# &app; Manual &manrevision;

## Introduction

&app; is a free audio/video-conferencing application for Linux and other Unices (e.g BSD or MacOSX). It was written by Damien Sandras and is licensed under the GNU/GPL. It is based on the H.323 ITU standard and allows you to make audio and video calls to remote users with H.323 hardware or software (such as Microsoft Netmeeting). It also supports all modern videoconferencing features, such as registering to an Internet Locator Service (ILS) directory, gatekeeper support, making multi-user conference calls using an external Multipoint Control Unit (MCU), using modern Quicknet telephony cards and making PC-To-Phone calls.

## Getting Started

When starting &app; 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 &app;. You should go through all of these steps properly, otherwise the assistant will re-appear (when it has not been completed) or Gnomemeeting will not function appropriately (if some of your answers have not been correct). You may run the Configuration Assistant at any time from the Edit menu.

### Tip

All settings can be changed via the preferences window at anytime.

Configuration Assistant Introduction

Throughout the entire configuration process navigation is available at the bottom of the window. You will be able to navigate through the questions using Back, Forward and Cancel. If you hit Cancel during the setup Gnomemeeting 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 towards the bottom of the window to start the configuration.

Personal information

The Personal Information window requires you to supply personal information for GM usage. This information is necessary for basic functionality when connecting to other audio/video conferencing applications, as well as additional services such as user directories or gatekeepers. As user directories do require a First Name, Surname and Email-Address, you must enter them here, before proceeding.

Callto URL

The Callto URL; Enter an email address for a callto://address (eg: callto:ils.seconix.com/user@domain.com). If you leave the checkbox unchecked, your personal information will be published on the user directory of online Gnomemeeting users and you will be visible to other Gnomemeeting users. If checked then you will not be registered and other users will have to know your IP address to contact you. Please press 'Forward' after having entered all required information to continue.

Connection Type

Gnomemeeting 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. Thus, not every codec will work properly with a very slow network connection. This option is necessary in the initial configuration of Gnomemeeting so that it chooses the optimal codec suited for 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 connect and adjust Gnomemeeting manually with the preferences window (codecs section) later on.

NAT Type

The NAT Type detection page will allow you to detect which type of NAT you are using (if any) and help configure GnomeMeeting appropriately. Clicking on the detection button will bring a popup indicating which type of NAT was detected. This will help to describe the most appropriate way to configure GnomeMeeting allowing it to operate best with your NAT configuration. No configuration should be required in most cases, however some NAT routers will require the administrator to forward a few TCP and UDP ports to the internal computer running GnomeMeeting. Please refer to the GnomeMeeting FAQ for configuration examples.

Audio Manager

The Audio manager manages everything audio. It can be ALSA, OSS or even Quicknet to manage a Quicknet card. The recommended Audio Manager is ALSA.

Audio Devices

Gnomemeeting requires audio devices to play and record sound. The audio output device ouputs the incoming sound stream during a call. Please select the device that your headset or speakers are connected to. The audio input device is where 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. It is generally recommended that you test your settings after having selected all the appropriate devices. Please press the 'Test those settings' button on the right. If this test was successful you can continue on to the next page in the Configuration Assistant. Otherwise you should change your devices and test your configuration again until you have a setup that works for you.

Video Manager

Please select the Video Manager from the list. It can be Video4Linux to manage webcams, or AVC / DC for Firewire cameras.

Video Devices

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. If you have a video device but it's not showing up in the selection list or you want help in setting up your video device you may refer to the following links.

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

Please hit the "Test those settings" button to ensure that your device works with &app; if so, continue on with the Configuration.

Configuration Complete

The configuration of Gnomemeeting is now completed. 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 'Back' button in the lower right hand corner of the window to move to any page of the assistant and correct the mistake.

If everything is correct please press the 'Apply' button to save the configuration. The assistant will be closed and the Main Window of Gnomemeeting will now appear. Remember, all settings can be changed via the preferences window at anytime.

## Basic Usage

### To Call Other Users

Calling other users via Gnomemeeting is available in different forms and is done primarily through the use of URL's. &app; supports two different kinds of URLs: h323:// and callto://. Callto URL's were used by Microsoft for Netmeeting, and H.323 URLs respect the H.323 standard and are thus the default for &app;.

