asterisk users - Mar 2005 - audio delay in meetme conference using ztdummy

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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Davin O'Neill wrote:
> 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!
try adding "q" flag to meetme app ...

On Tue, 22 Mar 2005, Davin O'Neill wrote:
> 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.
See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003599 on 
the bug tracker.

The problem with delay on VoIP channels is known. There is disagreement on 
how to fix the problem though. Can you try cvs head and see if the problem 
persists. If it does then it may be a good idea to add a comment to the 
bug report so the problem gets resolved.

Peter