Asterisk + Konference audio Latency Grows 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

as I was adviced that is unsupported version and also latest bbuild might have fixes for such problems so I upgraded to asterisk and app_konference 2.2 but still facing same problem
can anyone please guide me to find cause of this problem

Thanks in advance

app_konference appears to be a third party contributed application, supported at … rum/924974

hi david

I know app_koference is third party contributed application but I don’t know in first place this problem is caused by asterisk or by app_konference . If anyone can guide me to conclusion that app_konference is the culprit I might think of changing the module itself to some thing else.

Also if the app_konference is third party which conferencing is supported by asterisk and which one should I use If asterisk does not support all the features supported by app_konference?. also which conferencing is widely used and supported?.

and also I thought someone on asterisk forum might have faced this kind of problem and could guide me to cause of problem.

Thanks & Regards

hi I am re describing the issue as below

I’m using Asterisk

When I make a call from a SIP phone to a conference on the Asterisk server, the latency (“lag”) starts off very low
– a fraction of a second. But as several minutes of time goes by, the lag increases. After, say, 15 minutes, the lag will reach a couple of seconds, making conference calls unusable.

This does not happen on pure SIP-to-SIP calls, even when Asterisk is handling the RTP media.

If I hang up and immediately call back in (even to the same conference), the lag is reset to almost zero. If I put the call on “hold” and take it off hold, the lag is also gone.

During testing, I’ve found that this may be related to the Asterisk app on the server not getting all the CPU time it wants.

If I suspend the Asterisk process for two seconds and then resume it, the sound stops for two seconds, as I’d expect. But the conference calls that were active are then lagged by two seconds until they hang up (or get put on hold), even after Asterisk is resumed and getting all the CPU time it needs.

The conference application I am using is AppConference 2.2.

Any ideas on how to fix this? Do other people see the same thing happening? As I said, the gradual increase in lag over a long period of time makes long conferences unusable, unfortunately.

Please help…it’s URGENT!!!

Thanks in advance