* #36737: switch back to svn repo, remove assert in sip_transaction.c
diff --git a/jni/pjproject-android/.svn/pristine/8d/8dfb3d828e7ad371e51da3f7089598b848f215a0.svn-base b/jni/pjproject-android/.svn/pristine/8d/8dfb3d828e7ad371e51da3f7089598b848f215a0.svn-base
new file mode 100644
index 0000000..8605381
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/8d/8dfb3d828e7ad371e51da3f7089598b848f215a0.svn-base
@@ -0,0 +1,138 @@
+<?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="UAS answer multiple formats, UAS supports UPDATE method">
+ <!-- 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">
+ <action>
+ <ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
+ <ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
+ <assign assign_to="4" variable="5" />
+ </action>
+ </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:sipp@[local_ip]:[local_port]
+ Content-Type: application/sdp
+ Content-Length: [len]
+ Allow: INVITE, UPDATE, ACK, BYE
+
+ 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/AVP 0 8 3 111
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:8 PCMA/8000
+ a=rtpmap:3 GSM/8000
+ a=rtpmap:111 telephone-event/8000
+ a=fmtp:111 0-15
+
+ ]]>
+ </send>
+
+ <recv request="ACK" crlf="true">
+ </recv>
+
+
+
+ <recv request="UPDATE" 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:sipp@[local_ip]:[local_port]
+ Content-Type: application/sdp
+ Content-Length: [len]
+ Allow: INVITE, UPDATE, ACK, BYE
+
+ 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/AVP 0 111
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:111 telephone-event/8000
+ a=fmtp:111 0-15
+
+ ]]>
+ </send>
+
+ <pause milliseconds="2000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:[$5] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port]
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To[$3]
+ Call-ID: [call_id]
+ Cseq: 1 BYE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+
+ <!-- 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>
+