<?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@[local_ip]> Record-route: <sip:proxy@[local_ip]:[local_port]> Content-Type: application/sdp v=0 o=- 3442013205 3442013205 IN IP4 [local_ip] s=pjsip c=IN IP4 [local_ip] 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@[local_ip]> Record-route: <sip:proxy@[local_ip]:[local_port];maddr=[local_ip]> Content-Type: application/sdp v=0 o=- 3442013205 3442013205 IN IP4 [local_ip] s=pjsip c=IN IP4 [local_ip] 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="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>