Attached u find two files, one of user agent client and another is user
agent server. everything works good without jump but when i try to
execute it with jump action it wont works. im using latest version of sipp.
*ERROR IS*
*** buffer overflow detected ***: sipp terminated
======= Backtrace: =========
/lib/libc.so.6(__fortify_fail+0x48)[0xd67b58]
/lib/libc.so.6[0xd66200]
/lib/libc.so.6[0xd65918]
/lib/libc.so.6(_IO_default_xsputn+0xcd)[0xce3e8d]
/lib/libc.so.6(_IO_vfprintf+0xf44)[0xcbb624]
/lib/libc.so.6(__vsprintf_chk+0xad)[0xd659cd]
/lib/libc.so.6(__sprintf_chk+0x30)[0xd65900]
sipp[0x80757f9]
sipp[0x80590d7]
sipp[0x8055473]
sipp[0x80567af]
sipp[0x8074022]
i am thankful to u if u identify what problem is actually happening
Regards
Rabia
<?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 UAC">
<!-- Client Sending INVITE -->
<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=[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>
<!--Response receiving -->
<recv response="200" >
</recv>
<!--######################-->
<!-- Sending Ack -->
<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=[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>
<!-- INFO method containing MSCML -->
<send retrans="500">
<![CDATA[
INFO sip:[EMAIL PROTECTED] 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]>
Call-ID: [call_id]
CSeq: 1 INFO
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sMSCML
Content-Length: [len]
<?xml version="1.0" encoding="UTF-8" ?>
<MediaServerControl version="1.0">
<request>
<play>
<prompt duration="50">
<audio url="file:///var/ivr/prompts/welcome.wav"/>
</prompt>
</play>
</request>
</MediaServerControl>
]]>
</send>
<!-- 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">
</recv>
<!-- Regular Expression -->
<recv request="INFO">
</recv>
<nop>
<action>
<jump value="5"/>
<log message=" jumping"/>
</action>
</nop>
<nop>
<action>
<log message="not jumping"/>
</action>
</nop>
<label id="5"/>
<send >
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/mscml
Content-Length: [len]
<?xml version="1.0" encoding="UTF-8" ?>
<MediaServerControl version="1.0">
<response id="3" request="play" code="4xx" text="abc" reason="EOF" playduration="50" playoffset="3">
<error_info code="123" text="abc" context="asd"/>
</response>
</MediaServerControl>
]]>
</send>
<!-- SENDING BYE -->
<send>
<![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=[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>
<!-- receiving 200 ok -->
<recv response="200">
</recv>
<!-- 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>
<?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">
<!-- Receiving Client's INVITE -->
<recv request="INVITE">
</recv>
<!-- Sending 200 ok -->
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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>
<!-- Receiving ACK -->
<recv request="ACK"
crlf="true">
</recv>
<!-- Receiving INFO method containing MSCML -->
<recv request="INFO">
</recv>
<!-- Sending 200 ok for INFO -->
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/mscml
Content-Length: [len]
<?xml version="1.0" encoding="UTF-8" ?>
<MediaServerControl version="1.0">
<request>
<play>
<prompt duration="50">
<audio url="file:///var/ivr/prompts/welcome.wav"/>
</prompt>
</play>
</request>
</MediaServerControl>
]]>
</send>
<!--pause milliseconds="1000"/ -->
<!-- Sending MSCML response in INFO method -->
<send>
<![CDATA[
INFO sip:[EMAIL PROTECTED] 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]>
Call-ID: [call_id]
CSeq: 1 INFO
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sMSCML
Content-Length: [len]
<?xml version="1.0" encoding="UTF-8" ?>
<MediaServerControl version="1.0">
<response id="3" request="play" code="4xx" text="abc" reason="EOF" playduration="50" playoffset="3">
<error_info code="123" text="abc" context="asd"/>
</response>
</MediaServerControl>
]]>
</send>
<!-- Receiving 200 ok from client -->
<recv response="200">
</recv>
<!-- Receiving BYE from client -->
<recv request="BYE">
</recv>
<!-- sending BYE's response 200 ok -->
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- 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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