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Jesse Peterson

2004-Jan-15 12:39 UTC

[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

Alastair Maw

2004-Jan-15 15:17 UTC

head link

[Asterisk-Users] capacity testing

On 15/01/04 19:39, Jesse Peterson wrote:
> #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
Do you experience the same problems when you use the other (bundled) 
h323 driver? (asterisk/channels/h323/README for instructions)

Alastair

T. Chan

2004-Jan-15 15:31 UTC

head link

[Asterisk-Users] capacity testing

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.

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.

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.

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?

Thanks !

Tom

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@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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Chris Albertson

2004-Jan-15 16:29 UTC

head link

[Asterisk-Users] capacity testing

THat's not bad 20 calls through a 800Mhz P3.  I new 3Ghz P4
could likely handle 60 then. Not bad.

But don't beleive "top".  First off if acverages.  Think for
a minute.  We all kow a CPU can never by "20% in use" it is either
in an idle loop (at 0%) or doing real work (100%) it can't be
in an in-between state.  

I think with Asterisk what matteris is the probibility that when
a packet comes in the CPU is idle and available to process it.
When "top" says "20%" that means that is only an 80% chance
the
CPU is free.  Looks like 80 or 85 is about the braking point.

Same applies to bandwidth.  If a packets needs to go out, it
needs to go out NOW not some time later if the sound quality
is to be OK.  So you look at the "probibility of collision" not
just the available bandwidth.  If your bandwidth is half used
by, say web surfers, then half of your VOIP packets will be
delayed.  Jitter buffer can help, to a point.

Back to CPU utilization:  30% utiliation means it is not available
to handle a packet 30% of the time and that hansling must be
queued up or delayed.




--- Jesse Peterson <jesse@strata-com.com> wrote:> 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
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

====Chris Albertson
  Home:   310-376-1029  chrisalbertson90278@yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson@aero.org
  KG6OMK

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Jesse Peterson

2004-Jan-15 18:15 UTC

head link

[Asterisk-Users] capacity testing

I did initially, but I was having problems (possibly just in thinking it
through) getting the provided h323 driver to either
a) register as a gateway with my gatekeeper - that just does not seem to be and
option (please correct me if I'm wrong!!!)
or
b) setup a 'variable' extension (yes, extensions.conf) that would allow
me to route any number to it.
 
jesse
 
-----Original Message----- 
From: Alastair Maw [mailto:asterisk@almaw.com] 
Sent: Thu 1/15/2004 5:17 PM 
To: asterisk-users@lists.digium.com 
Cc: 
Subject: Re: [Asterisk-Users] capacity testing



	On 15/01/04 19:39, Jesse Peterson wrote:
	
	> #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
	
	Do you experience the same problems when you use the other (bundled)
	h323 driver? (asterisk/channels/h323/README for instructions)
	
	Alastair
	_______________________________________________
	Asterisk-Users mailing list
	Asterisk-Users@lists.digium.com
	http://lists.digium.com/mailman/listinfo/asterisk-users
	To UNSUBSCRIBE or update options visit:
	   http://lists.digium.com/mailman/listinfo/asterisk-users
	

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Jesse Peterson

2004-Jan-15 18:21 UTC

head link

[Asterisk-Users] capacity testing

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@lists.digium.com
[mailto:asterisk-users-admin@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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T. Chan

2004-Jan-15 19:02 UTC

head link

[Asterisk-Users] RE: PID

Hi, all !

I have a fast question, I am running a few Asterisk systems, but I just
noticed one thing quite peculiar. After I started "safe_asterisk", and
when
I ran PS or TOP, I could see 1 PID "safe_asterisk" and almost 10 PIDs
"asterisk -vvvg -c" even when there was no call. However, for the
other
couple, I started "safe_asterisk" and when I ran PS or TOP, I could
see 1
PID "safe_asterisk" and only 1 PID "asterisk -vvvg -c", they
are all with
Pentium Xeon chip and 512M RAM, no difference in Hardware, and all running
the same version of Asterisk on Redhat 7.3.

Does anyone have any idea why there is a difference please? The reason that
it is important as well is because each "asterisk -vvvg -c" is taking
up
certain memory and with 10 (more when there are calls) or more of these, I
am running into memory problem. However, in the other case, no matter how
many calls I have, I only see 1 PID of "asterisk -vvvg -c" and seems
that I
have less of a memory problem.

Any feedback to help solve the mystery? Thanks

TOm
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Jeremy McNamara

2004-Jan-15 20:11 UTC

head link

[Asterisk-Users] capacity testing

Jesse Peterson wrote:
>I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800
with 256megs RAM.
>
CVS UPDATE!   That code is hardcore old.



Jeremy McNamara

T. Chan

2004-Jan-15 20:53 UTC

head link

[Asterisk-Users] capacity testing

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Jesse Peterson

2004-Jan-15 21:59 UTC

head link

[Asterisk-Users] capacity testing

I know, but as I mentioned in the inital post, I haven't been able to get
the last 2 cvs versions I've pulled to run stable enough to test.
I've seen a 0.7.0 version number mentioned. Is there newer, mostly stable
version of code I should try that just hasn't been officially released?
 
jesse

	-----Original Message----- 
	From: Jeremy McNamara [mailto:jj@nufone.net] 
	Sent: Thu 1/15/2004 10:11 PM 
	To: asterisk-users@lists.digium.com 
	Cc: 
	Subject: Re: [Asterisk-Users] capacity testing
	
	

	Jesse Peterson wrote:
	
	>I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800
with 256megs RAM.
	>
	
	CVS UPDATE!   That code is hardcore old.
	
