119 lines
4.4 KiB
XML
119 lines
4.4 KiB
XML
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<?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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<!-- Sipp default 'uas' scenario. -->
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<!-- -->
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<scenario name="Basic UAS responder">
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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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<!-- The '[last_*]' keyword is replaced automatically by the -->
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<!-- specified header if it was present in the last message received -->
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<!-- (except if it was a retransmission). If the header was not -->
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<!-- present or if no message has been received, the '[last_*]' -->
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<!-- keyword is discarded, and all bytes until the end of the line -->
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<!-- are also discarded. -->
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<!-- -->
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<!-- If the specified header was present several times in the -->
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<!-- message, all occurences are concatenated (CRLF seperated) -->
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<!-- to be used in place of the '[last_*]' keyword. -->
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<send retrans="500">
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<![CDATA[
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SIP/2.0 422 Session Timer too small
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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:[local_ip]:[local_port];transport=[transport]>
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Min-SE: 5400
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Content-Length: 0
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]]>
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</send>
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<recv request="ACK"
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optional="true"
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rtd="true"
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crlf="true">
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</recv>
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<recv request="INVITE" crlf="true">
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</recv>
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<send retrans="500">
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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:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
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Allow-Events: telephone-event
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Supported: replaces
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Session-Expires: 3600;refresher=uas
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Require: timer
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Content-Type: application/sdp
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Content-Disposition: session;handling=required
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Content-Length: [len]
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v=0
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o=Some-UserAgent 68 210 IN IP4 [local_ip]
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s=SIP Call
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c=IN IP4 [local_ip]
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t=0 0
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m=audio 17294 RTP/AVP 0 101
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c=IN IP4 [local_ip]
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a=rtpmap:0 PCMU/8000
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a=rtpmap:101 telephone-event/8000
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a=fmtp:101 0-16
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a=ptime:20
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]]>
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</send>
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<recv request="ACK"
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rtd="true"
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crlf="true">
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</recv>
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<!-- Keep the call open for a while in case the 200 is lost to be -->
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<!-- able to retransmit it if we receive the BYE again. -->
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<pause milliseconds="4000"/>
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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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