Understanding the statisticsTo view the statistics, please select the Statistics tab in the control panel.The statistic visualizes the network traffic caused by . It draws a graph for each RTP stream. This means that - if audio and video are enabled in  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,  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 settingsYour 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 chatBesides the communication through live audio and video streams  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.
Text chat works with  to  clients only.Controlling the call 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.
All URLs supported by  (H.323, CALLTO and Speed Dials) can be used for call transfer.Taking a snapshotWhile 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 windowsHistory windows in  are comparable to logfiles. They keep chronological track of actions performed by  and provide additional information to the user.General HistoryThe 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 HistoryThe 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 . 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 .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.
