TRY THIS intermittent distorted voice calls

[iworkhere]

Running pbx in a flash virtualize using sip trunks

While talking to outside callers people here us very distorted and say we sound like darth vader. If you talk long enough, 2min, the call quality returns to perfect. This doest sound like jitter to me. Our network looks good and bandwidth is very low.

So far this is only heard by external callers and we on the inside don't hear any distortion.

Any ideas?

 

[iworkhere]

Sorry for spelling errors. Typing on phone with autocorrect....

 

[Jake]

What is your typical ping time to your VoIP provider? I would do ping tests when you're on and off the phone and during heavy and light internet usage. QoS would most likely help.

 

[iworkhere]

average ping times are 8ms. we have qos everywhere and only run voice on the sip line. i did notice that jitter buffer was disabled. i enabled this, do you think this will help?

see attached for the jitter buffer settings i enabled. also the audio codecs im using are ulaw alaw gsm, please let me know if you suggest anything else.

thanks again!!

 

[rjm]

Check the settings on your phones. Make sure your codecs are matching up to your server settings. Order is important too. Is your phone jitter buffer off or on? I am thinking you want all jitter buffers off, but someone else might know better.

 

[iworkhere]

so within asterisk settings i have ulaw alaw gsm

im using polycom phones and changed the order to:

G.711mu
G.711A
G.729AB

does that seem right?

I only turned jitter buffer on the server, i have not touched the client side of things.

 

[rjm]

Yes and no. The first two appear right, but unless you have g729 installed and licensed up on the server, that won't work for you.

I don't think you need the jitter buffer on the server. Google that one. I don't use or need it and my response times are not as good as yours. I am thinking you have jitter set to adaptive and it is taking some time to sort things out.

Check out http://literature.schmoozecom.com/a...es/asterisk_sip_settings-module-userguide.pdf and you will see that this matches your situation. The receiving side is hearing garbled voice and your pbx has jitter enabled, so that is the receiving side of the call.

I'm by no means an expert, but I would disable the jitter buffer and see how it goes for you.

R

 

[iworkhere]

im not seeing anywhere in the article where they discussed my situation and the bad sound quality. where do you see that?

I used to have jitter buffer disabled, i just enabled it to see if that would help.

 

[rjm]

I was referring to your post. have you tried turning it off? That is what I would try.

 

[iworkhere]

turning what off?

 

[hbonath]

I would not use server-side jitter buffer.
What you likely will need to do is packet capture the calls at various points in the network to see where the packet loss/jitter is occurring. When you say you are running QoS everywhere, what does that mean? Are you marking/policing at your gateway and on your switch trunk ports? What do your policy-maps look like?

This is a rough road you're going down, I've spent years honing my QoS/jitter/packet loss skills and have a lot of experience in this area.

I run VoIPMonitor at my Datacenter on my sip trunks, customer PBXes and remote agents at customer sites. It's a commercial product but I would be blind without it.
You can run multiple instances of their sniffer agent on SPAN ports on your network but you'll need Wireshark to graph out the RTP.

Start there, with pcaps taken from SPAN ports, run them through Wireshark and analyze the graphs to see where the jitter is coming from. Then track it down from there link by link. Most likely it's coming from the WAN unless you are using PRI/POTS for a trunk to local asterisk.