[asterisk-users] How can I improve call quality?

[Adrian Marsh]

Hi All,

I've a single 1.2.17 Asterisk system.  Gradwell here in the UK is used
for PSTN calls via IAX2.
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever
used 50% max bandwidth).  We've no E1/T1 links, everything is IP based.

My boss complains that many of the calls he holds with others has a bad
quality.  He also says that its not just him.

My iax.conf file has:  

disallow=all                    
allow=ulaw
allow=alaw
bandwidth=high
jitterbuffer=yes
dropcount=2
maxjitterbuffer=1000
maxjitterinterps=10
resyncthreshold=1000
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay
autokill=yes

He complains of broken audio, muffled audio, and says compared to Skype
its very poor, particularly during conference calls (zaptel meetme).
Most of these would be SIP based within our server though, rather than
IAX/PSTN based (X-lite/SJphone).


Obviously I can't do much about the far end IP connections/Mobiles etc,
but what can I do to tweak/improve the call quality on the A*k box
itself?

The CPU stays at a constant 10% usage, mainly due to a few other
monitoring apps on there (with these turned off, its < 2%, but still the
same issues).


Also - are there any useful stats/logs that I can examine to "see" the
quality of calls?

Thanks,

Adrian