<?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 --> <!-- --> <!-- Session timer where UAS incidates support for UPDATE. --> <!-- In this case, UAC will first use empty UPDATE, which we --> <!-- will reply with 400. UAC MUST retry sending UPDATE with SDP. --> <scenario name="Basic UAS responder"> <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:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Allow: UPDATE, INVITE Require: timer Session-Expires: 90;refresher=uac Content-Type: application/sdp 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:101 telephone-event/8000 a=fmtp:101 0-16 ]]> </send> <recv request="ACK" optional="true" rtd="true" crlf="true"> </recv> <recv request="UPDATE" crlf="true"> </recv> <send> <![CDATA[ SIP/2.0 400 Want SDP Body [last_Via:] [last_From:] [last_To:];tag=[call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Allow: INVITE Require: timer Session-Expires: 90;refresher=uac Content-Length: 0 ]]> </send> <recv request="UPDATE" 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:[local_ip]:[local_port];transport=[transport]> Allow: INVITE Require: timer Session-Expires: 90;refresher=uac Content-Type: application/sdp 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:101 telephone-event/8000 a=fmtp:101 0-16 ]]> </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>