Hi DRob,

thx a lot, I think that's exactly what I needed to know. Thought of
seconds as value... but you're right!
Changed the test and... we'll see if it works out. But seems to work
smoothly now!!

THX a lot!

greetz

Am 20.01.2014 13:06, schrieb Rob Day:
> Hi Moritz,
> 
> The error log you posted says:
> 
> 2014-01-20      11:17:57:194    1390213077.194807: Call-Id: 1-20926@1.2.3.4,
> receive timeout on message Test LCR:1 without label to jump to
> (ontimeout attribute): aborting call.
> 2014-01-20      11:17:57:195    1390213077.195609: Dead call 1-20926@1.2.3.4
> (aborted at index 1), received 'SIP/2.0 302 Temporarely Moved'...
> 
> What this means is:
> * at 11:17:57:194 SIPp decides it hasn't seen the 302 message in time,
> so it aborts the call
> * at 11:17:57:195 the 302 message arrives, but is ignored because the
> call has already failed
> 
> Looking at your SIPp script, you've set timeout="1" on your <recv>
> element for the 302. This means that if the 302 message doesn't arrive
> within 1 millisecond, SIPp will fail the call.
> 
> I expect what is happening is that, in the 199 out of 200 cases, the
> response is very fast and arrives in under a millisecond (and there
> might be a few milliseconds of grace in the SIPp scheduler). But some
> of the time, the response will take just more than a millisecond to
> arrive, and so you see logs like the above - where SIPp times the call
> out and the message arrives straight after.
> 
> I recommend setting your timeout to a more reasonable value - 1 second
> (timeout="1000") should be plenty.
> 
> Best,
> Rob
> 
> On 20 January 2014 10:45, Moritz Graf <moritz.g...@g-fit.de> wrote:
>> Hi Folks,
>>
>> i want to check the following scenario:
>>
>> SIPP               kamailio redirect server
>>   #                             #
>> ------------------------------->
>>   INVITE
>>
>> <-------------------------------
>>           302 Moved temporartely
>>
>> So I am sending an INVITE to a kamailio. In the 302 answer the server
>> submits a redirect destination in the CONTACT-Header. I want to verify
>> if there is a correct value.
>>
>> The Check works in 199 from 200 cases. But in this one case the check
>> fails... and I can't find the reason.
>>
>> I am calling sipp as follows:
>> ######################################################################
>> /usr/bin/sipp -trace_err -m 1 -r 5 -timeout 1s -send_timeout 1s
>> -recv_timeout 1s -f 10 -p 5065 -i 1.2.3.4 -sf
>> /usr/lib/nagios/plugins/check_redirect_stateless.sipp -inf
>> /var/lib/nagios/check_redirect_stateless.user 1.2.3.4:5070
>> ######################################################################
>>
>> The userfile looks like this:
>> ######################################################################
>> SEQUENTIAL
>> # $Id: ----
>> # This is the INPUT-File for check_simple_redirect
>> # SIPp-Testscript
>> #server,user,expectedResult
>> 1.2.3.4;B900100491234567;B191000491234567
>> 1.2.3.4;B900200491234567;B291000491234567
>> ######################################################################
>> First Column: kamailio Server IP
>> Second Column: The String to be sent in the INVITE in the TO: HEader
>> Third Column: The String I expect in the Contact: Header
>>
>> And finaly the sipp check:
>> ######################################################################
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <!-- $Id: -----
>>      This is the SIPp XML-File for a simple Redirect Check
>>      -->
>> <scenario name="Test LCR">
>>   <send>
>>     <![CDATA[
>>
>>       INVITE sip:[field1]@[field0]:[remote_port] SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>       From: sipp <sip:[field1]@[field0]:[local_port]>;tag=[pid]
>>       To: sut <sip:[field1]@[field0]>
>>       Call-ID: [call_id]
>>       CSeq: 1 INVITE
>>       Contact: <sip:[field1]@[local_ip]:[local_port];transport=udp>
>>       Content-Type: application/sdp
>>       Content-Length: [len]
>>
>>       v=0
>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>       s=-
>>       c=IN IP[local_ip_type] [local_ip]
>>       t=0 0
>>       m=audio [auto_media_port] RTP/AVP 8
>>       a=rtpmap:8 PCMA/8000
>>       a=rtpmap:101 telephone-event/8000
>>       a=fmtp:101 0-11,16
>>
>>     ]]>
>>   </send>
>>   <!-- waitr for 302 -->
>>   <recv response="302" optional="false" next="OK" test="test_OK"
>> timeout="1">
>>    <action>
>>    <!-- regex, contactpart to 'test' -->
>>     <ereg regexp="<sip:([0-9A-Z]+)@.*" search_in="hdr" header="Contact:"
>>                 case_indep="true" check_it="true" assign_to="null,test"/>
>>     <!-- field2 to fieldstring save for strcmp -->
>>     <assignstr assign_to="fieldstring" value="[field2]" />
>>     <!-- check if contact has correct string and save to result -->
>>     <strcmp assign_to="result" variable="test" variable2="fieldstring"
>>                 check_it="true"/>
>>     <log message="result=[$result] test=[$test]
>> fieldstring=[$fieldstring]" />
>>     <!-- check if strings are equal -->
>>     <test assign_to="test_OK" variable="result" compare="equal"
>> value="0.0" />
>>    </action>
>>   </recv>
>>
>> <!-- recv is only reached if the above check failed-->
>> <recv response="599" timeout="1"></recv>
>> <label id="OK" />
>>
>>   <Reference variables="test"/>
>>   <Reference variables="result"/>
>>   <Reference variables="null" />
>>   <Reference variables="test_OK" />
>> </scenario>
>> ######################################################################
>>
>> The ERROR logged by sipp is as follows:
>> ######################################################################
>> sipp: The following events occured:
>> 2014-01-20      11:17:57:194    1390213077.194807: Call-Id: 1-20926@1.2.3.4,
>> receive timeout on message Test LCR:1 without label to jump to
>> (ontimeout attribute): aborting call.
>> 2014-01-20      11:17:57:195    1390213077.195609: Dead call 1-20926@1.2.3.4
>> (aborted at index 1), received 'SIP/2.0 302 Temporarely Moved
>> Via: SIP/2.0/UDP 1.2.3.4:5065;branch=z9hG4bK-20926-1-0
>> From: sipp <sip:B900100491234567@1.2.3.4:5065>;tag=20926
>> To: sut
>> <sip:B900100491234567@1.2.3.4>;tag=56398122a1ea83737b581412f70829d1.df04
>> Call-ID: 1-20926@1.2.3.4
>> CSeq: 1 INVITE
>> Contact: <sip:B191000491234567@1.2.3.4:5070>
>> Server: kamailio (4.0.3 (i386/linux))
>> Content-Length: 0
>>
>> '.
>> ######################################################################
>>
>>
>> Any help appricaiated!!
>>
>> greetz
>>
>>
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>>
> 


-- 

Moritz Graf, B.Sc.
Betrieb NGN-Plattform

G-FIT GmbH & Co. KG
Greflingerstr. 26, 93055 Regensburg

Telefon +49 (9 41) 69 85 - 1 86
Telefax +49 (9 41) 69 85 - 2 86
mailto:moritz.g...@g-fit.de
http://www.g-fit.de

G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co.
KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg,
HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher

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