Voice Quality Issues

Discussion in '3CX Phone System - General' started by d.barnbrook, Nov 22, 2012.

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  1. d.barnbrook

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    Hi Guys,

    Just a quick question, we are new users to 3CX not VOIP in it self but to 3CX managing the calls etc. We have a dedicated 2mb adsl/sdsl hybrid line ( not sure of the full actuall type of the line but i know its a bit of a mix ) and we have a 16 users licencse for concurrent calls.

    Basically we are getting very muffled audio on calls at times, our suppliers of the system are trying to work through the issues but its taking some time I was wondering if anyone would have any ideas ?

    I cant see any pattery as to when the bad audio happens, we very rarely go above 5 concurrent calls in any one time, i have noticed poor quality when they are this amount of calls but also when we have only 1 call so im not entireley convinced its to do with bandwidth etc.

    Any help would be greatly appreciated.
     
  2. mixig

    mixig Active Member

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    Hi,

    we deployed identical setup to our customer as you have, 2/2mbps dedicated link (sdsl) with license for 32 concurrent calls (in real life there are no more than 15-20 simultaneous calls), codec is G711, and everything is working well (call centar). So each call is taking 85kbps, 16calls*85kbps=1360kbps. In this case i think that bandwith is not an issue... Maybe bad link?
     
  3. leejor

    leejor Well-Known Member

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    Muffled is not usually the way "bad" call audio in VoIP is described. If packets are being dropped because of bandwidth, it usually results in noise, popping, clicks, dropped "bits" of words, etc.

    Are you working with one VoIP provider, or does this happen with more than one? I assume that internal call quality is OK?

    It might be a CODEC issue, but without a second provider to test to, you can't narrow down as to whether it is your network, or your VoIP provider.
     
  4. markshehan

    markshehan New Member

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    I agree with Leejor on this. Muffled is not usually a description of poor sip quality lines.

    If you want to drop me a pm I will quickly set up a sip trunk for you to connect your 3cx up to and you can test it with our lines. We have exceptional quality. If you prefer to keep your current 3cx partner in the loop we are more than happy to work with them too as we have many 3cx partners as our partners too.
     
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  5. d.barnbrook

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    Hi Guys,

    I think Leejor described out issues better " popping, clicks, dropped "bits" of words " we have used our installers SIP trunk to test that also and the quality still isnt great to be honest although we only have my calls going through that.

    I found out thats its not actually an SDSL line, it is an ADSL line but one that is slightly modified "apparently" im not sure on the technology but apparently its LLU broadband, what ever that is. Im begining to wonder if this line is suitable for VOIP.....

    Does anyone have experience with this LLUI broadband ?

    Its running through EntaNet which I believe is then backed off through O2.

    Thanks for your comments so far guys, appreciate it :)
     
  6. markshehan

    markshehan New Member

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    To make a call using G711 (u or a law) needs about 100k each way for the audio (the audio needs about 85k but then when you add all the headers etc then it is about 106k so just work on 100k for ease).

    So if you are getting the audio dropouts on 1 call then you have a problem that possibly isnt bandwidth related.

    It could be as other people are using your dsl for data (check for video streaming etc.)

    Can you put QOS on the edge firewall/router/dsl modem?

    Is this being nat'd?

    Did you pass the firewall checks in 3cx?
     
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  7. d.barnbrook

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    Hi Mark,

    The line we had installed was for sole use of voice so there is no other device on our network set to go out through this new gateway apart from the 3cx server, QoS is enabled on the router also.

    And yes the 3cx server did pass firewall tests in 3cx.
     
  8. markshehan

    markshehan New Member

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    That's good. Have you double checked the codecs being used? For example you might have G711u set up when you should be using G711a or some other codec mismatch with your provider.

    You could also try a wireshark just before the dsl modem and then you can play back the audio on a call. That way you can see if it left 3cx ok (ruling out a hardware or switch problem)
     
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  9. lneblett

    lneblett Well-Known Member

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    Llu is local loop unbundling. It is a mechanism by which the local loop owner (the carrier that actually provided the physical line to the premise) is forced or able to share the line with another provider. Presumably it allows one to provide analog voice while another might provide dsl...as an example.

    My guess is that your dsl provider is having issues getting the data thru the pipe in an efficient manner and you have jitter and dropped packets. Go to http://www.pingtest.net and run a few tests to different locations and check the results. If the results are not in the a or b range, you can expect issues and switching sip trunking agents won't solve the problem as you need a better carrier to get the data to & from.
     
  10. BrianKidd

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    We too had this problem now it is solved by our technicians.
     
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  11. 3CXfoxhallsolutions

    3CXfoxhallsolutions New Member

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    We have seen this problem and solved it by changing the codec priority order in the 3CX SIP Trunk config' and the phones.
    What kind of phones are you using??? If you can tell, I may have some suggestions ...

    Best regards
     
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  12. d.barnbrook

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    We have a polycom kirk solution for the handsets,

    1 x Kirk Server IP6000
    3 x kirk basestation 12's
    50 x polycom kirk 5040 handsets

    I will try some of those ping tests shortly and post results.
     
  13. d.barnbrook

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    Ok here are my results from the tests you suggested :

    Test 1:
    Packet Loss - 0%
    Ping - 18ms
    Jitter - 1ms
    Line Quality - A (MOS 4.39)

    Test 2
    Packet Loss - 0%
    Ping - 18ms
    Jitter - 1ms
    Line Quality - A (MOS 4.39)

    Test 3
    Packet Loss - 0%
    Ping - 18ms
    Jitter - 1ms
    Line Quality - A (MOS 4.39)

    Each test at random intervals appear to be the same, I will look into the codec order you mentioned ( well pass it onto our support guys )
     
  14. markshehan

    markshehan New Member

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    The tests look good.

    I still think it is a codec mismatch.
     
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  15. lneblett

    lneblett Well-Known Member

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    Those are indeed good results. While I guess codecs could be a factor, where It bothers me is that you indicated that not all calls are impacted. Codecs are negotiated and I want to think that both sides would tend to agree on which to use and once done, the call quality issues you mention would not occur.

    Are inside calls clear? Have you noticed if there is a difference between the external calls depending upon whether your side initiates the call or receives?

    Assuming the carrier is good, then you should try a different sip provider. Mark can ster you to one.
     
  16. d.barnbrook

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    Internal calls are fine, i havent noticed any issues with those at all really, however now we know what codecs our SIP provider supports we are going to test a few different ones to hopefully fine tune it if indeed this is the issue, I keep you guys posted.

    Thanks again :)
     
  17. sigma1

    sigma1 Active Member

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    Kirk=Dect possible problem, it's wireless

    Codec mismatch PCMU vs PCMA cannot happen. Part of the call setup is codec negotiation, this is impossible.

    BW results are great, muffled is also not something that can be caused by your digital analog stream, you will have clicks, skips but not analog changes. Try getting a SIP phone that is actually wired (using Ethernet) and test that way. Also, you can run a wireshark capture on the server and send me directly a capture when the issue happens, I'd be able to give you a better idea of the culprit. If the audio comes muffled to the server (and most SIP trunks will do Server delivers audio) you will know.
     
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  18. d.barnbrook

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    Thanks Charles, ill keep you posted on this, sorry to sound really dumb but could you elaborate on what PCMU and PCMA is ? Im really new to this
     
  19. sigma1

    sigma1 Active Member

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    G771U=PCMU, G711A=PCMA

    Also, make sure that under your Codec selection for the SIP trunk you have only G711U, not GSM or G729
     
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