[Asterisk-Dev] Help Debugging Dropped Call Audio - Add'l Info

[Kevin P. Fleming]

Matt Roth wrote:

> All of this leads me to believe that the problem dwells somewhere in the
> code responsible for bridging the channels.  Does anyone have any ideas
> as to the specific cause or the direction that I should go to further
> pursue it?  I've heard that Asterisk's locking scheme is questionable,
> but I don't know if that would apply to this scenario.

That code lives in rtp.c, and is very simple. It also handles many 
thousands (if not millions) of calls every day all over the world, and 
if it had a major flaw like this we'd surely have heard about it before.

I really can't help much more in this thread without interfering with 
the tech support ticket/case that it already open... but I'll try to get 
some more information tomorrow to see what else you have tested since 
the last time I talked to tech support about it.