[Asterisk-Users] call quality monitoring

Chris Icide cicide at gmail.com
Sat Sep 11 10:20:16 MST 2004


mjr,

Satellite links can be pretty tough to troubleshoot.  It sounds like
you are running into a uplink buffer issue.  On heavily loaded
uplinks, the input buffers can get quite large, and if the satellite
provider isn't using some form of buffer handling that prioritizes udp
traffic, it may be that most of your voice packets are falling on the
floor of the uplink facility...

-Chris


On 10 Sep 2004 11:37:33 -0700, mjr-asterisk at ranney.com
<mjr-asterisk at ranney.com> wrote:
> I need to debug a call quality issue with remote users on the other
> end of a satellite link.  The symptoms are: we here on the Internet
> side can hear them just fine.  On their end, things work sorta OK most
> times, but they often suffer from severe dropouts and digital
> warbling, both of which I attribute to them missing packets.  Often
> times they can't make out a word we are saying while we can hear them
> crystal clearly.
> 
> Various pings and other network tests indicate that the underlying
> network is functioning as well as can be expected for a sat link.  In
> fact, the overall jitter seems to be pretty low (avg 20ms).  Packet
> loss is around 1-2%, and latency is around 700ms on average.
> 
> I'm left to assume that the jitter buffer on that end isn't
> functioning properly.  Both ends of the call have the same jitter
> buffer settings.  The call is carried by IAX2 and encoded with ILBC.
> 
> The iax.conf files on each end start like this:
> 
>   >    [general]
>   >    trunk=no
>   >    notransfer=yes
>   >    iaxcompat=no
>   >
>   >    bandwidth=low
>   >
>   >    disallow=all
>   >    allow=ilbc
>   >
>   >    jitterbuffer=yes
>   >    dropcount=3
>   >    maxjitterbuffer=500
>   >    maxexcessbuffer=150
>   >    minexcessbuffer=40
>   >    jittershrinkrate=1
> 
> Of course, perhaps the jitter buffer isn't to blame, but given that
> one side of the call sounds perfect, I can't think of anything else
> obvious that would cause this.
> 
> Is there any way to extract from asterisk some idea of why it thinks
> the calls sound bad?  For example, when the jitter buffer notices that
> packets are discarded because they are too late, when excessive
> packets are completely missing, etc.
> 
> I've been collecting a giant debug log for a while now, so I could
> pretty easily sift through it if there's something good to look for.
> 
> Thanks.
> --
> Matt Ranney - mjr at ranney.com
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