Distortion on SIP phones while using Trixbox.


We have a Trixbox server on a dedicated T1 line using SIP trunks.  Six Aastra 9133i SIP phones are registered to this server on the LAN.  Sometimes while they are using these phones, the external caller's voice will become very distorted (like the call has just gone underwater or very robotic) and the inside users will not be able to understand the external caller.  The external callers will have no trouble with distortion or with hearing the inside user while this is going on.  This happens randomly at least once a day.  However, the recordings of these calls show no sign of distortion.  This happens on our Trixbox with our Aastra phones on another site completely as well.  Is this something that can be fixed by adjusting the jitter buffer?  Are there any suggestions on how to diagnose this issue or fix it?

Thank you
OAC TechnologyProfessional NerdsAsked:
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ehatchellConnect With a Mentor Commented:
Well then, its either the unmanaged switch or the phone.  I would suggest opening a trouble ticket with Aastra, and getting your hands on a very well known high quality phone such as the Cisco or Polycom as previously mentioned...
Being that the switch is unmanaged, I figure LAN stats are out the question to see if there is some other traffic impeding the voice traffic here, but given its one way, again, I think its more the likely the phone as you seem sure the QoS is solid and the recordings on the server exibit no distortion for both sides of the conversation.
You can try turning off silence supression if enabled on the phone and see if just a buggy issue there.
Also, you can add a managed switch to the mix, start collecting some stats there.  Without stats on each leg and varying devices, its a crap shoot.
Adjusting a jitter buffer and other settings on the phone is probably irrelevant here since the the server records the call fine as you mentioned above (and your stated stats on the WAN seem fine, along with your stated QoS).  So, if there is something, its happening on switch or the phone (assuming the switch and asterisk box use the same switch) otherwise, all switches between them should be suspect.  What model of switch is in place?
First step, make sure your Aastra phones have the newest SIP load (firmware) on them.  Second, update the software on your Trixbox and also make sure its the most current stable release.
I don't know what kind of switches you are running, but make sure you have QoS enabled and properly configured for all of the switches touching these phones and up to the server and the router.
Can you pull stats from your Trixbox via SNMP?  How does the health of the box look at those times (CPU, Memory, I/O, etc)...?
What type of router do you have in place?  Does it have any form of QoS?  If so, make sure its configured correctly, and depending on the router, look at its CPU load/MEM also at the same time, perhaps a firmware update should be considered here to the latest stable release.
What codec are you using?  Are you performing any transcoding?  If so, where?
Who is your SIP trunk provider?  I'm assuming your connecting to them over the Internet.  Do a trace route between you and them and post here.  Also, do a constant ping to SIP provider from your router for at least 10 minutes and post here too.
Do you have any troubleshooting tools available to you?
Let's start with the foundation of VoIP and move to the more complex items for troubleshooting as we set the foundation.  Please post all steps performed, prior and current revisions of all updates, etc.
OAC TechnologyProfessional NerdsAuthor Commented:
The Aastra phones have the latest firmware and the Trixbox is all up to date with the most current stable releases as shown through the package updater.

The switch we are using is an unmanaged 10/100 switch.  The router is issued by the phone company and does have QoS enabled and is properly configured.

Stats on the server look good.  No swap is being used and the server still has at least 600MB out of 2GB free with the CPU load ranging from 1% to 35%.  

The codec we are using is ULAW with no transcoding.

Our SIP trunk provider is Cbeyond.  This is a direct connection to their hardware and does not go over the public internet.

The only troubleshooting tools I have are the GUI interfaces on Trixbox and using Putty to SSH in to the asterisk server.

Thank you
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Can you post a long ping to your SIP provider and also a traceroute?
Do you have another brand of phone (e.g. Polycom) you can try out here?  I've heard of issues with some IP phones, and although I'm aware of Aastra (and I've used their analog products, I have not their IP), I only deploy Cisco or Polycom in Asterisk type environments because that's what I feel confident with.  So, let's see if you can swap one out.
What type of router did the phone company put in?  What other traffic is on this T1?  If the router is properly configured for QoS, how are you able to verify its operationally working as configured?
Of course, the switch is suspect being unmanaged, but we'll just have to look past that at this point and assume its not the issue.
You state you are recording these calls?  What app are you using for recording, and how is this being done (trunk or station)?  And at the same moment when the calls are have distortion, you've been able to cleanly listen to the whole call with the external caller via recording?
OAC TechnologyProfessional NerdsAuthor Commented:
Ping results show a consistent reply of 15ms, and the tracert shows between 15-36ms across 7 hops.  The only traffic on that line is the SIP traffic.  It is a dedicated line.  I don't believe it is an outside networking issue because the recordings show that there is no distortion.  They use the built-in recording feature of Asterisk (using *1 during a phone call).  The problem must lie somewhere between the Trixbox and the phones.  We do not have any other phones to swap out.

To have reliable data during the problem, you need a dedicated tool to catch network statistics and help you to investigate.

I may recommand : something like VQmanager :

Hope this helps

SR20 Service / France
denisdsr20Connect With a Mentor Commented:
+ I know about some trouble with the echo cancelor on T1:T2 interfaces, it requires some burst of CPU that can decrease voice quality.

Try to switch off echo cancel and test ....


SR20 Service / France
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