129 lines
3.6 KiB
XML
129 lines
3.6 KiB
XML
<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Correct call cancellation on wrong SDP answer PR #3137 -->
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<scenario name="UAS sending 183 with incompatible codec">
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv request="INVITE" crlf="true">
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</recv>
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<send>
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<![CDATA[
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SIP/2.0 100 Trying
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Length: 0
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]]>
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</send>
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<send>
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<![CDATA[
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SIP/2.0 180 Ringing
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: sip:sipp@[local_ip]:[local_port]
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Content-Length: 0
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]]>
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</send>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 183 Session Progress
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: sip:sipp@[local_ip]:[local_port]
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=- 3441953879 3441953879 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 109 105
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a=rtpmap:109 EVS/16000
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a=fmtp:109 br=5.9-24.4; bw=nb-wb; max-red=220
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a=rtpmap:105 telephone-event/16000
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a=fmtp:105 0-15
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a=sendrecv
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]]>
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</send>
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<!-- Wait for CANCEL -->
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<recv request="CANCEL" crlf="true">
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</recv>
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<send>
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Length: 0
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]]>
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</send>
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<send>
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<![CDATA[
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SIP/2.0 487 Request Terminated
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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CSeq: [cseq] INVITE
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Contact: <sip:sipp@[local_ip]:[local_port]>
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Content-Length: 0
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]]>
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</send>
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<!-- Wait for CANCEL -->
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<recv request="ACK" crlf="true">
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</recv>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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