[asterisk-users] spa942 and asterisk 1.2

nivlekch nivlekch at aim.com
Tue Jan 16 07:10:12 MST 2007


currently using 1.2.14 and zaptel 1.2.12
i'm using mfc/r2 so i can't move to 1.4 with sip jitter control and 
improved jitter control in zaptel 1.4.

my problem is  excessive jitter  using linksys  spa942.
when i set canreinvite=no, which forces rtp to pass through *, quality 
is horrible. clicking sounds, pauses, etc. but when omitted or 
canreinvite=yes, sip to sip calls are ok. now, the problem comes to zap 
calls, i have a te110p using unicall mfc/r2, since rtp passes through *, 
quality is again awful.

just wanted to ask the list whether somebody out there had experience or 
had used linksys spa942 before. did you experience this phenomenon? how 
can you go around the zaptel jitter? obviously i tried using 
jitterbuffer=40 in zaptel.conf and/or even in unicall.conf to no avail.
 


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