Poor Call Quality and erratic performance

Discussion in 'General' started by kmullen, Oct 29, 2010.

  1. kmullen

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    We have been using a FonalityPBXtra for over two years now. With little to no issues. My setup is rather unique so I will describe.


    PRI <> FonalityPBXtra <> Switch
    /> 5 gig Wireless to remote building < (12 extensions)
    \> LAN POE (17 extensions)

    This system has been working fine for over two years.

    New Setup:

    VoIP<>Internet T-1<>Elastix 2.0
    /> 5 gig Wireless to remote building < (12 extensions)
    \> LAN POE (17 extensions)

    In general, we have very few problems in the building where the PBX is installed. However the remote building served by the wireless has had issues since installation 2 months ago. Poor call quality and erratic phone performance.

    What is different from PBXtra and Elastix that would create these issues and how can I resolve them.

    Note: 5 gig is running QOS with Voice priority. There have been no changes on the wireless setup. ZERO Packet loss on wireless link and latency less than 2 ms

    The two principal changes are VoIP on T-1 instead of PRI and replacement of PBXtra with Elastix. All phones have remained the same. They are Polycom HD550's
     
  2. coryjsanders

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    I have had customers who have gone through 3 T1s with different carriers. T1/Cisco 1841/ASA5505 using hosted VOIP. QoS set. Everything should work. Could never do it. Brought in Qwest DSL FTTN circuit and ran all SIP, or another stable broadband circuit. Perfect. You'll also save a bundle over the slow and expensive T1 if you can get broadband cable or DSL to your location.

    Who is you SIP provider?

    Could be some bad hops between you and the CO. Run a jitter test on the T1

    http://208.93.128.14:81/voip Click on the graph when the test is done.

    Ideally you want to be below 10ms jitter both up/down with no huge spikes. If you're above 10ms avg jitter and/or getting spikes at 20,30, 50ms (or forbid greater), you're sunk. Jitter is defined as the difference in time it takes packet A and B to get to the same spot. Too much difference, you lose packets, and you're in the cr p r or worse.
     
  3. kmullen

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    Thanks for the info. I will test soon. I have to reroute a host to test. All PC/Server Internet access goes through a secondary Internet Provider (Wireless). All VoIP traffic goes across Verizon T-1.
     
  4. coryjsanders

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    who is the SIP provider? Or are you getting DID from Verizon?
     
  5. kmullen

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    SIP provider is Aretta. aretta.com
    I have another customer utilizing 6 VoIP Channels on a Verizon T-1 out of the same CO w/o issue.
     
  6. coryjsanders

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    yep. Run that jitter test.

    By chance do you have Call Center up and working?
     
  7. kmullen

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    Call center is not installed.

    Any way to test from the CLI or Asterisk CLI?
     

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