We are a prospective client of 150 users. Our biggest hindrance is latency, specifically the end to end delay, ie how long it takes for sound to travel from mic of one phone, to speaker of another phone. For us this is often 450ms one direction for some external calls. We have tried moving 3cx to a physical machine, Cisco QoS, 3xc server rebuild, 3cx server on linux instead of windows, different SIP provider, different router, different phones, codecs, disable SIP ALG, wireshark, analyse pcap, etc. We paid for a 3cx support ticket and 3cx support proved that 3cx was passing packets on quickly, but we are no closer to solving the problem. We installed a temporary Avaya IP500V2(SIP) and the problem went away, which is a shame because we much prefer 3cx. It would be awesome if 3cx could capture end to end latency and jitter information, ideally for all calls but if that is not technically possible, then at least for 3cxclient, 3cxWebclient, and 3cxMobileClient. And put this information in the detailed call reports. Or add some clever "test latency" buttons or online guide which can help systematically troubleshoot this problem.