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

[Peter Svensson]

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