[Asterisk-Users] bad pstn audio coming from old * processes

[Steven Critchfield]

On Mon, 2004-01-12 at 07:09, Thilo Salmon wrote:
> Hi everybody,
> 
> I use asterisk as a PSTN gateway and found that the audio quality 
> would significantly decrease when asterisk has been busy for a while.
> Using tcpdump I can see approx. 25% fewer rtp packtes coming from the
> asterisk box than from my sip client. The lost packets seem to follow an
> even pattern. About every 4th packet seems to be missing. The audio
> sounds like gargling sounds are added to it. Audio to the PSTN however
> is not affected.
> 
> This behaviour can be observed on two different UP boxes - one with two
> E400p and another one with two TE410p in it. Both boxes experienced
> heavy traffic before showing this behaviour (close two 240 concurrent
> channels for about 24 hours). 
> 
> Trying to narrow down the problem I found the following: 
> 
> All calls routed to or from the PSTN showed this behaviour. The problem
> would show no matter whether IAX2 or SIP channels were used on the VOIP
> side. The direction of a call would not affect the problem. However,
> calling an IAX2 target from a SIP UA through
> the same asterisk box whould showed crystal clear audio both ways. The
> problem also exists independent of the sip ua (Grandstream, Zultys and
> X-ten were affected).
> 
> Interesting enough restarting asterisk would reliably solve this issue.
> Has anybody experienced something like this before? Or even better:
> knows how to fix it?

Sounds like it is a PSTN problem if VoIP to VoIP is clear even when
translated from IAX to SIP. Check with zttool when this is happening to
see if there are errors or irq misses showing up. 
-- 
Steven Critchfield <critch at basesys.com>