[Freeswitch-users] XML_CURL max connection limit

Wed Jun 12 18:02:49 MSD 2013

You could either throw Cacti on the server, or use server density/new relic
trial for quick and easy graphing.. or you can just watch `top` and push
number 1 whilst running to see the CPU breakdown.

Cal

On Wed, Jun 12, 2013 at 1:19 PM, Muhammad Shahzad <shaheryarkh at gmail.com>wrote:

> Steven, i am using soft timer. Should i try another? Please suggest.
>
> Cal, no there is nothing unusually changes in RAM or CPU. BTW, i have 24
> CPU cores with 16 GB of RAM on each server. Regarding CPU graph, how can i
> enable it.
>
> Thank you.
>
>
>
> On Wed, Jun 12, 2013 at 1:15 PM, Cal Leeming [Simplicity Media Ltd] <
> cal.leeming at simplicitymedialtd.co.uk> wrote:
>
>> This sounds quite interesting.. From my own benchmarks, I've seen that
>> IVRs applications can be quite stressful on the server.
>>
>> Have you checked to see if there is a CPU util spike or context switching
>> spike during these tests? Perhaps you are hitting the floor limit of what
>> your CPU is capable of?
>>
>> Do you have CPU graphs? If not, can you enable them and try again?
>>
>> Cal
>>
>>
>> On Wed, Jun 12, 2013 at 10:21 AM, Muhammad Shahzad <shaheryarkh at gmail.com
>> > wrote:
>>
>>> You are right. After many improvements and modifications to dail plan,
>>> removing unnecessary modules and using Ngnix web server, things are much
>>> better now. Except that directory information is fetched 3 times per call
>>> and there are overall 4 HTTP hits in a single call (3 for user + 1 for dial
>>> plan).
>>>
>>> Since Monday, we have started analyzing the results of load test,
>>> specially looking into packet traces, voicemail recordings and so on. We
>>> observed a strange thing, although we were able to test 1536 concurrent
>>> calls (the limit we deliberately put in switch.xml.conf) at up to 64 calls
>>> per second rate. But there is strange and unexplained delay in starting RTP
>>> from FS side.
>>>
>>> FS plays an IVR in early media state, then answers the call to directly
>>> start recording voicemail. The problem is if call rate is under 25 CPS,
>>> everything works fine. At 25 CPS there is minor delay in starting RTP in
>>> early media state. This delay is negligible after processing few hundred
>>> thousand calls but after about 1.2 million calls the delay has creeps up to
>>> about 1.2 seconds. After 30 CPS this delay is significantly increases to 2
>>> second after around 400 calls had done. At 50 CPS its ~3 second and after
>>> 60 CPS its around 4 seconds. Interesting thing is the delay is ONLY in
>>> starting playing IVR in early media state, once the IVR starts playing
>>> everything works perfectly fine. No call fails, but this delay would be
>>> annoying for anyone using the servers with over 30 CPS.
>>>
>>> Just to let you know, we checked everything, system resources, bandwidth
>>> usage, IVR file formats etc. all seems to be fine. We even tested with
>>> various FS start up options, e.g. -rp -nosql -nonat etc. etc. Nothing seems
>>> to have any impact on the results.
>>>
>>> I have run of out of possible explanations, so calling you guys for help.
>>>
>>> Thank you.
>>>
>>>
>>>
>>>
>>> On Fri, Jun 7, 2013 at 5:20 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I would need to see the specifics of your dialplan, but I suspect you
>>>> have some transfers or execute extension that are causing it to re-enter
>>>> dialplan.  Are you sure on all the others that they are making the exact
>>>> same requests?  Details here will be necessary to understand exactly what
>>>> is going on..
>>>>
>>>> Mike
>>>>
>>>> On Jun 7, 2013, at 4:07 AM, Muhammad Shahzad <shaheryarkh at gmail.com>
>>>> wrote:
>>>>
>>>> Guy, after detailed analysis i can confirm it was an Apache issue, and
>>>> changing Apache configuration changes the concurrent call limit. However,
>>>> one thing i observed about xml curl is that it makes several request with
>>>> same parameters per call. Even if we have no calls running on server, as
>>>> well as as Apache is completely free. I can see FS makes at least
>>>>
>>>> 1). 3 hits for dialplan.
>>>> 2). 2 hits for directory.
>>>> 3). 4 hits for voicemail.
>>>> 4). 3 hits for ivr.
>>>> 5). 2 hits for sofia.
>>>> and so on..
>>>>
>>>> While ideally only one hit should be sufficient to get all these
>>>> configurations. What is the point of making all these redundant hits? We
>>>> have roughly 15 - 20 hits made per call, which is huge and causes
>>>> significant load on Apache.
>>>>
>>>> Both Apache and FS are running on dedicated machines and connected over
>>>> Gigabit Ethernet with zero packet loss, so i doubt these hits are
>>>> "re-transmission".
>>>>
>>>> Thank you.
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Jun 6, 2013 at 1:24 AM, Muhammad Shahzad <shaheryarkh at gmail.com
>>>> > wrote:
>>>>
>>>>> Yes, its a feature, i know. I was just looking at how this feature was
>>>>> tested, as mentioned in its logs,
>>>>>
>>>>> 2010-01-04 15:01:51.945020 [INFO] mod_dialplan_xml.c:408 Processing
>>>>> 21653XXXXX->121637XXXXX in context termination
>>>>> 2010-01-04 15:01:52.054466 [ERR] mod_xml_curl.c:310 Received HTTP
>>>>> error 0 trying to fetch http://192.168.0.2/gateway.php
>>>>> 2010-01-04 15:01:52.054545 [CONSOLE] mod_xml_curl.c:317 XML response
>>>>> is in /tmp/6632ebb1-d19f-4d51-a07e-ec2ce0387f83.tmp.xml
>>>>>
>>>>> Thank you.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 6, 2013 at 12:48 AM, Michael Jerris <mike at jerris.com>wrote:
>>>>>
>>>>>> This is adding a feature to allow timeout to be set in ms instead of
>>>>>> seconds.  This is not a problem, its a feature, that was added a while ago.
>>>>>>  Until you can determine if the request is being sent for the one thats
>>>>>> failing, your not going to solve this issue.
>>>>>>
>>>>>> Mike
>>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>
>>>
>>> --
>>> Mit freundlichen Grüßen
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +49 176 99 83 10 85
>>> MSN: shari_786pk at hotmail.com
>>> Email: shaheryarkh at googlemail.com
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
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>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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