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Dan Bastone committed 2021-10-29 14:41:03 -04:00
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pjproject-2.11.1/tests/pjsua/scripts-sipp/uas-answer-200-multiple-fmts.xml
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<?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="UAS answer multiple formats">
<!-- 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/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:sipp@[local_ip]:[local_port]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4004 RTP/AVP 0 8 3 111
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 telephone-event/8000
a=fmtp:111 0-15
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<recv request="INVITE" crlf="true">
<action>
<ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
<ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
<assign assign_to="4" variable="5" />
</action>
</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:sipp@[local_ip]:[local_port]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4004 RTP/AVP 0 111
a=rtpmap:0 PCMU/8000
a=rtpmap:111 telephone-event/8000
a=fmtp:111 0-15
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<pause milliseconds="2000"/>
<send retrans="500">
<![CDATA[
BYE sip:[$5] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To[$3]
Call-ID: [call_id]
Cseq: 1 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
<!-- 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>