asterisk users - Jan 2004 - bad pstn audio coming from old * processes

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?

Thilo

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@basesys.com>

Steven,
> 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. 
You are right. I can see a total of 17 missed interrupts for this span
(since I last reloaded the drivers I think, which has been a week or two
now). However, while placing a call this number does not go up nor do
any other errors show up. 

What I forgot to mention: I think it is unlikely to be a problem with my
telco, since I am connected to two switches with dozens of E1s - all
showing the same behaviour. What really puzzles me is the fact that
simply restarting asterisk helps this problem.

Thilo