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

[Tony Mountifield]

I've been doing some experiments with app_meetme, and only have SIP phones
here to play with. I have been finding issues with audio delay that I
think may be to do with the use of pseudo channels to conference non-Zap
channels.

The easiest way to demonstrate it is first of all to make a pair of calls
to an extension that calls MeetMe(2222|). Speaking into both phones and
listening to them both gives an audio delay of about 300-400ms.

Then repeat the experiment using MeetMe(2222|q). This time the audio comes
back almost instantaneously.

I am suspecting that the problem is something to do with the conf_play()
of the enter and leave sounds. My guess is that by writing that raw data
into the pseudo device fd, it causes a backlog that never drains, because
the device is only getting emptied at the same rate as the conference is
filling it.

The delay does seem to be of approximately the same length as the enter
sound.

I don't know whether the same issue applies to direct Zap channels or not.

Any comments?

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