H.323 URL's

H.323 URL's are formatted as such "h323:[user@][host[:port]]"

This permits you to:

- Call a given host on a port different from the default port which is 1720: h323:seconix.com:1740

- Call a given user using their respective alias if registered to a gatekeeper: h323:jonita

- 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 URL: h323:003210111222

- Call a given user using their alias through a specific gateway or proxy: h323:jonita@gateway.seconix.com

- Call an MCU and join a specific room: h323:myfriendsroom@mcu.seconix.com

CALLTO URL's

Callto URL's are formatted as such "callto:[user@][host[:port]]"

Callto URL's and H.323 URL's are formatted exactly the same except however callto urls also support ILS lookups through callto URLS of the type: callto:ils_server/user_mail.

For example, calling callto:ils.seconix.com/joe.user@somedomain.com will look for the user with the joe.user@somedomain.com email address on the ILS server ils.seconix.com and proceed to initate a call.

Callto URL's also support more advanced parameters that are also supported by &app;, all parameters are defined in the Microsoft documentation and should be seperated by a '+' in the URL.

There are for example five different ways using &app; to call the user joe on the gateway gw.seconix.com:

- callto:joe@gw.seconix.com

- callto:joe+gateway=gw.seconix.com

- h323:joe@gw.seconix.com

- callto:joe if you specified gw.seconix.com as gateway in the &app; preferences

- h323:joe if you specified gw.seconix.com as gateway in the &app; preferences

Speed dials

&app; is able to associate speed dials with URLs using the address book. You can thus for example associate the speed dial 1 to the URL callto:ils.seconix.com/jonita@jxys.net. That speed dial can then be used as an H.323 or as a callto URL provided that it ends with #. For example, calling callto:1# will call callto:ils.seconix.com/jonita@jxys.net provided that both are associated together in the address book. Quicknet cards users will just have to dial 1 then hit the # key on the phone keypad to trigger the call to the associated URL.

Calling URLs

If you know the URL address of the party that you wish to call, you may enter that URL into the h323: or callto: input box at the top of the screen and press the Connect button; (which looks like an unplugged power socket and is conviently located to the right of the input box). eg: h323:192.168.100.1 and pressing the Connect button would call the user at that respective IP. When the call is connected and you have started communication with your party, the Connect button will now resemble a power plug connected to a socket.

#### Tip

If you use literal IPv6-addresses, square brackets are required around them so that the last part of the address won't be mistaken for a port: h323:[2001:638:500:200:210:5aff:fe4c:cfd1] or h323:[2001:638:500:200:210:5aff:fe4c:cfd1]:1740

Calling normal phones using PC-to-Phone

When you are doing PC-to-Phone calls, you are generally registered to a gatekeeper. It is thus enough to call the remote user using his phone number.

Calling users from the online address book

The online address book will permit you to call online users registered to an ILS server. You can bookmark their callto URL using drag-and-drop or the appropriate menus. See Managing contacts with the Address Book for further information.

Calling users from the offline address book

The offline address book permits you to bookmark your favorite users with their h323 or callto URLs.

Calling users from the calls history

The calls history remembers the callto and h323 URLs that have been used when giving or receiving calls. You can call any user again that is already present in your calls history by double-clicking on his entry in the history.

### To be Reached by Other Users

There are several ways to get a universally reachable address that other clients can use to call you. The most classical ones will be described here.

Using an ILS directory

If you are registering to an ILS directory like ils.seconix.com, then other users will be able to call you using an URL such as callto:ils.seconix.com/myself@example.com whatever your IP address is.

Using a dynamic hostname

You can also register to a service providing DNS services for dynamic IP's. That way, your machine will always be associated to the registered hostname. This way other users will not need to know your IP address to be able to call you.

Using a gatekeeper

If your company is running an H.323 gatekeeper on a public IP, it can be configured to accept calls and redirect them to the correct GnomeMeeting client based on the called URL. For example, if a gatekeeper is installed on example.com and user joe is registered to that gatekeeper, then other users will be able to call joe using the following URL: h323:joe@example.com.

Using an ENUM record

You can register an ENUM record on www.e164.org. That will permit you to associate your real phone number to any valid H.323 URL, and other users will be able to call you using your real phone number. For example, +3210444555 will be directly linked to h323:myhostname.com and other users will be able to call you using h323:3210444555.

### To Manage Calls

Understanding the statistics

To view the statistics, please select the Statistics tab in the control panel.

