TIPS help reducing audio lag

[JerryS]

Hi,

I'm a happy user of PIAF.

I am trying to sort out the significant audio delays in my system. When I do an echo test on the local lan and snap my fingers, there is probably a half second delay hearing the response. When I have a call come in from outside and then forward it back out, delays are more like 2 seconds. I am using a lightly loaded vm server (I want to put this in a proxmox container instead, but haven't had time to figure out all the things to make asterisk happy in that mode.) The local phones are polycom sp335s. I am using flowroute as my SIP provider, and my ping times there are in the order of 25ms.

Here what status shows about the system.

PIAF Installed Version = 3.0.6.5 under *KVM* │
FreePBX Version = 2.11.0.38 │
Running Asterisk Version = 11.10.2 │
Asterisk Source Version = 11.10.2 │
Dahdi Source Version = 2.9.1.1 │
Libpri Source Version = 1.4.15 │
IP Address = 10.1.15.192 on eth0 │
Operating System = CentOS release 6.5 (Final) >< │
Kernel Version = 2.6.32-431.20.3.el6.x86_64 - 64 Bit

What can I change to improve this? It seems that I have two different slow things that only have the pbx in common.

thanks in advance,
jerry

 

[atsak]

Any chance there's a duplex mismatch setting on the NIC connecting the VM or something like that? What kind of network switches?

 

[sko001]

Did you check the settings for the jitter buffer?

 

[JerryS]

Hi,

Jitter buffer is off. The server is connected directly to a brocade fastiron 48 port switch and shows no other problems. Still searching for the lag.

 

[sko001]

Try connecting the server directly in the internet router.