Sys Mon RTP (S)RTP Sessions View IPO 11.1 SP 1

[Stengrad]

I hope someone can provide me with a bit of info on this tool. I'm trying to track down some QoS/Call dropping issues, and was using SSA QoS Alarms to help, until I discovered that those alarms aren't very reliable. I read about the Sysmon RTP Session view and checked it out. I don't understand the column showing "Loss" and why at some point during the call, the number in that column goes to 256, and none of the other columns update with any further info on Delay, or Jitter. Attached a screen shot.

Thanks for the help!

Senior Field Technician
Hampton Roads Communication Technologies

Http://hrct.net

 

[CarGoSki]

Wireshark is the answer for SIP or h323 calls. You have to convert h323 packets before they can be translated correctly but SIP needs no translation.
You are better off collecting packet traces then replay the actual call.
You will be able to see packet loss for the call and two way audio so you can view where are the audio issues.
If it is a Server Edition then turn on packet tracing from webcontrol session via port 7071. Be sure to turn it off when complete.
Otherwise get a laptop and install wireshark. Connect to net and port mirror IP Office lan port.
Start wireshark and capture packet and once you have capture loaded in wireshark:
Select Telephony\VoIP Calls > the app will present a window with calls found.
Highlight one call then select Prepare Filter button.
Select the first RTP packet then select Telephony\RTP\Stream Analysis.
When complete you will see stats for the calls including packet loss and packets out of sequence.
Select Play Streams button.
You should now see both distinct call streams and you can playback and listen to hear audio issues for the send and receive ends of the call.

Happy hunting.

 

[derfloh]

Wireshark only helps for unencrypted connections.

IP Office remote service
IP Office certificate check
CLI based call blocking
SCN fallback over PSTN