we are facing the crackle issue during call in asterisk 13 by using sip trunk line service and some time in PRI ,we have set the G729 codec in sip.conf and do all troubleshooting but problem is still there,some time sound quality comes good and sometime is very bad, if anyone have any proper solution then share.we are using 1 mbps sip trunkline service so with codec g729 how much concurrent call can be in live ? 1024/31= 33 is right or wrong???
I assume you meant 1 Mb/s not 1 mb/s.
Physical layer bit rates are normally specified with decimal multipliers.
31 should be 31.2 Kb/s.
That gives 31 channels, assuming there is absolutely no other traffic. I’d probably de-rate that by at least 20%., to about 25 channels.
This assumes that there are no bottlenecks a the other end of the wire. In the modern internet is common to run with very large backlogs in buffers, resulting in significant jitter.
If this is s a private trunk, I believe IAX has options for multiplexing multiple streams in one packet, which can reduce overheads, significantly.