Asterisk 1.6.2 with Konference audio Latency Grows Over time

I am currently facing similar kind of problem of Latency growing over Time

I am using currently CentOs as opertating system and Asterisk® Open Source PBX asterisk and for conferencing I am using app_konference 2.2 and pjsua1.8 as sip user agent or sip client on hard phones on local network (private network)
what happens is when any Konference() is created it works fine with latenct 250-300ms for some hour or hour and half after that there is 1-3 seconds of delay in audio
one more unusual thing I found during testing is asterisk is taking too much of memory which is continuously growing that is when I do top I get

 PID 	      USER      PR  NI  VIRT  RES  SHR S %CPU  %MEM     TIME+       COMMAND
 3161        root      15   0  293m  24m  13m S  0.7   0.6     152:40.32   Xorg 
 15977      root      15   0  174m 160m 4456 S  0.7   4.2     0:08.62     asterisk
 3406        root      15   0  448m 402m 9.9m S  0.3  10.5     19:48.41    gnome-terminal 

and sip call which I was testing is continuous type of call so I kept it for 24 hrs next day I found that asterisk is taking 63% memory as follows

 PID 	USER      PR  NI  VIRT    RES   SHR  S  %CPU  %MEM     TIME+       COMMAND
15977  root     15   0  2449m   2.4g  4540 S  0.7   63.7     2:17.68     asterisk  

my asterisk settings are as follows

node2*CLI> core show settings 
PBX Core settings
  Maximum calls:               Not set
  Maximum open file handles:   Not set
  Verbosity:                   3
  Debug level:                 0
  Maximum load average:        0.000000
  Minimum free memory:         0 MB
  Startup time:                10:15:21
  Last reload time:            10:15:21
  System:                      Linux/2.6.18-274.7.1.el5PAE built by root on i686 2012-01-13 07:07:46 UTC
  System name:                 
  Entity ID:                   84:2b:2b:a1:ff:1f
  Default language:            en
  Language prefix:             Enabled
  User name and group:         /
  Executable includes:         Disabled
  Transcode via SLIN:          Enabled
  Internal timing:             Disabled
  Transmit silence during rec: Disabled

* Subsystems
  Manager (AMI):               Enabled
  Web Manager (AMI/HTTP):      Enabled
  Call data records:           Enabled
  Realtime Architecture (ARA): Disabled

* Directories
  Configuration file:          
  Configuration directory:     /etc/asterisk
  Module directory:            /usr/lib/asterisk/modules
  Spool directory:             /var/spool/asterisk
  Log directory:               /var/log/asterisk

actually I thought that this problem of audio latency is due to app_konference so I upgraded my app_Konference module version to 2.2 from 1.3 so what ever is causing this problem is somewhere related to asterisk than app_konference module thats what I think.

similar to what fizmod mentioned New calls(channels ) to join do not experience this latency. If a participant drops off and dials back in, the latency disappears for that caller.

as time grows delay in audio grows is there any thing I am doing wrong or configuring wrong please guide me to find out root cause of this problem.

any help is appreciated

Thanks & Regards


I’m afraid you’re probably not going to get too much help with your problem. You’re running both an unsupported version of Asterisk and a third-party module.

You could try Asterisk 10 or Asterisk 11 with app_confbridge, which I believe offers similar functionality to app_konference, though without knowing your specifics or being totally familiar with app_konference I can’t say with any surety, and see if there’s a latency problem then.

hi malcolmd,
Thanks for the help actually we are using this since last two years and from your suggestion I decided to upgrade to asterisk which is end of life build of 1.6.2 branch so and app_konference 2.2 which is latest I guess and testing now hope this solves the issue

Thanks a lot
Pranav :smiley:

hi all
upgrade to asterisk which is end of life build of 1.6.2 branch so and app_konference 2.2

still do not solve my problem but the unusual memory usage by asterisk is stopped .

Please can any one tell me tweaking what modules of asterisk at build time may affect the app_konference performance. because app_konfeerence do not have any configuration or menu select at build time

any help is appreciated

Thanks in advance


I am using dahdi-linux-complete- on some post I read that due to dahdi_dummy clock support also some times causes delay or latency in audio If no hardware connected.
But actually I have Openvox a400p hardware connected with pci slot to my server.
is there any thing in dahdi that is causing this audio latency problem?

any help is appreciated.
Thanks in advance