Advanced Usage
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## Advanced Usage

### Registering Additional Accounts

#### The accounts window

You can open the accounts window by selecting Edit -> Accounts. 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 services can be an Internet Telephony Service provider (like ekiga.net), or an IPBX (like CISCO, Nortel, or Asterisk).

#### Adding an Ekiga.net account

To add an Ekiga.net account, simply select Accounts -> Add an Ekiga.net Account in the menu. A dialog will appear and allow you to enter several parameters:

- User: You can enter your login.

- Password: You can enter your password.

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

#### Adding an Ekiga Call Out account

To add an Ekiga Call Out account, simply select Accounts -> Add an Ekiga Call Out Account in the menu. A dialog will appear and allow you to enter several parameters:

- Account ID: You can enter your account ID.

- PIN Code: You can enter your PIN code.

If you do not have an Ekiga Call Out account yet, you can subscribe for one using the 'Get an Ekiga.net Call Out account' link in the dialog. 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 Account menu of the window when the account is highlighted.

#### Adding a SIP account

To add a SIP account, simply select Accounts -> Add a SIP Account in the menu. A dialog will appear and allow you to enter several parameters:

- Name: You can enter the account name.

- Registrar: The registrar to 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.

- User: You can enter your login.

- Authentication User: 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.

- Password: You can enter your password.

- Timeout: The timeout after which the registration should be refreshed.

#### Adding an H.323 account

To add an H.323 account, simply select Accounts -> Add an H.323 Account in the menu. A dialog will appear and allow you to enter several parameters:

- Name: You can enter the account name.

- Gatekeeper: The gatekeeper to 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.

- User: You can enter your login.

- Authentication User: 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.

- Password: You can enter your password.

- Registration Timeout: The timeout after which the registration should be updated.

### Understanding URIs

#### SIP URI's

SIP URIs are formatted as such "sip:user@[host[:port]]"

This permits you to call the given user or extension on the specified SIP proxy: sip:jonita@ekiga.net

#### H.323 URIs

H.323 URIs 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 URI: 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

### Controlling the Video Bandwidth

Ekiga uses a best-effort algorithm to maintain a low bandwidth when transmitting video. It will permit Ekiga 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 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 Ekiga will dynamically increase them so that the quality and the framerate are always the best possible.

### Monitoring lines

Ekiga 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.

To enable that feature on Ekiga, simply add the contact with his URI in the roster. If the server supports publishing presence information, Ekiga will automatically publish your own presence information and display the presence of contacts in your roster.

### Managing Codecs

#### Audio Codecs

The Ekiga 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.

#### Video Codecs

The Ekiga 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. Ekiga supports codecs like H.261, H.263+, H.264, MPEG-4 or Theora.

#### Reordering the codecs

When you reorder the codecs, you are reordering the local capabilities table, ie the codecs you will use for sending. The codec used is the first active codec on receiver which is active on sender.

#### 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 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.

#### Adjusting the jitter buffer

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.

### Changing Ports

#### The listen ports

The main port used to listen for incoming connections in Ekiga for SIP is port 5060 (UDP), while 1720 (TCP) is used by H.323. To change those ports you need to load "gconf-editor". Open gconf-editor, select apps from the left hand side menu and then select Ekiga, Protocols. Then select "sip" or "h323", 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 ranges.

#### Explanation of the port ranges

1. The "listen_port" value is the port Ekiga will use to listen for incoming connections. It is different for SIP and H.323.

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

3. The "tcp_port_range" value is the range of TCP ports beside the listen_port that Ekiga 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 Tunneling is enabled, which is generally the case, except when calling old H.323 implementations like Netmeeting.

### Controlling the SIP and H.323 Settings

#### Controlling the SIP Settings

##### Misc Settings

Outbound Proxy

The outbound proxy is the SIP proxy that will relay your calls. The behavior of a SIP proxy is similar to the behavior of an HTTP proxy, i.e. some entity that issues the requests on your behalf and proxies the streams.

Forward URI

The URI to which SIP incoming calls should be forwarded if configured in the preferences.

#### Controlling the H.323 Settings

##### Misc Settings

Forward URI

The URI to which H.323 incoming calls should be forwarded if configured in the preferences.

##### Advanced Settings

Ekiga 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 achieving 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.

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