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 126.96.36.199 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 node2*CLI> PBX Core settings ----------------- Version: 188.8.131.52 Build Options: LOADABLE_MODULES, DEBUG_CHANNEL_LOCKS, BUSYDETECT_TONEONLY, BUSYDETECT_DEBUG 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 node2*CLI>
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