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regs at kinetix.gr
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 Posted: Thu Jun 04, 2009 10:05 am Post subject: [Freeswitch-users] Testing Freeswitch performance led to str |
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Brian West wrote:
| Quote: |
On Jun 4, 2009, at 9:32 AM, Apostolos Pantsiopoulos wrote:
| Quote: | NOTE No 1 : All the performance recommendations found in the wiki has
been applied. In fact only the essential modules that could make this
scenario work
were loaded.
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What are you testing against? What OS, Hardware, Distro and such?
|
The small server tests were made on a 5-year old PC (32 bit, 3 Ghz P4,
Cetnos 5.3).
The large server 1 : Quad-Core AMD Opteron(tm) Processor 2350 HE (64
bit, Centos 5.3)
The large server 2 : Dual-Core AMD Opteron(tm) Processor 2214 HE (64
bit, Centos 5.3)
The large server 3 : Intel(R) Xeon(R) CPU E5345 @ 2.33GHz
(32 bit, Centos 5.3)
| Quote: |
| Quote: | NOTE No 2 : I tried using asterisk (as a point of reference - don't get
me wrong - I am not trying to start a flame war here). And it succeeded
doing on the same hardware 60 calls/sec with a channel limit of 400
sim. calls using only 50% of the cpu (maximum). No under any
circumstances I have seen the behavior above (this inability to hang
call legs in a timely manner). Even when I pushed asterisk to the limits
(80 calls per second 600 max call limit) and it started failing on some
calls it never failed to hangup the calls for both legs on exactly 10
secs.
|
Load testing is a science and you can do it wrong most of the time
unless you know exactly what you're doing. If you're going against the
default dialplan its heavy and not something I would load test against.
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The dialplan :
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="mydialplan">
<extension name="dial1">
<condition field="destination_number" expression="^.*$">
<!-- Dial Back -->
<action application="set"
data="absolute_codec_string=PCMA"/>
<action application="bridge"
data="sofia/gateway/sipp01/$1"/>
</condition>
</extension>
</context>
</include>
I think it is the simplest that can be used in this scenario.
| Quote: |
| Quote: | NOTE No 3 : As you can tell I was using a very small machine for my
tests. When I moved the same tests to larger installations (Quad Core
Opterons and Xeons) I got proportional results to the above.
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What are you testing on now? Hope its 64bit.
|
Most of the platforms were 64 bit (although the results that I posted
were from the small 32-bit server, the results from the 64-bit servers
were proportional to those). In other words we needed a large call/sec
rate for the high end servers but in any case the same phenomenon
occured at around 60% idle cpu.
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
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regs at kinetix.gr
Guest
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| Posted: Thu Jun 04, 2009 1:02 pm Post subject: [Freeswitch-users] Testing Freeswitch performance led to str |
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Michael Collins wrote:
| Quote: |
The dialplan :
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="mydialplan">
<extension name="dial1">
<condition field="destination_number"
expression="^.*$">
You forgot the parens around .*
It should be expression="^(.*)$" if you plan to use $1 later in the
extension...
<!-- Dial Back -->
<action application="set"
data="absolute_codec_string=PCMA"/>
<action application="bridge"
data="sofia/gateway/sipp01/$1"/>
... like here ^^^^^^^
-MC
|
You are right! Although, I don't think that would change the outcome of
my test
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regs at kinetix.gr
Guest
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| Posted: Fri Jun 05, 2009 3:37 am Post subject: [Freeswitch-users] Testing Freeswitch performance led to str |
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Anthony Minessale wrote:
| Quote: | FS uses async rtp timers so you may want to set rtp-timer-name=none in
the profile param to simulate asterisk conditions.
|
I tried that - although I am not using rtp in my scenario - with the
same results.
| Quote: | Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit
single cpu box because that was what was popular when it was designed
and the chance for race conditions is minimal because there is only 1
cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic
difference.
|
Yes I know that this machine is not well suited for today's test needs.
But the issue occurs in every machine as long as it is pushed near (but
not quite near) to its limits. I have the same odd durations using a 64
bit low end server. In this case I could achieve a better call/sec rate
than that of the crappy PC but around 50-60 calls/sec the same problem
showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the
same thing happened at a higher rate.
| Quote: |
I will be happy to investigate this issue a bit if you'd like but i do
not have any box like you describe so if I can't find anything
you may have to lend us your lab.
|
I would appreciate it if you did. After all there this might be a
problem that has not surfaced yet but someday will as more and more
production boxes start using FS. So it would be better to investigate it
now.
I don't think lending you access to my old P4 PC would help you very much
If you have access to a normal 2-4 core system you can easily start
raising the sipp parameters until it starts happening. However if you
really think it is appropriate to use my test machines I'd be happy to
grant access to our low-end Opteron machine (just send me a personal
email). I cannot grant you access to larger systems because they are
used in production.
I used the embedded sipp scenarios :
on the UAS side :
sipp -i <UAS_IP> -mi <UAS_IP> -ci <UAS_IP> -mp 8000 -sn uas
on the UAC side :
sipp <FS_IP>:5060 -s 44050505-i <UAC_IP> -mi <UAC_IP> -ci <UAC_IP> -r 70
-d 5000 -l 500 -m 2000 -sn uac
The dialplan :
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="mydialplan">
<extension name="dial1">
<condition field="destination_number" expression="(^.*)$">
<!-- Dial Back -->
<action application="set"
data="absolute_codec_string=PCMU"/>
<!-- <action application="set"
data="proxy_media=true"/> -->
<action application="bridge"
data="sofia/gateway/sipp01/$1"/>
</condition>
</extension>
</context>
</include>
If you need anything else from the config just notify me.
In order to verify that at some point the calls start having a
duration larger than the scenario's 5secs you can tcpdump on the sipp
machine or turn on the cdrs logging (I know that it degrades
performance, but as I said it is not a matter of when exactly it
starts happening, it is a matter that it DOES start happening).
| Quote: |
On Thu, Jun 4, 2009 at 12:47 PM, regs@kinetix.gr
<mailto:regs@kinetix.gr> <regs@kinetix.gr <mailto:regs@kinetix.gr>> wrote:
Michael Collins wrote:
| Quote: |
The dialplan :
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="mydialplan">
<extension name="dial1">
<condition field="destination_number"
expression="^.*$">
You forgot the parens around .*
It should be expression="^(.*)$" if you plan to use $1 later in the
extension...
<!-- Dial Back -->
<action application="set"
data="absolute_codec_string=PCMA"/>
<action application="bridge"
data="sofia/gateway/sipp01/$1"/>
... like here ^^^^^^^
-MC
|
You are right! Although, I don't think that would change the outcome of
my test
| Quote: |
</condition>
</extension>
</context>
</include>
|
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<mailto:Freeswitch-users@lists.freeswitch.org>
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
<mailto:MSN%3Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
<mailto:PAYPAL%3Aanthony.minessale@gmail.com>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<mailto:sip%3A888@conference.freeswitch.org>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org>
pstn:213-799-1400
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--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------
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