am using currently CentOs as opertating system and Asterisk(R) 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
as I was adviced that 188.8.131.52 is unsupported version and also latest bbuild might have fixes for such problems so I upgraded to asterisk 184.108.40.206 and app_konference 2.2 but still facing same problem
can anyone please guide me to find cause of this problem also
after entering cli command
cli> konference end <conferencename >
conference is ended and again started immediately
also I am not able to get mailing list address please guide me.
Thanks & Regards
1.8 and 1.6 were never really developed we will be releasing 10/11 support next week
can it used under 1.8 for reslove the latence bug?
Log in to post a comment.
Sign up for the SourceForge newsletter:
You seem to have CSS turned off.
Please don't fill out this field.