Re #1166 (SDP offer/answer glare): added SIPp scenario file to reproduce this
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3382 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/tests/pjsua/scripts-sipp/uas-reinv-glare.xml b/tests/pjsua/scripts-sipp/uas-reinv-glare.xml
new file mode 100644
index 0000000..60c65f7
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-reinv-glare.xml
@@ -0,0 +1,153 @@
+<?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="Offer answer glare (#1166)">
+ <!-- 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>
+
+ <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]
+
+ v=0
+ o=- 1 1 IN IP4 192.168.0.15
+ s=pjmedia
+ c=IN IP4 192.168.0.15
+ t=0 0
+ m=audio 4004 RTP/AVP 0
+
+ ]]>
+ </send>
+
+ <recv request="ACK" crlf="true">
+ </recv>
+
+ <recv request="UPDATE" 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" />
+ <ereg regexp=".*" search_in="hdr" header="Via" assign_to="6"/>
+ <ereg regexp=".*" search_in="hdr" header="CSeq" assign_to="7"/>
+ </action>
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[$5] 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[$3]
+ Call-ID: [call_id]
+ Cseq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 2 2 IN IP4 192.168.0.15
+ s=pjmedia
+ c=IN IP4 192.168.0.15
+ t=0 0
+ m=audio 4004 RTP/AVP 0
+
+ ]]>
+ </send>
+
+ <recv response="491" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[$5] 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[$3]
+ Call-ID: [call_id]
+ Cseq: 1 INVITE
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ Via[$6]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ CSeq[$7]
+ 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>
+ <!-- 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>
+