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

[Vincent Nonnenmacher]

>¤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
>

we have seen the same problem (smaller number of 
phones ‰ 50 on two sites) and same conclusion 
(2000 to 3000ms and becoming unreacble unless 
qualify is more than 5000).

Still unable to find/qualify audible rtp problems 
at those occurence, we are working on making some 
Sipp test cases in order to find a way to push 
asterisk in order to reproduce it, but not seen 
them yet.

We came to the same conclusion about the need of 
a network probe that could be integrated inside 
asterisk, based on a sort of 'time sloted' UDP 
echo tester that could report (CLI) from * to any 
networked counterpart infos about packet loss, 
jitter, latencies and al.

So to be able (without any rtp complication) to 
check the network transport as seen be asterisk 
and how timing could affect the * internal frames 
handling.

Anyonw working on that kind of *_app ?


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Vincent Nonnenmacher
AgentSpace
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