<?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"> <!-- 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 request="INVITE" crlf="true"> </recv> <!-- The '[last_*]' keyword is replaced automatically by the --> <!-- specified header if it was present in the last message received --> <!-- (except if it was a retransmission). If the header was not --> <!-- present or if no message has been received, the '[last_*]' --> <!-- keyword is discarded, and all bytes until the end of the line --> <!-- are also discarded. --> <!-- --> <!-- If the specified header was present several times in the --> <!-- message, all occurences are concatenated (CRLF seperated) --> <!-- to be used in place of the '[last_*]' keyword. --> <send retrans="500"> <![CDATA[ SIP/2.0 422 Session Timer too small [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Min-SE: 5400 Content-Length: 0 ]]> </send> <recv request="ACK" optional="true" rtd="true" crlf="true"> </recv> <recv request="INVITE" crlf="true"> </recv> <send retrans="500"> <![CDATA[ SIP/2.0 200 OK [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:[local_ip]:[local_port];transport=[transport]> Supported: replaces Session-Expires: 3600;refresher=uas Require: timer Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: [len] v=0 o=Some-UserAgent 68 210 IN IP4 [local_ip] s=SIP Call c=IN IP4 [local_ip] t=0 0 m=audio 17294 RTP/AVP 0 101 c=IN IP4 [local_ip] a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ]]> </send> <recv request="ACK" rtd="true" crlf="true"> </recv> <!-- 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>