	
	
	Jeremy McNamara
	
	
	
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Jesse Peterson

2004-Jan-16 10:31 UTC

head link

[Asterisk-Users] capacity testing

1) Yes, I did get that. I've never seen a segmentation fault message, but
that should be b/c I've been running the process in the background since it
is obviously seg-faulting. I believe you are also correct that most people are
not trying to put the load on it that we are.
 
2) I always see 'safe_asterisk' and 'asterisk -vvvg' running. my
monitoring was always done with top, but I've checked w/ ps a couple times
and I believe only ever see 1 of each of those processes. I may have to do some
tests again to double check that. My CPU problems did not come until the last 10
- 30 seconds before asterisk crashed. This is still odd that our memory &
processor observations are opposite... the next thing I'm going to try is a
dual xeon pIII 800 or 1ghz machine to see what happens.
 
3) I'm running oh323. It was the one I could get to register w/ my
gatekeeper as a gateway - that made it much easier for me to do call routing on
both sides. I have also noticed some inconsistencies in the call flows like you
mention, but haven't taken the time yet to pinpoint exactly what and when
they are happening.

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of T. Chan
Sent: Thursday, January 15, 2004 22:54
To: asterisk-users@lists.digium.com
Cc: Alan Chan
Subject: RE: [Asterisk-Users] capacity testing


Hi all, and Jesse
 
1. So, you did get the experience of crashing all of a sudden with the
"Disconnected from Asterisk server" error message. I got both this and
the segmentation error when crashing. I am running the version of asterisk,
libpri and zaptel updated to about 5 days ago, but I have had tested Asterisk
for more than a month already and needless to say I have had this experience
since Day 1, meaning it has always been a problem even in the previous
revisions. Henceforth, I feel that it is an intrinsic Asterisk problem, rather
than just the problem with specific versions / revisions. I have posted this
problem a few times before, I feel that this is a major problem but
surprisingly, I was not getting any feedback at all. I have this feeling that
more than 90% of the Asterisk community is using the system for PBX application
rather than VOIP, may be, just may be, Asterisk has not been tested with a good
number of simultaneous calls.
2. I am using Xeon 2.6G chip, much more powerful than yours, I have not got any
problem with CPU usage, at least not during the time that I was watching. The
thing is when I start 'safe_asterisk' , I could see when doing a PID, 1
"safe_asterisk" PID session and at least 10 (or more especially when
there are more calls) "asterisk -vvvg -c" PID session. Each session
takes up about 18M to 20M RAM, when that is why I am seeing all very high memory
usage. How many sessions of Asterick do you see running after you loaded it?
3. Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's
and have had this inbound H323 problem. I tried OH323 (Michael) as well, but for
some reasons, I am getting this false connect signal, that is, I made an
outbound H323 call to a CiscoAS5300 for example, I heard the ring and
immediately on my "Asterisk", it showed call answered when it was
still ringing. Do you have that experience?? What setting you have if you do not
have that experience?
4. Lets talk off list at utitc@hotmail.com.
 
Thanks
 
Tom

-----Original Message-----
From: Jesse Peterson [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of
Jesse Peterson
Sent: Thursday, January 15, 2004 8:21 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] capacity testing


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@lists.digium.com
[ mailto:asterisk-users-admin@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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asterisk@lists.styx.org

2004-Jan-16 11:33 UTC

head link

[Asterisk-Users] RE: PID

On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:> 
> Does anyone have any idea why there is a difference please? The reason that
> it is important as well is because each "asterisk -vvvg -c" is
taking up
> certain memory and with 10 (more when there are calls) or more of these, I
> am running into memory problem. 
The fact that you see multiple processes is symptomatic
of a wonky threading implementation -- all threads should
belong to the same process. The exception to this is
chan_h323 which actually fork(2)s a new process.

Also, as threads share memory space there is no extra
usage with extra threads (except for the overhead of
keeping the thread state itself, which is minimal).

-w
-- 
/~\  The ASCII Ribbon Campaign
\ /    No HTML/RTF in email
 X     No Word docs in email
/ \  Respect for open standards

T. Chan

2004-Jan-16 15:12 UTC

head link

[Asterisk-Users] RE: PID

Dear All,

So are you saying that I should see 1 PID for "safe_asterisk" and many
PIDs
for "asterisk -vvvg -c" or just 1 PID for "asterisk -vvvg
-c", the problem
is I am seeing alot of PIDs for "asterisk -vvvg -c" on a couple of my
systems and only 1 PID for "asterisk -vvvg -c" for the other couple,
which
way is correct and how do I rectify that? When I said seeing multiple PIDs,
I mean I actually see like 20 different PID numbers (the second column when
I do a ps), not sure if that means 20 different threads or 20 PIDs. How do I
take care of wonky threading implementation as you suggested?

THanks

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of
asterisk@lists.styx.org
Sent: Friday, January 16, 2004 1:33 PM
To: asterisk-users@lists.digium.com
Cc: Alan Chan
Subject: Re: [Asterisk-Users] RE: PID


On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote:>
> Does anyone have any idea why there is a difference please? The reason
that> it is important as well is because each "asterisk -vvvg -c" is
taking up
> certain memory and with 10 (more when there are calls) or more of these, I
> am running into memory problem.
The fact that you see multiple processes is symptomatic
of a wonky threading implementation -- all threads should
belong to the same process. The exception to this is
chan_h323 which actually fork(2)s a new process.

Also, as threads share memory space there is no extra
usage with extra threads (except for the overhead of
keeping the thread state itself, which is minimal).

-w
--
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