Choppy sound, SIP calls within LAN

Hi!

I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE repository). As a clients I use XLite on Mac, all on the same LAN. Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM and plenty of disk space on LEVEL 5 RAID.

Calls to another SIP server (also asterisk) hosted by another company are 100% OK, so it is clearly problem with my server setup.

Background music (before pickup) runs fine, but transmitted voice sound is very choppy, no matter of which codec I use.

I have searched over net, and implemented one by one every reasonable receipt found, including.

[b]highpriority = yes
internal_timing = yes

transmit_silence = no

nat = yes
localnet=192.168.0.0/255.255.0.0
externip = xx.xx.xx.xx

dtmfmode=rfc2833 [/b]

Downgrading asterisk did not solved problem, too.

Anyone please help if possible…

Many thanks in advance for any suggestion(s).

Have you tried building asterisk from source yourself and seeing if the problem still exists?

Other things to check:
Make sure your clients are not sending comfort noise. Asterisk does not fully support it and it can cause audio quality problems.

Are there any warning/error messages being displayed on the cli when a call is made. Use ‘asterisk -rvvvvv’ to get some more verbosity in the cli output.

Do you have dahdi installed? What are the results when you run dahd_test?

Check dmesg for any errors and problems with the system.

Make sure the ethernet interface on the asterisk box is running in full-duplex. You should be able to use ethtool to confirm this.

[quote=“bkofd”]Have you tried building asterisk from source yourself and seeing if the problem still exists?
.[/quote]

Nope, I installed pre-compiled RPMs from SuSE repository.

.[quote]
Other things to check:
Make sure your clients are not sending comfort noise. Asterisk does not fully support it and it can cause audio quality problems.
.[/quote]

hmmm can you please explain how to do it?

.[quote]
Do you have dahdi installed? What are the results when you run dahd_test?
Make sure the ethernet interface on the asterisk box is running in full-duplex. You should be able to use ethtool to confirm this.[/quote]

Seems like no

dahdi_test
Unable to open dahdi interface: No such file or directory

I do not use any zaptel devices. Do I still need DAHDI?

Easiest way to check for comfort noise is simply to have the asterisk console up while making a call. If a client is sending comfort noise you will see an warning on the screen to the effect of “Please disable comfort noise on the client as it is not fully implemented in Asterisk”.

Dahdi can also providing a timing source even if you have no dahdi hardware installed. Asterisk 1.6 does have its own internal timing that does not rely on dahdi, but it might be worth trying dahdi to see if you get better results from it. I do not know how the Suse RPM was built, so you may need to compile dahdi and asterisk from source to use it.

Another benefit would be you could run dahdi_test to see how the timing on the system is. If it is way off this can indicate hardware or irq problems.

I had a similar issue and for me it had to do with the way the client system handles sip packets.

Most desktops don’t do SIP QOS.

The company I set the phone system up for used a very heavy database app that would pull data constantly and using the app while on a soft phone call caused quality loss.

I installed hard phones and the issue went way.

Sometimes soft phones are not the best solution.

This could be a bandwidth and or SIP service provider issue couldn’t it?

Imbus

Technology evangelist for automated phone technology - IVR - Predictive Dialers - Appointment Reminders - PBX - Auto Dialers