[Asterisk-Users] Need help with jitter buffer/quality settings

[Matt Lawson]

I'm using Asterisk to do audio as well as H.263 video over SIP. 
 Actually the video works pretty well but I have trouble with the audio. 
 I'm wondering if someone can suggest codec/jitter settings or other 
tweaks.  The system looks like this:

Softphone <---ulaw---->  Asterisk #1 <------IAX (usually GSM)---------> 
Asterisk #2 <-------- IAX (usually GSM) --------> Asterisk #3 
 <------ulaw----->  softphone.

Now originally I had all jitter buffers turned off.  Today I tried 
turning the jitter buffers for Asterisks #1 and #3 on with 
maxjitterbuffer and maxexcessbuffer set to 1000.
That seemed to help a little.  Then I tried those settings for Asterisk 
#2 and it seemed the same or worse.  I have tried several different 
audio codec choices as well.

The puzzling thing is, the video works pretty well but the audio has the 
trouble.  It's largely bandwidth-related but a residential DSL line 
ought to be able to get out a good audio signal!  Sometimes the audio is 
good but most of the time it's garbled or cuts out completely.  We've 
tried reducing the bandwidth of the video signal, but that's not the 
whole story.

Now when I do an IAX2 show channels on Asterisk #1 or #2 (during a 
call), I see Lag and Jitter values in the tens of thousands and 
climbing, in spite of the fact that I set jitterbuffermax to 1000.  That 
also is puzzling.  I suspect I shouldn't be seeing large numbers like that.

We also do audio-only calls of course.  Sometimes those sound great but 
a lot of times those break up as well.

So, my Q's are:

1.  What should I set the jitter buffers settings for each of the 3 
Asterisks above?  The links between them may be low or high speed.

2.  Should I force a particular audio codec?

3.  Is there anything else I can do to improve the audio quality? 
 (besides a faster connection)

Thanks.