The statistic visualizes the network traffic caused by &app;. It draws a graph for each RTP stream. This means that - if audio and video are enabled in &app; and the client of the remote party - you will see four different graphs. (incoming audio stream, incoming video stream, outgoing audio stream, outgoing video stream)

Lost packets: The percentage of lost packets, ie of packets from the remote user that you did not receive. A too high packets loss during the reception can result in voice and/or video distortion and is usually caused by a bad network provider or by settings requiring much bandwidth.

Late packets: The percentage of late packets, ie of packets from the remote user that you received but too late to be taken into account, &app; being sending and receiving real-time video and audio.

Round-trip delay: The required time for a packet to arrive at its destination and come back. You can see the Round-Trip delay during a call as a connection quality indicator together with the Lost and Late packets statistics.

Jitter buffer: The Jitter buffer is the buffer where received sound packets are accumulated. When the buffer is full, then the sound is played. If your network is of bad quality, then you need a big jitter buffer, ie a big delay before sound is played back, because you need more time before being able to play audio back.

Adjusting the audio and video settings

Your audio and video settings can be adjusted through the control panel while you are in a call. If you want to change the audio input or output devices during a call, simply select the Audio tab in the panel. The brightness, whiteness, color and contrast of your video input device are changed via the Video tab.

Using text chat

Besides the communication through live audio and video streams &app; also provides a text chat interface. This panel is opened via the text chat button in the control bar of the main window. Enter your message into the text entry box under the chat window and press enter to send it to the remote party.

#### Tip

Text chat works with &app; to &app; clients only.

Controlling the call

&app; supports several actions which can be performed when in a call. These actions enable you to control active sessions.

Ending a call: The communication to the remote user can be ended by selecting Call->Disconnect.

Holding a call: You can hold a remote party call by selecting Call->Hold. This effectively pauses Video and Audio transmission, to continue transmission again you select Call->Retrieve Call and Video and Audio Transmission will begin again.

Mute Audio: This effectively prevents all Audio communication to your respective party.

Suspend Video: This effectively prevents all Video transmission to your respective party.

Transferring the remote party: You can transfer the remote user to another H.323 or CALLTO URL by using the appropriate menu entry in the Call menu or by double-clicking on an user in your address book, or in the calls history.

#### Tip

All URLs supported by &app; (H.323, CALLTO and Speed Dials) can be used for call transfer.

Taking a snapshot

While in a call you can take a snapshot of the remote party via Call -> Save Current Picture. A PNG-file will be saved in the current directory. The filename consists of three parts: the save_prefix, date and current time. (e.g. gnomemeeting-snap-2003_06_19-024316.png).

Watching calls execution using the history windows

History windows in &app; are comparable to logfiles. They keep chronological track of actions performed by &app; and provide additional information to the user.

General History

The General History window keeps track of many operations which are mainly performed in the background. It displays information about audio and video devices, connections to ILS directories and gatkeepers, codecs and other details. The latest operations can be found at the bottom, older entries are shown on the top. You can access this information by opening Tools->Generic History.

Calls History

The Calls History window stores information (date, duration, URL, Software, Remote user) about all outgoing and incoming calls. They are divided into three groups - Received calls, Given calls and Unanswered calls. The first group, Received call, contains all incoming calls which were accepted by &app;. Given calls keeps track of all attempts - succesful or not - to call another user. The last group, Unanswered calls, shows incoming calls which timed out or were rejected (if Do Not Disturb is enabled, for instance) by &app;.

#### Tip

Double-clicking on a row in the Calls History will call the selected URL or transfer any active call to that URL. Notice that you can also drag and drop entries from the Calls History into the address book.

This information can be accessed by opening Tools->Calls History and by switching between the three tabs.

### To Register to ILS Directories

Registering to ILS directories

This section describes what ILS directories are and how you can register to them.

What is an ILS directory?

The Internet Locator Service (ILS) is a directory similar to a telephone directory. Think of it as a phone book, when you want to call someone you just look up the number for that particular person and dial away. The ILS directory operates on the same concept except in this case the number you're looking up is an IP address.

The default ILS directory for Gnomemeeting users is ils.seconix.com, you can use any ILS directory of your choice.

Registering to an ILS directory

&app; allows you to register to one ILS directory at a time. All settings can be configured with the preferences window. You can register to an ILS server with the following steps:

* Start Gnomemeeting

* Select Edit -> Preferences and choose "Directory Settings" (located in the General subcategory) to your left.

* Enter in the 'User Directory' input box the hostname or IP address of the ILS server you wish to register with.

