Hi,
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
My sip.conf
[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
Why do you have a jitter buffer enabled on what appears to be a SIP channel? The jitter buffers should be in the gateways to analogue and circuit switched parts of the network (i.e. in the phone itself and on the PSTN or ISDN connection.
Large single hop round trip times indicate network problems, possibly including buffer bloat. You may be able to prioritise RTP in the network, either purely by network settings or by enabling differentiated service in the network and setting expedited flow in the Asterisk class of service.
Either you are using an obsolete version of Asterisk, or your configuration contains long deprecated and possibly obsolete settings.
Really getting direct media to work will remove Asterisk from the equation.