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Audio delay - LATENCY

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

Audio delay - LATENCY 4.8 5 4votes
4.8/5, 4 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

    timmbo New Member

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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?
     
  5. palmaz

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    Request to 3cx developers: While you are building a soft client into the WebClient, if you are adding jitter buffer (which adds latency), please could you make this setting configurable.
     
    TP_ likes this.
  6. TP_

    TP_

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    Why no info on that topic?
     
  7. palmaz

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    More detail:
    *777 echo test from Snom 300 === 50ms (good)
    *777 echo test from Yealink T46S (with jitter buffer=20ms) === 65ms (good)
    *777 echo test from 3cxClient v15.5.8801.3 on Windows Pro === 200ms (terrible)
    (wired 3.5mm mic & speaker, ie no headset or USB latency).
    So it looks like 3cxClient adds 150ms for the echo test.

    Yealink T46S to Mobile === 333ms
    3cxClient to Mobile === 440ms (so 3cxClient is adding 100ms - one direction)
    3cxClient to Mobile on 3cx audio conference === 600ms (so 3cx audio conference adds 150ms - one direction)

    To put this into perspective, a Shoretel/Cisco/Avaya (virtualised) audio conference including mobile phone and soft phone is approx 300ms.
    All tests were done with using PCMA/711a throughout, and with echo cancelling type features off.
    We have tried installing clean Windows 7 on server grade hardware and using that as a client machine.

    I think there are 3 problems:
    1) 3cxClient (softphone) adds 100ms latency (for audio coming in). (compared to 50ms for Cisco/Avaya softphones).
    2) 3cx audio conferencing adds 150ms latency (even when 3cx is on HP Gen8 hardware) because it is slow at transcoding.
    3) When a cellular participant is involved, the latency problem is magnified.


    We could move audio conferencing to the cloud, but that does not fix ad-hoc audio conferencing, when you want to merge calls.
    And also that 3cx has no call monitoring stats, so it becomes very difficult to find/analyse/fix this sorts of problems.

    On the whole 3CX is an excellent product but unfortunately this latency delay issue is show-stopper for us.

    I have to wonder if WebRTC in the WebClient will be worse for latency, since there is additional codec translation, and a browser to contend with.

    I hope that 3cx give serious attention to latency, call quality monitoring and audio conferencing.
    If you agree please vote.
     
    #7 palmaz, Jul 25, 2018
    Last edited: Jul 25, 2018
  8. palmaz

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    @Nick Galea - You mention in another post that 3cxClient is "here to stay and also will be improved and rewritten"
    Will the rewrite address latency? And are there any plans to reduce latency inside conferences?

    Great to hear that 3cxClient will stay.
     
  9. TP_

    TP_

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    Thank you for the nice Test!

    @3cx any comment on that?

    Bear in mind too that 3cx plan to remove CTI from 3cxClient making it just a softclient

    is that true!? Link?
     
  10. palmaz

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  11. TP_

    TP_

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