* Ensure that the 'Enable Registering' check box is selected.

* You may also choose to allow other registered users to view your details (as defined in the Personal Data settings) in the user directory. If you do not select the last check box you will be invisible from other users but your callto:// address will still be functional.

Browsing an ILS directory

The ILS directory can be browsed by selecting the menu option Tools -> Address Book or by selecting the ILS address book icon from the main window. To view the updated contents of the directory click on the 'Find' box.

Searching for users

The address book can be searched on the first name, last name and email address fields. Select the field you wish to search, enter the search query and press enter.

### To Manage Contacts

Managing my contacts with the Address Book

The Address Book is a feature which allows you to find users to call and/or to save your list of persons that you call on a regular basis in a 'Group'. It loads the list of users from the ILS directory and will store locally CALLTO and H.323 URLS and associated speed dials as well as addresses of ILS directories.

Basics of the Address Book

To open the Address Book select Tools -> Address Book and the GnomeMeeting Addressbook window should appear. To your left there will be a list dialog showing the Servers you have added to the list as well as a list of Groups. The defaults are ils.seconix.com for Servers and the Personal address book.

GnomeMeeting is able to use several types of address books. The most common address book type is the ILS directory where you can find information about registered users. GnomeMeeting is also able to browse real LDAP directories and use a specific attribute as calling URL. For example, you could have a gatekeeper in your company where all users are registering using their email address. All users would also be listed in the company's LDAP server, and a double-click on a specific user would trigger a call through the gatekeeper using the selected user's e-mail address. GnomeMeeting is also able to detect other GnomeMeeting users on the LAN using Rendez-Vous technology popularized by Apple (tm). That supposes you have a local mDNSResponder daemon running on your computer. Finally, GnomeMeeting is able to bookmark contacts in the local address book, shared with the Evolution suite.

To receive a list of users on a certain ILS or LDAP server, right click on the server in question and select 'Refresh'. This will contact the server, eg: ils.seconix.com and fetch its user list. It will then sort the user list by callto:// URL and you can now begin to contact people by double clicking on their highlighted field. You can also Drag-N-Drop to call a specific party by selecting the highlighted field and dragging it into the Main Window.

If the persons name is in the color 'gray' it means that this person is in Busy mode or currently already involved in a Gnomemeeting call with another user, the Status field 'S' will show a phone that is off the hook.

In certain cases you'll want to search specifically for a persons name or email on a ILS directory. For instance an ILS directory might have a large amount of people and to scroll through each and every single name looking for the one you wish to call would be time consuming. With the Addressbook you can do this by selecting from the drop down list the persons First Name, Last Name or What their Email Contains and entering the appropriate search item into the input box then select 'Refresh'.

Managing servers, groups and contacts

To add a server, highlight 'Servers' and right click. The option of adding a New server will appear. You then click on 'New Server' and at this point a dialog will appear asking you to enter the server name. Enter the name and select 'OK' and the new server should now appear in the Servers list.

To add a group, highlight 'Groups' and right click. The option of adding a New group will appear. You then click on 'New group' and at this point a dialog will appear asking you to enter the group name. Enter the name and select 'OK' and the new group should now appear in the Groups list.

To add a contact to one of your Groups you would select the Group you wish to add the contact to and right click. The option of adding a New Contact will appear and you may now enter the Contacts name and CALLTO or H.323 URL. After complete select 'OK' and now your contact is added to the list in that group. You can also add a contact from the ILS directory list by selecting the highlighted Contact and dragging them to the specific group you wish to add them to or by using the right-click menu when selecting that contact. Finally, you can edit the groups your users belong to using the User Properties dialog from the main menu or from the right-click menu, or using drag-and-drop between groups.

### To Manage Incoming Calls

Managing incoming calls

&app; supports different policies for incoming calls. Per default it displays a popup window which allows you to decide whether you want to refuse or accept the request for an incoming call. Furthermore Gnomemeeting offers three additional behaviors: Busy mode, Free for Chat and Forward.

Busy mode

If this mode is enabled Gnomemeeting refuses all incoming requests and only allows outgoing calls. You are not able to receive any call and do not notice if another user tries to contact you except when looking at the Calls History.

This mode can be enabled by selecting Call -> Busy in the main window.

Free for Chat mode

If this behavior is activated Gnomemeeting accepts all incoming calls. It does not display a popup window but tries to establish the connection to the remote party immediately.

