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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 'uas' scenario. -->
<!-- -->
<scenario name="Strict route test">
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 407 Proxy Authenticate
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Proxy-Authenticate: DIGEST realm="test", nonce="12345", algorithm=MD5
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<recv request="INVITE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
]]>
</send>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
]]>
</send>
<send>
<![CDATA[
SIP/2.0 183 progress
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:target@192.168.0.13>
Record-route: <sip:proxy@192.168.0.13>
Content-Type: application/sdp
v=0
o=- 3442013205 3442013205 IN IP4 192.168.0.13
s=pjsip
c=IN IP4 192.168.0.13
t=0 0
m=audio 4002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</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:target@192.168.0.13>
Record-route: <sip:proxy@192.168.0.13;maddr=192.168.0.13>
Content-Type: application/sdp
v=0
o=- 3442013205 3442013205 IN IP4 192.168.0.13
s=pjsip
c=IN IP4 192.168.0.13
t=0 0
m=audio 4002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:target@192.168.0.13>
Record-route: <sip:proxy@192.168.0.13;maddr=192.168.0.13>
Content-Type: application/sdp
v=0
o=- 3442013205 3442013205 IN IP4 192.168.0.13
s=pjsip
c=IN IP4 192.168.0.13
t=0 0
m=audio 4002 RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="false"
rtd="true"
crlf="true">
</recv>
<recv request="BYE" crlf="true">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
]]>
</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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