Hello Everyone, I am facing a strange problem with SIPp.
When I was trying to make a call between a sipp client and sipp server (without putting any application in between), I saw memory leakage in SIPp application . But when I put one of my application(say Application A) in between sipp client and sipp server everything goes fine . But there is one more issue, actually I have two applications and one of them works fine with SIPp(Application A) but second one (Application B)shows the same kind of problem (i.e. memory leak). I have attached both the scripts (one for client and one for server side with scenario files). Just have a look and let me know if there is any problem. So now I can say that this problem is not because of any SIPp version not because of any OS (by the way I am testing it on Red Hat 5 Enterprise Edition) because if these were issues so it wont work for any application in any condition(these are my conclusion but then it doesn't stop to share your own too). Any Suggestion friends..???? So please tell me what should I do because due to memory leak after some time system memory were consumed and everything hangs. Waiting for your quick replies. Thanks Pramod Singh Ph: +919987532721
uac.sh
Description: Binary data
<?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. --> <!-- <!DOCTYPE scenario SYSTEM "service.dtd"> --> --> <!-- 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. --> <!-- --> <!-- INVITE sip:sip:[local_ip]:[local_po...@[remote_ip]:[remote_port] SIP/2.0 --> <!-- tel:+919987532721;phone-context=open-ims.test SIP/2.0 --> <!-- P-Charging-Vector: icid-value="AyretyU0dm+6O2IrT5tAFrbHLso=023551024";orig-ioi=orig1.fr;term-ioi=term1.fr --> <scenario name="Basic Sipstone UAC_1"> <send> <![CDATA[ INVITE sip:[remote_ip]:[remote_port] SIP/2.0 Call-ID: [call_id] CSeq: 1 INVITE Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]> Require:precondition Route: <sip:[email protected]:6001;lr> Supported: precondition,100rel P-Asserted-Identity: <sip:[email protected]> P-Charging-Function-Addresses:ccf=10.10.10.10;ccf=10.10.10.11;ecf=10.10.10.12;ecf=10.10.10.13 Contact: <sip:[local_ip]:[local_port]> Allow: INVITE, ACK, CANCEL, BYE, PRACK, UPDATE, REFER, MESSAGE P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=pramod1234 Privacy: none 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 ]]> <action> <ereg regexp=".*" search_in="hdr" header="Via" occurence="1" assign_to="1" /> </action> </send> <recv response="408" next="2" optional="true"> </recv> <recv response="400" next="2" optional="true"> </recv> <recv response="500" next="2" optional="true"> </recv> <recv response="481" next="2" optional="true"> </recv> <recv response="100" crlf="true" rtd="true" optional="true"> </recv> <recv response="408" next="2" optional="true"> </recv> <recv response="400" next="2" optional="true"> </recv> <recv response="500" next="2" optional="true"> </recv> <recv response="481" next="2" optional="true"> </recv> <recv response="183" crlf="true" rtd="true" rrs="true"> </recv> <!-- <recv response="500" next="2" optional="true"> --> <!-- </recv> --> <pause milliseconds="50"/> <send> <![CDATA[ PRACK sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];[branch] Max-Forwards: 70 From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Route: <sip:[email protected]:6001;lr> CSeq: 2 PRACK RACK: 314 1 INVITE Contact: <sip:[local_ip]:[local_port]> Content-Length: 0 Supported: precondition Require:precondition, 100rel ]]> </send> <recv response="408" next="2" optional="true"> </recv> <recv response="200" rtd="true" crlf="true" rrs="true"> </recv> <pause milliseconds="50"/> <send> <![CDATA[ UPDATE sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];[branch] Max-Forwards: 70 From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]>[peer_tag_param] Route: <sip:[email protected]:6001;lr> Call-ID: [call_id] CSeq: 3 UPDATE Contact: <sip:[local_ip]:[local_port]> Content-Length: 0 ]]> </send> <recv response="408" next="2" optional="true"> </recv> <recv response="200" rtd="true" crlf="true" rrs="true"> </recv> <recv response="408" next="2" optional="true"> </recv> <recv response="180" rtd="true" crlf="true" rrs="true"> </recv> <recv response="408" next="2" optional="true"> </recv> <recv response="200" rtd="true" crlf="true" rrs="true"> </recv> <pause milliseconds="50"/> <send> <![CDATA[ ACK sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];[branch] From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Route: <sip:[email protected]:6001;lr> CSeq: 1 ACK Contact: <sip:[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause milliseconds="1000"/> <send> <![CDATA[ BYE sip:[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];[branch] From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Route: <sip:[email protected]:6001;lr> CSeq: 4 BYE Contact: <sip:[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true" rtd="true" next="3" rrs="true"> </recv> <label id="2" /> <send> <![CDATA[ ACK sip:[remote_ip]:[remote_port] SIP/2.0 Via[$1] From: "AAM"<sip:[email protected]>;tag=[call_number] To: "BAAM"<sip:a...@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] Route: <sip:[email protected]:6001;lr> CSeq: 1 ACK Contact: <sip:[local_ip]:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <label id="3" /> <!-- pause 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>
uas.sh
Description: Binary data
<?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 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder_1">
<recv request="INVITE" crlf="true" rtd="true" rrs="true">
<action>
<ereg regexp=".*" search_in="hdr" header="Via" occurence="1" assign_to="1" />
<ereg regexp=".*" search_in="hdr" header="Via" occurence="2" assign_to="2" />
</action>
</recv>
<send >
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
CSeq: 1 INVITE
[last_Record-Route:]
Contact: <sip:[local_ip]:[local_port]>
Content-Length: 0
]]>
</send>
<pause milliseconds="50"/>
<send>
<![CDATA[
SIP/2.0 183 Session progress
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
CSeq: 1 INVITE
[last_Record-Route:]
Require:100rel
Rseq: 314
Contact: <sip:[local_ip]:[local_port]>
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 request="PRACK" crlf="true" rtd="true" rrs="true">
</recv>
<send >
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 2 PRACK
[last_Record-Route:]
Contact: <sip:[local_ip]:[local_port]>
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 request="UPDATE" crlf="true" rtd="true" rrs="true">
</recv>
<send >
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 3 UPDATE
[last_Record-Route:]
Contact: <sip:[local_ip]:[local_port]>
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>
<pause milliseconds="50"/>
<send >
<![CDATA[
SIP/2.0 180 Ringing
Via[$1]
Via[$2]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 1 INVITE
[last_Record-Route:]
Contact: <sip:[local_ip]:[local_port]>
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>
<pause milliseconds="50"/>
<send >
<![CDATA[
SIP/2.0 200 OK
Via[$1]
Via[$2]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 1 INVITE
Contact: <sip:[local_ip]:[local_port]>
Record-Route: <sip:[local_ip]:[local_port]>
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 request="ACK" rtd="true" crlf="true" rtd="true" rrs="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[routes]
[last_Call-ID:]
CSeq: 4 BYE
Contact: <sip:[local_ip]:[local_port]>
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. -->
<!-- pause milliseconds="1000"/>
<!-- 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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