This mode can be enabled by selecting Call -> Free For Chat in the main window menu.

Forward

Gnomemeeting has the ability to forward calls (H.323 Call Forwarding) to another host. Which allows you to configure Gnomemeeting to forward all incoming calls to a specified URL. Furthermore it is able to forward calls interactively when you do not answer the call after a configurable amount of time or when you are busy.

Call Forwarding can be configured by selecting Call -> Forward in the main menu or through the preferences window. Notice that you need to specify an URL where to forward calls in the preferences to be able to activate tht option. Open the preferences window by choosing Edit -> Preferences in the main window and select Call Forwarding on the left. You will now see the appropriate section. It contains three checkboxes for the three cases described above and one textfield for the IP address/hostname of the host the calls shall be forwarded to.

## Advanced Usage

### To Register to Gatekeepers

Registering to Gatekeepers

This section describes what gatekeepers are and how you can register to them.

What is a gatekeeper?

A H.323 gatekeeper controls all H.323 clients (e.g. &app; or NetMeeting) in its zone. Although gatekeepers have a large range of different features, one of their most important functions is to perform address translation between symbolic alias addresses and IP addresses. This means that you can use aliases instead of IP addresses to call another user. For instance, if you want to call the user "Damien" but you do not know his current IP address (dynamic IP addresses) you can easily use his alias at the gatekeeper. The gatekeeper will now direct your call to the correct IP address.

There are many reasons for users to register to a gatekeeper. An alias at a gatekeeper requires a registration, for example. Some gatekeepers do not allow unregistered users access to any of their features, in such a case you may not be able to call any user with an alias via the gatekeeper.

Registering to gatekeepers

&app; allows you to register to one gatekeeper. All settings can be configured with the preferences window. Please select Edit-> Preferences and choose Gatekeeper Settings (located in the H.323 Settings subcategory) on the left. The display window contains all information necessary for a correct registration to a gatekeeper.

&app; provides three different methods to register to a gatekeeper. Available values for the Registration method are as follows:

- Gatekeeper host - This method should be chosen when you know the hostname or IP address of the gatekeeper. &app; will directly connect to the gatekeeper.

- Gatekeeper ID - This method should be used when you have a valid ID at a gatekeeper but you do not its hostname or IP address. In this case &app; broadcasts a message and registers to the gatekeeper which answers to your ID.

- Automatically discover - If you are not sure which method to choose you can select this method. But please be aware that this method does not always work and is not recommended.

After having chosen the Registration method there are still four further fields - Gatekeeper ID, Gatekeeper host, Gatekeeper alias and Gatekeeper password. The four fields should contain the following information:

- Gatekeeper ID - The Gatekeeper identifier to register with. This field is required when you have chosen Gatekeeper ID as the Registration method.

- Gatekeeper host - Hostname or IP address of the gatekeeper. This field is required when you have chosen Gatekeeper host as the Registration method.

- Gatekeeper alias - Gatekeeper alias to use when registering.

- Gatekeeper password - Password to use for H.235 authentication when registering to the gatekeeper.

Gnomemeeting will automatically register/unregister to the gatekeeper when it is started/closed.

### To Do Calls Using a Gateway/Proxy

What is a gateway/proxy?

An H.323 gateway/proxy is software or hardware that will route and/or proxy calls and translate them from one protocol to another. A gateway translating from H.323 on IP to ISDN is an example and the Open Source PBX Asterisk is another example of gateway.

Doing calls using a gateway

Doing calls using a gateway is very simple. You can fill in the gateway hostname or IP address in the Gateway/Proxy section of the &app; preferences, and then simply use classical H.323 or CALLTO URL's to call users using your gateway or proxy.

#### Tip

You can not use CALLTO URLs looking on an ILS server when using a gateway or proxy. Notice also that it might be conflicting with your Gatekeeper following your Gatekeeper configuration.

### To Use a Quicknet Card

A Quicknet card is hardware that can be used to replace your soundcard. It has several features that are not available to normal soundcard users like echo cancellation, G.723.1 audio codec support, POTS - Plain Old Telephone System - management.

When using a Quicknet card, you can use either an old phone plugged into the card or an headset. You will need to select Quicknet as plugin in the Audio Devices section of the preferences, select the required Audio Input Device and then indicate if you are using a POTS or a headset with your Quicknet card. If you are using a phone, simply plug its cable into the Quicknet card and you are ready to do &app;-managed calls with your phone.

