diff --git a/jni/pjproject-android/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml b/jni/pjproject-android/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml
new file mode 100644
index 0000000..9f4df39
--- /dev/null
+++ b/jni/pjproject-android/tests/pjsua/scripts-sipp/uas-422-then-200-bad-se.xml
@@ -0,0 +1,118 @@
+<?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/2.0 422 Session Timer too small
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
+      Min-SE:  5400
+      Content-Length: 0
+
+	]]>
+  </send>
+
+  <recv request="ACK"
+        optional="true"
+        rtd="true"
+        crlf="true"> 
+  </recv> 
+ 
+
+  <recv request="INVITE" crlf="true">
+  </recv>
+
+  <send retrans="500"> 
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
+      Allow-Events: telephone-event
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]> 
+      Supported: replaces
+      Session-Expires:  3600;refresher=uas
+      Require: timer
+      Content-Type: application/sdp
+      Content-Disposition: session;handling=required
+      Content-Length: [len]
+ 
+      v=0
+      o=Some-UserAgent 68 210 IN IP4 [local_ip]
+      s=SIP Call
+      c=IN IP4 [local_ip]
+      t=0 0
+      m=audio 17294 RTP/AVP 0 101
+      c=IN IP4 [local_ip]
+      a=rtpmap:0 PCMU/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-16
+      a=ptime:20 
+
+    ]]>
+  </send> 
+  
+  <recv request="ACK"
+        rtd="true"
+        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>
+