[asterisk-dev] latency bursts (qualify=yes)

[fadey]

Hi, everyone

I'm curently debugging an issue in a small cable network (about 150
SIP-enabled EMTAs) with asterisk as their sip proxy. I have qualify=yes
in sip.conf. The problem is sporadic latency bursts as seen with "sip
show peers". Normaly EMTAs would show 30-100 ms. But once every 2-3
minutes you'd see a random EMTA showing 1000-3000 ms (I had to change
qualify=yes to qualify=90000 to actualy see it). I've seen this behavior
both with asterisk 1.2 and 1.4.
First I though it's a networking issue. To check it I wrote a simple
tester, which would send INVITE packets to EMTAs and note the responce
time. And it shows stable 30-100 ms. So I'm pretty sure it is the
asterisk that causes those bursts.
Has anyone seen this behavior before? Does it affect voice calls as well
(I mean, are there the same latency bursts in RTP voice traffic? I
couldn't find a way to measure that)?
Thanks in advance