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:
- when PCs that share internal LAN network with IP phones don’t download and upload anything from and to Internet;
- 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:
- wich program use FreePBX to show the actual latency of a SIP connection?
- 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.