We are facing high latency & voice breakage issues now days. i would like to know how can i overcome with voice breakage issue as our work is voice critical .
We use to have RTT 160 - 220ms but now its between 250 - 600 or sometimes 800ms.
Current asterisk version - Asterisk 11.15.0
[general] bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) permit=0.0.0.0/0.0.0.0 disallow=all allow=g729:30 allow=alaw:30 allow=ulaw:30 context=default notifyringing=yes tos_sip=cs3 tos_audio=ef cos_sip=3 cos_audio=5 jbenable=yes jbforce=no jbmaxsize=600 jbresyncthreshold=2000 jbimpl=adaptive jbtargetextra=60 autoframing=yes dtmfmode=rfc2833 canreinvite=yes