[Asterisk-Users] call quality monitoring

[mjr-asterisk at ranney.com]

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