Troubleshooting voice problems


We have Asterisk installed on Ubuntu at first ISP network behind ISP’s gateway nat with forwarded ports.
We have our office with router on Debian at Second ISP behind NAT.
Almost all users use 3CX Phone for making outgoing calls.
The problem is that I can have 2 managers sitting near by with no voice problems and 2 more managers near them who have voice loosing.

On asterisk i see this:

It’s at night time, when almost no one is in office, latency reaches 130ms, some why…

The traceroutes from both sides:

When I ping from host behind office router asterisk’s gateway with 1400 packet size i see latency not more then 18-19ms.

I’ve started snmp monitoring with cacti of Ubuntu server where Asterisk is, but traffic, cpu and memory is OK.

I’ve used cisco ip sla with jitter-udp on other network, it really could help, but I really don’t know how to measure udp jitter between at least Debian gateway and Ubuntu ASterisk server.

I can just assume that I need to lanuch some application from office to asterisk, which could measure packet delay and packet jutter, or laucnh some kind of RTP and SIP sniffer on asterisk which will show this information…

Can you please help me with advice?

Asterisk will use RTCP, if the peer supports it, to measure and exchange this information. Use the CLI rtcp commands to recover the results.

Thanks for the answer!

I see rtcp set debug on\off\ip and rtcp set stats on\off

Where can i see debug and stats?

Do a packet capture on all points IE gateway and servers then compare.

details on how to do it is in a link in my Sig

On the CLI and/or the log files, depending on which categories you have enabled in logger.conf.

thanks for the answers!)

is there any safe way to monitor asterisk load?

i’m thinking about just couple of parameters:

  • number of logined sip accounts at a time
  • number of active channels per at a time
  • active trunks
  • delay in seconds for connecting to asterisk and listening sound file
  • delay in seconds for connecting to asterisk from one sip1 to sip2 and play sound on sip2

The delays due to Asterisk will be negligible, so you will not be able to monitor them from within Asterisk.

is there any best practices?

I can monitor outside the asterisk, i can monitor it on the asterisk.

I’ve just created scripts for getting number of active channels, number of active calls, output to files channel.num and call.num and finding maximum in them but it actually isn,t looking good

It would be great if there weve at least a few graphics…