Audio delay - LATENCY

Discussion in 'Ideas' started by palmaz, Mar 23, 2018.

Audio delay - LATENCY 4.5 5 2votes
4.5/5, 2 votes

  1. palmaz

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    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.
     
  2. timmbo

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    Hi,
    have you tested it with *777(Echo)?

    Bye
    Timm
     
  3. palmaz

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    We found that Snom 300 phones are crazy fast on the *777 echo test, I'm guessing <60ms (1/8 sec). 3CX support also proved that our 3CX server is passing packets within 1ms.

    We also managed to fix the latency problem on the Yealink T46S using the following settings, applied manually on the phone (not via 3CX console). The end result is about 60-100ms which is fast enough for us.
    Browser > Yealink T46S > Settings > Voice :
    Jitter buffer = Fixed
    Jitter min delay = 20
    Jitter max delay = 20
    Jitter normal = 20

    The 3cxClient for Windows still adds 200-250ms of latency on the echo test. This is a problem because if there are 2 people on an audio conference, then this quickly jumps to 500ms.
    I logged this with 3CX and they say "there are no known issues with 3cxClient" implying that the issue must be on our network. They say there are no settings to control jitter buffer on the 3cxClient. I suspect that the 3cxClient automatically adds 200ms buffer to help smooth jitter.

    If anyone out there gets better than 200ms on the *777 echo test from 3cxClient for Windows, I'd be interested to know how your doing it.
     
  4. TP_

    TP_

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    hmm any info on that?