[Asterisk-Users] audio delay in meetme conference using ztdummy

[Davin O'Neill]

I have Asterisk running on a Linux 2.4.x box with ztdummy.  Once I did a
modprobe on ztdummy I was able to enter into a conference room using my
softphone clients.  I'm using SJphone and Firefly.  I have noticed a
significant delay (1 to 3 seconds) while talking within the conference room.
I have tried both clients, SIP and IAX protocols and various codecs.  I have
also tried it from different host machine.  They are all on the same LAN, so
that shouldn't be an issue.  I can call a client directly with SIP or IAX
and have clear, timely audio.  I have also done echo tests (dialing 600)
through Asterisk and that works fine too.  The delay only occurs within the
conference room.  I'm wondering if I just need to purchase one of the zaptel
cards.  I would appreciate any thoughts or suggestions.
 
Thanks!
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