[Freeswitch-users] Testing Freeswitch performance led to strange behavior

Apostolos Pantsiopoulos regs at kinetix.gr
Wed Jul 1 03:21:44 PDT 2009


I am writing this to let you know that this behavior
persists in the 1.0.4pre9.

Could the calls/sec issue be due to the single threaded nature of Sofia?
Because I am getting the feeling that the number of simultaneous 
channels doesn't really burdens FS, but many Calls/sec does.



Apostolos Pantsiopoulos wrote:
> Anthony Minessale wrote:
>> 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.
> 
>> 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.
> 
> 
>> 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/$1"/>
>                   </condition>
>               </extension>
> </context>
> 
> </include>
> 
> If you need anything else from the config just notify me.
> 
> In order to verify that at some point the calls start having a
> duration larger than the scenario's 5secs you can tcpdump on the sipp 
> machine or turn on the cdrs logging (I know that it degrades 
> performance, but as I said it is not a matter of when exactly it
> starts happening, it is a matter that it DOES start happening).
> 
> 
>>
>> On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr 
>> <mailto:regs at kinetix.gr> <regs at kinetix.gr <mailto:regs at kinetix.gr>> wrote:
>>
>>     Michael Collins wrote:
>>      >
>>      >
>>      >     The dialplan :
>>      >
>>      >     <?xml version="1.0" encoding="utf-8"?>
>>      >     <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
>>      >     <include>
>>      >
>>      >     <context name="mydialplan">
>>      >                  <extension name="dial1">
>>      >                      <condition field="destination_number"
>>      >     expression="^.*$">
>>      >
>>      >
>>      > 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 :)
>>      >
>>      >
>>      >
>>      >                      </condition>
>>      >                  </extension>
>>      >     </context>
>>      >
>>      >     </include>
>>      >
>>      >
>>      >
>>     ------------------------------------------------------------------------
>>      >
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>>
>>
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>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com 
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
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>>
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>>
>>
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> 
> 


-- 
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs at kinetix.gr
-------------------------------------------




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