---------- Forwarded message ----------
From: michal javorka <kostr...@gmail.com>
Date: 2011/4/10
Subject: Re: [Sipp-users] testing kamailio / openser
To: mayamatakeshi <mayamatake...@gmail.com>
Hi
I was trying to find something how to meassure the some metrics from RFC
6076 i found something about how to define som metrics in xml scenario,
something like this <send start_rtd="1" start_rtd="3" counter="1"
retrans="500"> , so this start_rtd="1" should start meassuring the variable
3 and rtd="3" in <recv response="200" rtd="3"> should stop it. I neet to
dump this variable into statistic, i tried to do it with -trace_stat but i
just get amount of calls in which variable was under 20ms and so on, but i
want to do graph where would be call rate and for example session request
delay from RFC, i dont know how to dump that into statistic and than do
graph from it.
Thanks for any suggestion.
Best redards
Michal
2011/4/5 michal javorka <kostr...@gmail.com>
> hi
> that scenarios worked great, but i have another problem,
> i want to meassure some of RFC 6076 - Basic Telephony SIP End-to-End
> Performance Metrics like SRD and so on, but i dont know how to meassure them
> i was looking for some articles on net but i have found nothing.
> Thanks
> BR
> Michal
>
>
> 2011/3/30 mayamatakeshi <mayamatake...@gmail.com>
>
>>
>>
>> On Tue, Mar 29, 2011 at 3:20 AM, michal javorka <kostr...@gmail.com>wrote:
>>
>>> HI it seems that your scenario will work properly, but i need somethnig
>>> without authentification i tried to do scenario from yours FINALY ACK pass
>>> throught proxyOpenser but i have same problem with BYE now , i was thinkig
>>> maybe you can help me with it?!
>>> client.xml in attachmend ist that i made.
>>> Thanx very much.
>>>
>>> 2011/3/28 mayamatakeshi <mayamatake...@gmail.com>
>>>
>>>>
>>>> On Tue, Mar 29, 2011 at 1:59 AM, michal javorka <kostr...@gmail.com>wrote:
>>>>
>>>>> Hi,
>>>>> I am trying to test an opeser proxy server, i am using this tutorial
>>>>> http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling
>>>>> i found this in the text: "In the sample XML files of SIPp,
>>>>> record-routing is not supported. Please change the script accordingly." i
>>>>> dont know how to change them i was trying it long time, can anybody give
>>>>> me
>>>>> suggestion how to do this?
>>>>>
>>>>
>>>> Try the attached scenario. It is the one I use.
>>>> HTH.
>>>>
>>>
>> I forgot that you also need the modified uas scenario for record-routing.
>> Try the attached file.
>>
>>
>
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