How to measure realtime latency, jitter and packet loss

General Help
1

Hi!
I need to test and compare different QoS solutions for my VoIP installations.
For this I need to measure parameters like latency, jitter and packet loss between my FreePBX VoIP server and my SIP (VoIP) provider in two basics situations:

  1. when PCs that share internal LAN network with IP phones don’t download and upload anything from and to Internet;
  2. when the same PCs are downloading and uploading files from and to Internet.
    If I look on “Asterisk Info > Peers” section of my FreePBX dashboard, I can see the - for each SIP connection - a number that I believe is the actual latency (i.e. 20 ms, 50 ms, …).
    But this number is not realtime updated.
    The questions are:
  3. wich program use FreePBX to show the actual latency of a SIP connection?
  4. may you suggest a software (free, if it is possible) that can perform the measurements that I need in real time and during a VoIP call?
    Thanks in advance.
2

I don’t know anything that will measure these parameters during a call but if you want to test system performance and reliability (in terms of latency, jitter, etc) then one tool is SIPp http://sipp.sourceforge.net/. This is a load testing tool for SIP servers really but it works reasonably well as a general testing tool as well.

1 Like
3

Most phones output RTCP-XR data after a call with performance stats. Asterisk has rudimentary tools to view this information.