[Asterisk-Users] IAX and Jitter problem

Joe Dennick joe at dennick.net
Tue Oct 7 11:32:39 MST 2003


I'm coming at this thing from an Operational standpoint rather than a development standpoint.  Viewing your problem from that angle, I wonder how well your network is performing.  Could you have a cable problem that the Asterisk server hasn't reported (Layer 1); or perhaps your * Server is connecting at 100 megabit/half duplex, but the switch is configured (or auto-detected) 100 megabit/full duplex (Layer 2); or perhaps you have a bad port on your switch (Layer 1 or 2); or perhaps there is a mis-typed subnet mask or default gateway somewhere between the systems that hasn't been caught yet (Layer 3).  Is your Asterisk server busy doing anything else that's tying up resources?

You also menitoned that you haven't yet found a soft-phone.  X-Lite (from www.eten.com) works really well on Windows workstations.


Michael T Farnworth <mtf at maximasystems.com> wrote the Oct 7, 2003 12:52 PM:

> Thought I would just mention that I have a Pentium 150 with 64MB of RAM,
> asterisk installed, 2 Budgetone 102's and an X100P.  No problem with
> jitter here or anything like that.  I don't use mp3 music on hold because
> I doubt the hardware would cope particularly well.  Has anybody got 
> Asterisk running on anything lower spec than this?
> 
> Michael
> 
> On Tue, 7 Oct 2003 silverflash at bancclub.net wrote:
> 
> > Hello, 
> > 
> > I've been playing around with * for quite a while now, and have run into a 
> > problem that I just cannot seem to figure out. 
> > 
> > When using * and any IAX client (I have tested with GnoPhone and both 
> > clients from iaxclient.sourceforge.net) I have incredibly bad jitter on the 
> > connection. 
> > 
> > What I'm running is a P3-1Ghz machine with 512mb ram for a server.  The 
> > other end has been various machines (all connected via 100mb switch) ranging 
> > from a AMD K6-2 350 running Win98 to another P3-1Ghz running RH Linux 9.0 
> > and GnoPhone. 
> > 
> > I've tried changing the jitterbuffer settings in iax.conf (including turning 
> > it off as I've seen some recommendations on the archives) and I've even 
> > tried rebuilding zaptel with the various jitter control switches. 
> > 
> > At this point I have extension 8500 setup to take me to voicemailmain.  When 
> > I connect (IAX only - I do not have any Digium cards in the server at all) I 
> > can generaly not tell what is being said at all.  I've used sox and a player 
> > and know that the .gsm files are okay. 
> > 
> > Anybody have any suggestions of what to try?   So far this has been 
> > something I've been playing with before I attempt to put it in a production 
> > system, but so far am not having a whole lot of luck. 
> > 
> > I've not been able to try SIP as of yet, as I've not found a softclient and 
> > the application I will be using * for would require this. 
> > 
> > Thanks,
> > Mike Atkinson 
> > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
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