[Asterisk-Users] Audio pausing over IAX trunk

[Florian Overkamp]

Hi Steve, 

> -----Original Message-----
> > I am having a problem with periodic breaks in audio over an 
> IAX trunk. 
> > The interruption only happens in one direction, and (I think) only 
> > with clients built on the open source libiax.
> >
> > Codec is irrelevant, and jitterbuffer on/off seems to make no 
> > difference either. The pause happens every few seconds, and 
> is regular.

> Not unless you can describe the problem more clearly.
> 
> Which direction does this happen in, what exactly are these clients 
> you're talking about, and what is does the network look like 
> between the 
> endpoints.

Okay, in my scenario it's like this:

SIP or MGCP phone (mixed env.) -> Asterisk box -> IAX -> Asterisk box ->
PSTN or other Asterisk box

We notice users complaining of the fact that the remote end (PSTN)
complained about audio drops, while the local user keeps hearing everything.
I am not entirely sure if it is just that direction, because I hear
noticeable crackles during the call from my (user) end too.

This appears to happen especially when the asterisk boxes involved have a
few calls happening, when its nice and quiet on the box, things seem ok.
This kind of thing is not or hardly noticable when calling yourself, which
makes diagnosis difficult.

I've discussed this with other people on the list, and we notice the
following: IP links are _not_ congested and latency is very stable, so we
are not looking at a network issue. Others have observed that changing the
protocol from IAX2 to SIP is a good workaround. I have not yet been able to
confirm this because we are tied to Asterisk-stable which does not yet have
a very useable SIP dialling format. It's very hard to get a good handle on
this issue, because it pretty much requires a multihomed production box to
work with :-(

Florian