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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