* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/tests/pjsua/scripts-sipp/uac-bad-ack.xml b/jni/pjproject-android/tests/pjsua/scripts-sipp/uac-bad-ack.xml
new file mode 100644
index 0000000..d93e30c
--- /dev/null
+++ b/jni/pjproject-android/tests/pjsua/scripts-sipp/uac-bad-ack.xml
@@ -0,0 +1,132 @@
+<?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 -->
+<!-- -->
+<!-- -->
+
+<scenario name="UAC with bad ACK">
+ <!-- UAC with bad ACK causes assertion with pjsip 1.4 -->
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <recv response="100"
+ optional="true">
+ </recv>
+
+ <recv response="180" optional="true">
+ </recv>
+
+ <!-- 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 response="200" rtd="true">
+ </recv>
+
+ <!-- Packet lost can be simulated in any send/recv message by -->
+ <!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- This delay can be customized by the -d command-line option -->
+ <!-- or by adding a 'milliseconds = "value"' option here. -->
+ <pause/>
+
+ <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-1
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200" crlf="true">
+ </recv>
+
+ <pause milliseconds="2000"/>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-1
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+ Call-ID: [call_id]
+ CSeq: 2 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Performance Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <!-- 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>
+