| 1 | <?xml version="1.0" encoding="ISO-8859-1" ?>
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| 2 | <!DOCTYPE scenario SYSTEM "sipp.dtd">
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| 3 |
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| 4 | <!-- This program is free software; you can redistribute it and/or -->
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| 5 | <!-- modify it under the terms of the GNU General Public License as -->
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| 6 | <!-- published by the Free Software Foundation; either version 2 of the -->
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| 7 | <!-- License, or (at your option) any later version. -->
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| 8 | <!-- -->
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| 9 | <!-- This program is distributed in the hope that it will be useful, -->
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| 10 | <!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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| 11 | <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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| 12 | <!-- GNU General Public License for more details. -->
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| 13 | <!-- -->
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| 14 | <!-- You should have received a copy of the GNU General Public License -->
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| 15 | <!-- along with this program; if not, write to the -->
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| 16 | <!-- Free Software Foundation, Inc., -->
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| 17 | <!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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| 18 | <!-- -->
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| 19 | <!-- Sipp default 'uac' scenario. -->
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| 20 | <!-- -->
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| 21 |
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| 22 | <scenario name="Basic Sipstone UAC">
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| 23 | <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
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| 24 | <!-- generated by sipp. To do so, use [call_id] keyword. -->
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| 25 | <!--
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| 26 | -->
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| 27 | <send retrans="500">
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| 28 | <![CDATA[
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| 29 |
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| 30 | INVITE sip:test66@192.168.1.67:5060;line=127;transport=tcp SIP/2.0
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| 31 | Via: SIP/2.0/TCP [local_ip];branch=[branch]
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| 32 | Route: <sip:[remote_ip]:5060>;transport=tcp;lr;drr
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| 33 | Route: <sip:192.168.1.67:5060>;transport=udp;lr;drr
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| 34 | Max-Forwards: 70
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| 35 | From: "" <sip:0000000000@[local_ip]>;tag=[pid]SIPpTag00[call_number]
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| 36 | To: <sip:test66@192.168.1.67:5060;line=127;transport=tcp>
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| 37 | Call-ID: [call_id]
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| 38 | CSeq: 92738401 INVITE
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| 39 | Contact: <sip:mod_sofia@[local_ip]:5060;transport=tcp>
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| 40 | User-Agent: FreeSWITCH-mod_sofia/1.6.7+git~20160401T013613Z~d38d065f51~64bit
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| 41 | Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
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| 42 | Supported: timer, path, replaces
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| 43 | Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description
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| 44 | Content-Type: application/sdp
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| 45 | Content-Disposition: session
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| 46 | Content-Length: [len]
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| 47 | X-FS-Support: update_display,send_info
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| 48 | Remote-Party-ID: <sip:0000000000@[local_ip]>;party=calling;screen=yes;privacy=off
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| 49 |
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| 50 | v=0
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| 51 | o=FreeSWITCH 1466121506 1466121507 IN IP6 local_ip
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| 52 | s=FreeSWITCH
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| 53 | c=IN IP[media_ip_type] [media_ip]
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| 54 | t=0 0
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| 55 | m=audio 27424 RTP/AVP 0 8 18 9 3
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| 56 | a=rtpmap:0 PCMU/8000
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| 57 | a=rtpmap:8 PCMA/8000
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| 58 | a=rtpmap:18 G729/8000
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| 59 | a=rtpmap:9 G722/8000
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| 60 | a=rtpmap:3 GSM/8000
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| 61 | a=ptime:20
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| 62 |
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| 63 | ]]>
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| 64 | </send>
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| 65 |
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| 66 | <recv response="100"
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| 67 | optional="true">
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| 68 | </recv>
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| 69 |
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| 70 | <recv response="404" auth="true">
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| 71 | </recv>
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| 72 |
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| 73 |
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| 74 | <!-- definition of the response time repartition table (unit is ms) -->
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| 75 | <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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| 76 |
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| 77 | <!-- definition of the call length repartition table (unit is ms) -->
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| 78 | <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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| 79 |
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| 80 | </scenario>
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| 81 |
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