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.
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="BGS as destination">
<!-- arguments to sipp must include: -->
<!-- -i : local_ip -->
<!-- -p : local_port -->
<!-- -sf : scenario file -->
<!-- Ex.: sipp -i 192.168.2.121 -p 6060 -sf uas.xml -->
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv request="INVITE" crlf="true" >
</recv>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:uas@[local_ip]:[local_port]>
[last_Via:]
User-Agent: uas
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:uas@[local_ip]:[local_port]>
[last_Via:]
User-Agent: uas
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:uas@[local_ip]:[local_port]>
[last_Record-Route:]
Supported: timer
Session-Expires: 21600; refresher=uas
[last_Via:]
User-Agent: uas
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sip 1220546354 1220546354 IN IP[local_ip_type] [local_ip]
s=SIP_Call
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0 98
a=sendrecv
a=rtpmap:98 telephone-event/8000
]]>
</send>
<!-- The [last_Via:] above will cause all Via Headers received in the last message to be added at that point in the SIP message, but they will be combined in a single Header (separated by commas). So the message will not be exactly the same as generated by a real CSP -->
<recv request="INVITE" optional="true" />
<!-- the above is used to ignore duplicated INVITE sent by OpenSER -->
<recv request="ACK"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_To:]
[last_From:]
[last_Call-ID:]
[last_CSeq:]
[last_Via:]
User-Agent: uas
Content-Length: 0
]]>
</send>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
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