VoIP Sound Quality : Highroad Sound

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Asterisk Users 8 Comments

Hello,

what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing.

To be clear : the person calling is not standing next to a highway.

But there seems to be a noise “on the line” of busy highroad that makes that the caller can not be understood.

What can be causing this kind of “poor quality” ?

Is it lack of resources on the Asterisk-server (codec translation ?) Is it lack of bandwith ? Is it a problem of CentOS (the underlying OS) ? Is it a physical problem of the server components (network interface ?) ?

Kind regards, Jonas.

8 thoughts on - VoIP Sound Quality : Highroad Sound

  • Current situation :

    sip1*CLI> sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %)
    Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 0000000243 0000000000 ( 0.00%)
    0.0000 0000000576 0000046854 (8134.38%) 0.0002
    X.X.X.42 3c8956648ce 00:02:27 0000007301 0000000000 ( 0.00%)
    0.0000 0000007318 0000000001 ( 0.01%) 0.0020
    X.X.X.224 684333f5650 00:00:03 0000000000 0000000000 ( 0.00%)
    0.0000 0000000178 0000000000 ( 0.00%) 0.0000
    X.X.X.98 5eceb3a5624 0000000000 0000000000 ( 0.00%) 0.0000
    0000000000 0000000000 ( 0.00%) 0.0000
    X.X.X.98 25ae26ee564 00:00:03 0000000179 0000000000 ( 0.00%) 0.0000
    0000000000 0000000000 ( 0.00%) 0.0000
    X.X.X.98 6b26738a0c4 00:00:43 0000000137 0000000000 ( 0.00%) 0.0000
    0000000137 0000000000 ( 0.00%) 0.0001
    X.X.X.100 2f9a96ec3e1 00:00:42 0000000138 0000049741 (99.72%)
    0.0000 0000000136 0000000000 ( 0.00%) 0.0031
    X.X.X.70 68289fc05ff 00:02:27 0000007318 0000060143 (89.15%) 0.0000
    0000007301 0000000000 ( 0.00%) 0.0001
    X.X.X.184 6893e957-fa 00:05:59 0000000576 0000000000 ( 0.00%)
    0.0000 0000000243 0000000000 ( 0.00%) 0.0027
    9 active SIP channels

    Thanks.

    Jonas.

  • Yes, all SIP.

    Current situation :

    sip1*CLI> sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %)
    Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 0000000243 0000000000 ( 0.00%)
    0.0000 0000000576 0000046854 (8134.38%) 0.0002
    X.X.X.42 3c8956648ce 00:02:27 0000007301 0000000000 ( 0.00%)
    0.0000 0000007318 0000000001 ( 0.01%) 0.0020
    X.X.X.224 684333f5650 00:00:03 0000000000 0000000000 ( 0.00%)
    0.0000 0000000178 0000000000 ( 0.00%) 0.0000
    X.X.X.98 5eceb3a5624 0000000000 0000000000 ( 0.00%) 0.0000
    0000000000 0000000000 ( 0.00%) 0.0000
    X.X.X.98 25ae26ee564 00:00:03 0000000179 0000000000 ( 0.00%) 0.0000
    0000000000 0000000000 ( 0.00%) 0.0000
    X.X.X.98 6b26738a0c4 00:00:43 0000000137 0000000000 ( 0.00%) 0.0000
    0000000137 0000000000 ( 0.00%) 0.0001
    X.X.X.100 2f9a96ec3e1 00:00:42 0000000138 0000049741 (99.72%)
    0.0000 0000000136 0000000000 ( 0.00%) 0.0031
    X.X.X.70 68289fc05ff 00:02:27 0000007318 0000060143 (89.15%) 0.0000
    0000007301 0000000000 ( 0.00%) 0.0001
    X.X.X.184 6893e957-fa 00:05:59 0000000576 0000000000 ( 0.00%)
    0.0000 0000000243 0000000000 ( 0.00%) 0.0027
    9 active SIP channels

    Thanks.

    Jonas.

  • 2013-11-12 17:42, Jonas Kellens skrev:

    A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this
    (it gives you a pcap for each call), but tcpdump works fine also.

    This could be a congested link, a broken media gateway, or anything.

  • I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the “highroad” sound (always on the upload side).

    What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ?

    Kind regards,

    Jonas.

  • I frequently use Audacity to analyze the audio data. In many cases I can see from the spectra
    (and other graphical representations) with what kind of problem I am dealing. Meanwhile, for most of my problems I no longer depend on an audio editor. I don’t know whether this is helpful in your case.

  • 2013-11-13 11:55, Jonas Kellens skrev:

    Here is some suggestions to get started:
    http://www.enterprisenetworkingplanet.com/unified_communications/troubleshooting-common-sip-problems-with-wireshark.html

    Maybe one of your connections get congested? For example, if the two endpoints is your phone and the upstreams teleco. If the side from the teleco are bad and not the phone you need to take a closer look at the switches and routers on the way to the teleco. For example you can run tcpdump on your gateway to your ISP.

    If you see the problem here as well it may be your link or a upstreams problem. If you don’t see it here it is somewhere in between..

    Good luck!