* #36737: switch back to svn repo, remove assert in sip_transaction.c
diff --git a/jni/pjproject-android/.svn/pristine/3d/3d6bf116bdf7dd158613263767ebe829d64f2187.svn-base b/jni/pjproject-android/.svn/pristine/3d/3d6bf116bdf7dd158613263767ebe829d64f2187.svn-base
new file mode 100644
index 0000000..fe5169b
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/3d/3d6bf116bdf7dd158613263767ebe829d64f2187.svn-base
@@ -0,0 +1,108 @@
+<?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 -->
+
+
+<!-- -->
+<!-- Session timer where UAS doesn't indicate support for UPDATE. -->
+<!-- In this case, UAC MUST use re-INVITE with SDP. -->
+
+<scenario name="Basic UAS responder">
+ <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:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Require: timer
+ Session-Expires: 90;refresher=uac
+ Content-Type: application/sdp
+ 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:101 telephone-event/8000
+ a=fmtp:101 0-16
+ ]]>
+ </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:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Require: timer
+ Session-Expires: 90;refresher=uac
+ Content-Type: application/sdp
+ 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:101 telephone-event/8000
+ a=fmtp:101 0-16
+ ]]>
+ </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>
+