Misc (ticket #772): added several SIPp scenario files
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@2575 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/tests/pjsua/scripts-sipp/strict-route.xml b/tests/pjsua/scripts-sipp/strict-route.xml
new file mode 100644
index 0000000..a855209
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/strict-route.xml
@@ -0,0 +1,190 @@
+<?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="Strict route test">
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 407 Proxy Authenticate
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Proxy-Authenticate: DIGEST realm="test", nonce="12345", algorithm=MD5
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="false"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 100 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 180 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 183 progress
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:target@192.168.0.13>
+ Record-route: <sip:proxy@192.168.0.13>
+ Content-Type: application/sdp
+
+ v=0
+ o=- 3442013205 3442013205 IN IP4 192.168.0.13
+ s=pjsip
+ c=IN IP4 192.168.0.13
+ t=0 0
+ m=audio 4002 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ ]]>
+ </send>
+
+ <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:target@192.168.0.13>
+ Record-route: <sip:proxy@192.168.0.13;maddr=192.168.0.13>
+ Content-Type: application/sdp
+
+ v=0
+ o=- 3442013205 3442013205 IN IP4 192.168.0.13
+ s=pjsip
+ c=IN IP4 192.168.0.13
+ t=0 0
+ m=audio 4002 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="false"
+ rtd="true"
+ 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:target@192.168.0.13>
+ Record-route: <sip:proxy@192.168.0.13;maddr=192.168.0.13>
+ Content-Type: application/sdp
+
+ v=0
+ o=- 3442013205 3442013205 IN IP4 192.168.0.13
+ s=pjsip
+ c=IN IP4 192.168.0.13
+ t=0 0
+ m=audio 4002 RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="false"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="BYE" crlf="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ ]]>
+ </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>
+
diff --git a/tests/pjsua/scripts-sipp/uas-invite.xml b/tests/pjsua/scripts-sipp/uas-invite.xml
new file mode 100644
index 0000000..040f14b
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-invite.xml
@@ -0,0 +1,81 @@
+<?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="Basic UAS responder">
+ <!-- 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:192.168.0.15>
+ Content-Type: application/sdp
+
+ 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/SAVP 0 101
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:101 telephone-event/8000
+ a=fmtp:101 0-15
+ a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:D4Mf5fIPqxwse/lLrVc2XhLk7NSL6JI0k0Jps4Br
+
+ ]]>
+ </send>
+
+ <recv request="ACK" crlf="true">
+ </recv>
+
+ <!-- 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>
+
diff --git a/tests/pjsua/scripts-sipp/uas.xml b/tests/pjsua/scripts-sipp/uas.xml
new file mode 100644
index 0000000..a6d4854
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas.xml
@@ -0,0 +1,67 @@
+<?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="Basic UAS responder">
+ <!-- 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="REGISTER" 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:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ [last_Contact:]
+ Expires: 300
+ Content-Length: 0
+ ]]>
+ </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>
+