The Calls tab provides information on all the calls that have happened in the time duration as set in the calendar. Detailed quality metrics and trends for all calls are displayed through:
This is the first UI that opens in the calls tab showing the details of the active calls happening in the network.
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The QoS of the active calls are depicted in the form of intuitive charts. The average values of all the QoS parameters such as Delay, Jitter and Packet Loss are plotted in the chart.
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A Bandwidth utilized graph is also provided that shows the most recent bandwidth utilization in the network. The bandwidth usage is split into SIP, Skinny, H.323, Voice and Others. On rolling the mouse pointer over the different areas in the pie chart, respective volume of packets for the bandwidth components are shown in kbps.
By default, this page gets reloaded every 20 seconds to show the latest Active calls that are happening in the network. The user can configure this refresh rate. This option is found in the top left hand side of the UI just below the line of tabs. Next to the “Refresh Every” field click on the edit icon next to the linked text entry. Select the required refreshing time interval (refresh rate) from the drop-down list that appears.
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Below the QoS charts is a list of all the current active calls in the network. Information such as the Initiator of the call, call initiation time, Duration, Status Code, and MOS of the call are displayed by default in the table.
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One can mark any stale/frozen calls in this list as unmonitored or even delete calls that wrongly appear as active in this list. These options are found above the list through the two buttons "Delete" and "Mark as Unmonitored".
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You can view other details such as participant/initiator's ip, participant/initiator's url, call initialization time, call ringing time, duration, call setup time, MOS of initiator and participant, Minimum, Maximum and Average values of delay, packet loss, Rfactor and Jitter through the 'additional columns' option. To view the additional columns, click the column chooser icon
present at the top right of the 'Active Calls' table. Click "Save" after selecting the required columns.
This list can be arranged by ascending or descending order of the Initiator, Participant, Duration and other column fields. Clicking on the column heading, that you want the list to be sorted according to, does this.
The Active Calls list can be exported as a PDF or CSV file or emailed as a PDF attachment to a number of E-mail Ids. This can be done through the respective icons - CSV icon, PDF icon and mail icon, found on the right-hand side just above the Active Calls list.
The list can be searched for a particular call by providing either Initiator / Participant / Call Status / Status Code. Click on the lens icon found on the extreme right hand side of the list’s header. Text-boxes will appear in which you can provide the Initiator / Participant / Call Status / Status Code to search for the endpoint. After providing the search criteria, click on the Go button found below the lens icon to search for the call according to the provided search term/terms
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On clicking any Initiator of a call in any of the Call lists, you are brought to details of that particular call. This UI shows a graphical display of the call QoS trends, participating endpoint details, CODEC details and a pictorial representation of the packets exchanged during the call.
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The Call Trend graph is similar to the Voice Quality graph in the Monitor tabs Summary Report View. All the QoS parameters
such as Delay, Jitter, Packet Loss, MOS and R Factor are plotted in the graph. This shows the trends for the various QoS parameters for the duration of the call.
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To the right of the Call Trend graph, the details of the call and the participants of the call are displayed. Call details such as Start Time, Duration, Call Identifier and Call Status are shown. The Call Identifier is provided for SIP-based and H.323-based calls and is a unique identifier for each call. The linked character set shown for the Call Identifier is displayed as a truncated item. To view the full Call Identifier, click on the displayed Call Identifier to open up a dialogue box having the Call Identifier in full. Endpoint details such as Phone ID, Name, Skinny/SIP/H.323 URL, IP Address, Media Port, Octets and Packets are shown.
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On clicking the linked item View Trend found below the Name of the participants, you can see the individual QoS trend graphs for each participant during the call.
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Codec
Details give information about the Codec used in the call along
with its characteristics such as Bits per sample, Frame Size, RTP Clock Rate, Payload Type, Sampling Rate, Packet Size and No. of Channels.
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The Call
Flow ladder diagram provides a pictorial representation of the packets exchanged during the call initiation, progress and end.
It plots all the SIP, Skinny and H.323 requests that took place from call start to call
end. This is useful for debugging error calls that failed due to some
unknown reasons
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Call
Trace provides summarized information for each of the SIP/ Skinny/ H.323 and RTCP packets
that were exchanged during the call process. Details of Capture Time, Protocol, Status, Code, Source Port, Destination and Destination Port are provided for each packet.
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You can see the raw SIP, Skinny and H.323 packet details along with their Call Trace information by clicking on each packet in the Call Flow diagram. This information is displayed below the Call Flow diagram.
"Tunnelled" H.323 packets are represented in the call-flow diagram as a three-dimensional box.
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This listing is found as a separate menu bar on the left hand pane of the UI below the Quick Links section. Details of all the Error Calls, Unanswered Calls, Good Quality Calls, Poor Quality Calls, Unmonitored Calls and All Calls can be viewed by clicking the respective links in the menu bar
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Error Calls are those that did not connect ("Not Found", "Temporary Unavailable", "Cancelled") or those which failed abruptly after connecting(server shutdown, connection failed etc.)
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Unanswered Calls are those in when the called party does not answer. These could be because the called party is on another call ("User busy") or simply does not answer
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VQManager distinguishes from Good and Poor Quality calls based on the user's setting in the Monitor tab in the "Configure" section of the Voice Quality graph. By default, calls with MOS (Mean Opinion Score) greater than 3.1 are classified as Good Quality Calls. Those calls with MOS less than 3.1 are marked as Poor Quality Calls
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Unmonitored calls are those that VQManager has abruptly stopped monitoring - this could be due to:
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an abrupt stop and restart of the VQManager server, such calls are identified by status code "8001"
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an abrupt stop and restart of the VQManager sniffer, such calls are identified by status code "8002"
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Signaling flow interruption as configured in the "Advanced settings" of the sniffer configuration. Such calls are identified by status code "8003"
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User manually marking a call as "Unmonitored" in the "Active calls" section of the "calls" tab. These calls are identified by status code "8000".
The last link in the Calls by Status menu bar shows the details of all the calls that have taken place in the time duration as set in the calendar.
