Misc (ticket #772): added several SIPp scenario files

git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2575 74dad513-b988-da41-8d7b-12977e46ad98

3 files changed, 338 insertions, 0 deletions

diff --git a/tests/pjsua/scripts-sipp/strict-route.xml b/tests/pjsua/scripts-sipp/strict-route.xml
new file mode 100644
index 00000000..a855209f
--- /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 00000000..040f14ba
--- /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 00000000..a6d4854f
--- /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>

+