On Mon, Apr 11, 2011 at 5:14 AM, michal javorka <kostr...@gmail.com> wrote:
> 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.
Sorry, I cannot help here as I don't do any work with metrics (never read
the RFC 6076) and never found the need to get stats beyond the ones that
show up in real-time on the sipp output.
regards,
takeshi
>
>
> 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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