does not work with symmetric NATs: those that are strict about incoming packets matching precisely a previous outgoing packet. Many NAT routers allow Application Rules to be defined which allow specified ports to be opened dynamically to incoming packets when triggered by an outgoing packet. If this trigger is fired when ekiga sends a packet to a STUN server, ports can be opened so that the STUN protocol sees essentially an open NAT: a full CONE. Actually, the trigger rules below just open a restricted range, so this is more accurately a "Port restricted NAT" which is how ekiga may describe the situation once the rules are in place. This rule is shown below on a DLink router It is only necessary to trigger ondespite the previous example. Troubleshooting is covered in more detail in the Troubleshooting Section in the wiki. These are the error message Ekiga presents. OPAL sends a numeric code to Ekiga and Ekiga presents these phrases to the user. Ekiga has ENUM support. See binary details for technical details on how ENUM works. If you call a SIP URL without a region i. You have to start the phone number with binary country code, i. So you can enter sip to reach the Austrian ENUM test number. Some technical people might prefer OSS4 audio device over ALSA or Pulse. This appendix presents how to configure Ekiga to use it. Some distros such as Arch Linux provide packages in the main repository to install OSS 4. If your distro does not provide an OSS4 package, you may build OSS4 from source. As such, it is not available anymore starting with Ekiga The connection type page of the assistant is a bit confusing and error-prone. As such, it is not available anymore starting with Ekiga The audio devices page of the assistant is not really useful, since by default Ekiga selects the right device. As such, it is not available anymore starting with Ekiga The video device page of the assistant is not really useful, since by default Ekiga selects the right device. As such, it is not available anymore starting with Ekiga The jitter buffer will adapt itself to the lowest delay allowing for optimum transmission. A bad voice quality in reception is not due to a too high jitter buffer value, but to bad internet connection quality The proxy tree currently used by all accounts. See feature request outbound SIP proxy setting should be attached to accounts.
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