[Asterisk-Dev] Re: Audio delay in MeetMe using SIP when not 'q' mode

[Tony Mountifield]

Andrew Lindh <asterisk at ntplx.net> wrote:
> 
> It would be better to overlay/merge the audio rather than just skip it.
> Someone could be saying something important at that moment...

I take your point, but the enter sound is generated by the caller just
entering, and it is their audio that would be skipped. They are unlikely
to be saying anything important until they have finished joining the
conference.

Having said that, I've now tried out the idea, and although it improves
things, it doesn't appear to solve it completely, so there may be
something else going on too.

> The problem has been around for a while. Open a bug report on it and
> post your patch there. That way there is a better record of the updates.

Yes, that's this morning's job. :-)

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org