[Asterisk-Users] capacity testing

Thu Jan 15 18:21:29 MST 2004

Sorry for the malformed mail. My responses are marked with '***' below.

jesse
======
Hi,

I am a newbie in Asterisk as well, intending to use it in a similar way as
you are, communicating with AS5300 as well as other gateways including
MAXTNT.

I have had similar, but yet different experiences than yours.

1. Asterisk does crash with the number of calls, but in my case, about or
less than 20 calls, then I would get either a Segmentation Error and then
crashed OR it would just crash saying "Disconnected from Asterisk server"
all of a sudden.
*** The crashes I experienced were fairly transparent. When I had the console (asterisk -r) running, I saw the 'Disconnected' message you mention.

2. I am using Pentium Xeon chip and hence more powerful than yours with 512M
RAM, my CPU usage has always been low, however, I have not had a chance to
look at the CPU usage just before crashing, but all the time that I was
looking, it has been low. Rather the MEMORY has always remained high at 450M
usage even with no calls. This is a different experience as compared to
yours.
*** A Xeon of the same speed (800mhz in my case) should certainly perform better - lower, I don't know. I find it a little odd that you experienced basically the opposite of my testing. What version are you running?

3. I have also noticed that with more calls, and after a certain random
period of time, any H323 calls going into the Asterisk would fail, my AS5300
and MAXT TNT would get their calls all rejected from Asterisk. However,
Asterisk was still running at the time and I could actually call in and out
the zap interface and outbound H323 from Asterisk was not a problem. It
seems that something got hung with H323, causing inbound H323 calls into
Asterisk to all fail. In this situation, I would have to stop the Asterisk
and rerun it to fix the problem.
*** Interesting - I have not experienced that (yet...).

4. I have not tried the 0.7.0 version, but with existing version, I am not
getting reliable and stable system, nothing close to Cisco and Lucent which
are rock solid. However, I really love the power and the features of
Asterisk, and I remain in good faith to see improvements.

Any associate out there who can shed some lights into this? I am rather
curious as to why I seem to be using up all memory although I am not running
any unnecessary processes, or should I actually disable all modules, other
than really necessary ones to support VOIP?

*** Since you and I are working in what sounds to be a familiar environment, maybe we should communicate about our test scenarios, etc off list to both help each other and see if we can find some similarities to share with others.

Thanks !

Tom

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Jesse
Peterson
Sent: Thursday, January 15, 2004 2:40 PM
To: Asterisk-Users (E-mail)
Subject: [Asterisk-Users] capacity testing

Hello all. I'm new to asterisk and have been using and testing it for about
a week now. My initial hope has been to use it as a sip<->h323 gateway to
tie SIP & H323 based ip phones together with my Cisco AS5300 and Lucent
MaxTNT/MVAM networks.

I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800
with 256megs RAM. I have tried a couple CVS version from the past week
(maybe 01/09/04 and 01/14/04) and have not been able to get them to work
semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported
those ok. Primarily test cases have involved sending ip phone calls via SIP
to Asterisk and having Asterisk route the calls using h323 via a gatekeeper
to my TNT network which then sends it out the PSTN... and the opposite path,
PSTN->TNT->Asterisk->SIP Phone. Another test has been sending a call from a
AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked
very well with the voice quality being excellent (actually better than a
SIP->ISDN T1 hardware solution we've been working with - audiocodes mediant
2k for those interested). This is the test case I describe below as it was
the one the allowed me to load Asterisk up with the most calls.

Anyway, I know that what I'm doing is not exactly the intended primary use
of Asterisk. That said, here's what I found.

Voice quality was very good until I had approx. 25 calls up. At that point
there were intermittent issues with garbled voice, a little echo, etc. When
it reached a little over 30 calls, Asterisk just died (oops).
During the test, I was trying to keep an eye on proc. & memory util. Memory
never seemed to be an issue - even right before the crash the Asterisk
process was not using more than 20 - 25MB.
Processor utilization was interesting to watch though. I couldn't make any
direct/firm correlation, but it seemed like my spikes were coming when
Asterisk was doing call setup. Even up to about 25 calls, utilization didn't
spike to more the 25% for long, and with ~25 calls seemed to 'idle' around
15%. Above the 25 (when also started noticing voice quality issues), the
proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%.
Then it went to 99% for a moment, voice quality was horrible if you could
hear anything, and Asterisk crashed.

I did not find anything in the logs to inidicate any problems, though I've
found that to be the case pretty much everytime Asterisk crashes.

I saw a list thread in which a developer asked for some gdb output... in it,
he said this:
> Run asterisk with "-vvvcg".
> Do your test (core file generated).
> Run "gdb /usr/sbin/asterisk <core_filename>"
>  From within gdb run "bt" and send me the output
> of it.

if it is of use, here it is (from asterisk v.0.5.0)
-----------------------------
(gdb) bt
#0  ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at frame.c:72
#1  0x41eb00b1 in oh323_write (c=0x8214488, f=0x5de5c4a8) at
chan_oh323.c:1504
#2  0x0805884f in ast_write (chan=0x8214488, fr=0x5de5c4a8) at
channel.c:1385
#3  0x0805afa1 in ast_channel_bridge (c0=0x5de5c4a8, c1=0x0, flags=0,
fo=0x6ef20e50, rc=0x6ef20e54) at channel.c:2262
#4  0x418bdd7a in ast_bridge_call (chan=0x5de5ed98, peer=0x8214488,
allowredirect_in=0, allowredirect_out=0, allowdisconnect=0) at
res_parking.c:224
#5  0x41d6bfeb in dial_exec (chan=0x5de5ed98, data=0x41d6d19b) at
app_dial.c:668
#6  0x08061a5a in pbx_exec (c=0x5de5ed98, app=0x80f0f98, data=0x6ef216e8,
newstack=1) at pbx.c:396
#7  0x08068c61 in pbx_extension_helper (c=0x5de5ed98, context=0x5de5eeec
"longdistance", exten=0x8214488 "H323:8257", priority=2,
    callerid=0x5de10048 "\"Jesse Peterson\" <2474766>", action=1104606132)
at pbx.c:1150
#8  0x0806392c in ast_pbx_run (c=0x41d6f3b4) at pbx.c:1634
#9  0x08069321 in pbx_thread (data=0x84a5038) at pbx.c:1855
#10 0x40026484 in start_thread () from /lib/tls/libpthread.so.0
-----------------------------

If anyone has tried something like this or has any comments, I'd be
interested in hearing from them.

jesse

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