Hi Rodrigo
I have created a simple XML based on default UAC scenario, and just added a
pause of 3 seconds at the end of the script (see attached).
When using it with macthing -r and -l params, U can get the result U need.
Example:
When running the scenario with "sipp -sf uac_delay.xml <server ip> -r 10 -d
5000 -l 10"
U will get 10 calls coming up (and only 10 due to -l), maintained for 5
seconds, dropped and then script will halt for 3 seconds, and all over again.
Set -r and -l to the amount of concurrent calls U need, just make sure they
have the same value.
If rate is to high for your SUT, make the delay at the end bigger, and make
longer calls, so it will allow enough time for current calls to end before next
iteration begins.
I will be glad to know if it works for you,
Itzik.
________________________________
מאת: [EMAIL PROTECTED] בשם Peter Higginson
נשלח: ה 17/05/2007 00:23
אל: 'Rodrigo Pompei'; [email protected]
נושא: Re: [Sipp-users] Support to SIPp Interval between call
I don’t know of a direct method (but there are many recent changes I have not
kept up with).
What we have done locally is make a cps of –n correspond to n/10 calls/sec
(i.e. n calls in 10 seconds). It is a hack but is fairly simple to implement
(it takes advantage of the fact that the code tends to use atoi() for reading
parameters). We did experiment with –n is a call every n seconds but n/10 was
much better because it allows 2.5cps for example. If you are prepared to get
your hands dirty or any of the maintainers want the code for the standard
version I can sort out what the changes are.
Peter
________________________________
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Pompei
Sent: 15 May 2007 21:50
To: [email protected]
Subject: [Sipp-users] Support to SIPp Interval between call
ALL,
I need your support. I need make one test that the interval between calls to be
3 seconds.
What parameters I should set-up on SIPp?
Thanks,
Rodrigo Pompei
System Tester
Daitan Labs
Where Innovation Sparks Opportunity
Phone: +55 19 3707-5448
Fax : +55 19 3207-1437
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<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- 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 response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<pause milliseconds="3000"/>
<!-- 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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