#### Tip

You can use the '*' sign of the keypad of your phone to replace the '.' if you want to dial IPv4 addresses. Notice also that if you want to dial complex URLs, you can associate them to speed dials in the address book, and simply dial the given speed dial followed by '#' to immediately start calling that URL.

### To Control the Video Bandwidth

&app; is using a best-effort algorithm to maintain a low bandwidth when transmitting video. You can adjust the video quality between 1 and 100%, and the number of transmitted images per second in the Video Codecs section of the preferences. It will permit &app; 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.

Notice that the algorithm is a best-effort algorithm, which means that if you are giving too high values for the minimum video quality and frames per seconds to transmit, it will be impossible to respect the requested video bandwidth. However, if the video bandwidth permits to transmit with a better quality, or faster than the requested values, then &app; will dynamically increase them so that the quality and the framerate are always the best possible.

Choosing a higher framerate and a lower quality will have the same result in terms of video bandwidth than choosing a higher quality with a lower framerate. It depends if you prefer using your bandwidth to transmit more lower quality images or fewer big quality images.

### To Manage Codecs

Audio Codecs

The &app; audio codecs table in the preferences permits you to change the codecs 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-8k will give an average voice quality but requiring a very low bandwidth usage.

Reordering the codecs

When you reorder the codecs, you are reordering the local capabilities table, ie the codecs you will use for sending. You will always transmit audio using the first codec in the table that is in common with the remote user. The remote user will transmit audio using the first codec in his table that is common with you.

Forcing the use of a specific codec

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 allow that specific codec. The best is to put your prefered 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.

Adjusting the delay

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 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 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.

### To Define Ports

Setting &app; up to use Different Ports

The main port listening for incoming connections in &app; is port 1720, to change this port you need to load "gconf-editor". Open gconf-editor, select apps from the left hand side menu and then select gnomemeeting. Then select "ports" it should give you a list in the corresponding window to your right. Select listen_port and change it to your desired value. You can also change the UDP/RTP port range, The TCP port range and the UDP port range used to register to a gatekeeper. These are described in detail below.

Explanation of Port Ranges

1. The "listen_port" value is the port &app; will listen for incoming connections on.

2. The "rtp_port_range" value is the range of UDP ports that &app; will use for RTP (audio and video communication channels). You can change the UDP port range if you are using IP translation and if you need to have several &app; running behind your NAT/PAT router. &app; needs to be restarted for the new values to take effect.

3. The "tcp_port_range" value is the range of TCP ports beside the listen_port that &app; will use for the H.245 channel. You can change the TCP port range if you are using IP translation and if you need to have several &app; running behind your NAT/PAT router. &app; needs to be restarted for the new values to take effect. This port range has no effect if both participants to the conference are using H.245 Tunneling.

4. The "udp_port_range" value is the range of UDP ports that &app; will use when registering to gatekeepers. You can change the UDP port range if you are using IP translation and if you need to have several &app; running behind your NAT/PAT router. &app; needs to be restarted for the new values to take effect. This port range has no effect if you are not registering to a gatekeeper.

### To Control the H.323 Advanced Settings

&app; permits a fine control of the H.323 settings in the Advanced H.323 Settings section of the preferences. You can enable H.245 Tunneling, Early H.245 and Fast Start.

H.245 Tunneling

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

Early H.245

This enables H.245 early in the setup and permits to achieve faster call initiation.

Fast Start

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.323 version 2.

## About &app;

&app; is written by Damien Sandras (dsandras@seconix.com). To find more information about &app;, please visit the &app; Home Page.

To report a bug or make a suggestion regarding this application or this manual, follow the directions in this document.

This program is distributed under the terms of the GNU General Public license as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. A copy of this license can be found at this link, or in the file COPYING included with the source code of this program.

## Appendix

### Related Software

Gatekeepers, Gateways and Proxies

- OpenH323 Gatekeeper: http://www.openh323.org

- GNU Gatekeeper: http://www.gnugk.org

- OpenH323 Proxy: http://openh323.sourceforge.net

- H323 - ISDN Gateway: http://www.telos.de/linux/H323/

Conferencing/VoIP Software

- OpenMCU: http://www.openh323.org

- Asterisk PBX: http://asterisk.org

Other H.323 Clients

- OpenPhone: http://www.openh323.org

- Netmeeting: http://www.microsoft.com

- SJPhone: http://www.sjlabs.com/