Hi,
I am running the basic register ,invite call flow script from sipp client to
the server.the call flow runs fine with v4 addressing. when i run the same
script with the v6 addressing i do not receive the 200OK for the register
message i sent.is there something i am missing.attaching the scripts for
reference.please help.
branchc.xml is the client script
branchs.xml is the server script
Thanks,
Sheetal
<?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 'branchc' scenario. -->
<!-- -->
<scenario name="branch_client">
<send retrans="500">
<![CDATA[
REGISTER sip:CA.cym.com SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: ua1 <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: ua1 <sip:[EMAIL PROTECTED]:[local_port]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: sip:[EMAIL PROTECTED]:[local_port]
Content-Length: 0
Expires: 300
]]>
</send>
<!-- simple case - just jump over a line -->
<recv response="200" rtd="true" next="5">
</recv>
<recv response="200">
</recv>
<label id="5"/>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: ua[call_number] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: ua2 <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>
<!-- Do something different on an optional receive -->
<recv response="403" optional="true" next="1">
</recv>
<recv response="200">
<action>
<ereg regexp="ua25"
search_in="hdr"
header="From: "
assign_to="8"/>
</action>
</recv>
<!-- set variable 8 above on 25th call, send the ACK but skip the pause for it -->
<send next="1" test="8">
<![CDATA[
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: ua1 <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: ua2 <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>
<pause milliseconds="5000"/>
<label id="1"/>
<send retrans="500">
<![CDATA[
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: ua1 <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: ua2 <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="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>
<?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 'branchs' scenario. -->
<!-- -->
<scenario name="branch_server">
<recv request="REGISTER">
</recv>
<send>
<![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-Length: 0
Expires: 300
]]>
</send>
<!-- Set variable 3 if the ua is of the form ua2... -->
<recv request="INVITE" crlf="true">
<action>
<ereg regexp="ua2"
search_in="hdr"
header="From: "
assign_to="3"/>
</action>
</recv>
<!-- send 180 then trying if variable 3 is set -->
<send next="1" test="3">
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<!-- if not, send a 403 error then skip to wait for a BYE -->
<send next="2">
<![CDATA[
SIP/2.0 403 Error
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="1"/>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<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: 136
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<label id="2"/>
<recv request="BYE">
</recv>
<send>
<![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-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="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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