Analyze details of VOIP-calls in your network
Voice Engineers depend on proper information to handle issues in a VoIP network.
Logfiles and CDR's might be a first source of information to understand the core of a reported problem, but compatibility issues, timeouts, and weird "Information Elements" do not show up in a CDR.
To get to the bottom of this, you need a Protocol Analyzer.

Now, to start a "Wireshark"-Trace every time is very time consuming and—on top of that—the reported behaviour needs to be reproduced. Eventually, your customer looses patience.
With SipXpose though, you already have that data. You simply search a dialed number, IP-address, Call-ID, Release Cause, etc. and all messages related to the call are in front of you in the browser.

SipXpose records in real-time; all data is available within 5 to 10 seconds, and you can tell your Interconnection Partner or customer immediately about necessary adjustments, regarding "number format" for B-number and CLI, Diversion Headers, Codecs in SDP.

The web-interface allows you to find the calls you are interested in by the following

- To-URI (called number)
- From-URI (calling number)
- Date/Time from
- Date/Time to
- Release-Cause
- Call-State
- Duration (range)
- PDD (range)
- src-ip
- dst-ip