Fize Size:30864KB Language:English OS:Win2000/XP/2003 Date added:October 11,2007
With comprehensive built-in and customized reports, it gives network administrators critical insights into bandwidth usage.
ManageEngine VQManager is a powerful, web-based, 24x7 real-time QoS monitoring software for VoIP networks.
VQManager can monitor any device or user-agent that supports SIP(RFC 3261) and RTP / RTCP (RFC 3550).
VQManager aids troubleshooting VoIP calls for failures and quality deterioration. Once identified, alarms are generated by the system to notify administrators. Subsequently, administrators can drill down on the alarms in order to isolate the root cause.
It enables IT administrators to monitor their VoIP network for:
�� Voice Quality
�� Call Traffic
�� Bandwidth Utilization
�� Active calls and Failed calls
�� With comprehensive built-in and customized reports VQManager gives network administrators critical insights into bandwidth usage trends, traffic patterns, Call Details Reports etc
Here are some key features of "VQManager":
Non-Intrusive Real-Time Monitoring
�� Proactive, continuous monitoring of the QoS & Bandwidth of the VoIP network
�� Real-time display of key information such as Jitter, Latency, Packet loss, Call count, Average call duration, Bandwidth utilization, Alarms, Top talkers etc.
�� ITU standard E-Model based MOS calculation
�� Drill down of information to minute level of precision
�� Different options to import call information using CDR files through FTP access or uploaded via HTTP. Alternatively, the CDR can be sent as a Syslog message too
Alarms and Notifications
�� Alarm generation on vital parameters viz., Too many consecutive incomplete calls, ALOC (High or low), Low ASR , High Average Answer delay time and High Voice Bandwidth utilization based on user-defined threshold and severity
�� Configurable E-mail alerting
�� SNMP trap generation and forwarding to integrate with higher level network management systems
Call Details View
�� Overall QoS trend pertaining to call and participants
�� Codec details
�� Pictorial representation of the call flow plotting all the SIP requests that took place from call start to call end. Will come in handy for debugging error calls
�� Call Trace of each and every SIP/RTCP packets transferred in a call
�� Pictorial representation of call flow information to debug call setup issues
�� Bandwidth usage graph with split up between voice and non-voice data to help analyze bandwidth congestions
End Point Details View
�� Detailed view of call usage trend & QoS Metrics
�� Call Metrics such as frequently dialed & received numbers, total active Call duration, Average call duration & ASR
�� Call categorization such as incoming, outgoing, unanswered, error, good and poor quality calls
Capture and Display Filtering
�� Filtering criteria to capture / monitor call traffic from specific IP Addresses /locations
�� Flexible filtering of data based on IP address, SIP URL, phone number and name
�� Can be used to monitor any device or user-agent that supports SIP and RTP/RTCP
Web-based User Interface