Asterisk + Konference audio Latency

  • Pranav

    Pranav - 2012-11-27

    am using currently CentOs as opertating system and Asterisk(R) 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  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

  • John J Hass

    John J Hass - 2013-02-16

    1.8 and 1.6 were never really developed we will be releasing 10/11 support next week

  • James

    James - 2013-05-17

    can it used under 1.8 for reslove the latence bug?

    Last edit: James 2013-05-17

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