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J4618C Keysight IP Telephony Analyzer

Status do Produto: Obsoleto | Veja opções de Serviço
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The IP Telephony Analyzer is a full-featured intelligent baselining and diagnostic instrument for analysis of VoIP installations. Voice playout allows a network engineer to isolate the cause of bad voice quality within the VoIP network. In some instances, listening to the audio is the only way to diagnose a problem. For example, an end user may complain of echo but when a network engineer listens to the audio he or she discovers the problem is background noise being fed in, even when the parties are speaking.

All IP Telephony terminating devices such as end points or gateways take out the jitter caused by IP networks. This de-jitter function is configurable. Someone listening to the audio inside the network may wonder if this de-jitter buffer will remove the audio impairments. A wonderful feature built into the J4618C IP Telephony Analyzer allows the user to hear the audio with jitter or "de-jittered".

In addition to audio playout, the J4618C IP Telephony Analyzer adds features that automate the troubleshooting of networks suffering from jitter and packet loss and also enhance the unique real time analysis of the Keysight Advisor. For example the IP Telephony Analyzer plots the average jitter and packet loss for each measurement interval. The user can click on a point of interest on the graph and a drill down window appears showing each packet within the measurement interval, its inter-packet arrival time and its jitter and exactly which packets were missing. A further click on graph point such as a jitter maxima takes the user to the decode view and places the cursor on the chosen packet. This takes the user instantaneously and automatically to packets surrounding the symptom and allows the user to troubleshoot in the vicinity of the problem.

  • Packet loss and jitter calculated for each individual RTP stream
  • Graphs the distribution of RTP packet loss and jitter values and over time
  • Comprehensive real-time Expert Commentators for VoIP Signaling and Voice packet transport
  • Replay previously captured voice packet sessions
  • Audio Playout - forward, receive, full-duplex, with jitter or de-jittered
  • Automated troubleshooting with drill down on suspect packets
  • Call Detail Records for each call or part call
  • Real-time decodes, filtering and analysis on LAN, WAN and ATM
  • The most Extensive list of VoIP decodes available including recent drafts