[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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