[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
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>>
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>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> <http://iax:guest@conference.freeswitch.org/888>
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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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