From 7c36fbc0053afd30c65eee0babdb99e582d37580 Mon Sep 17 00:00:00 2001
From: Jehan Monnier <jehan.monnier@linphone.org>
Date: Wed, 16 Sep 2015 17:36:05 +0200
Subject: [PATCH] add missing sipp scenario

---
 ...te_within_icoming_reinvite_with_no_sdp.xml | 175 ++++++++++++++++++
 1 file changed, 175 insertions(+)
 create mode 100644 tester/sipp/sip_update_within_icoming_reinvite_with_no_sdp.xml

diff --git a/tester/sipp/sip_update_within_icoming_reinvite_with_no_sdp.xml b/tester/sipp/sip_update_within_icoming_reinvite_with_no_sdp.xml
new file mode 100644
index 000000000..eef5c3cd2
--- /dev/null
+++ b/tester/sipp/sip_update_within_icoming_reinvite_with_no_sdp.xml
@@ -0,0 +1,175 @@
+<?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 'uac' scenario.                       -->
+<!--                                                                    -->
+
+<scenario name="Basic Sipstone UAC">
+  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
+  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:[service]@sip.example.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>
+      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>
+
+  <recv response="200" rrs="true" rtd="true">
+  </recv>
+
+  <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@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>[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>
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:[service]@sip.example.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+    <recv response="100"
+        optional="true">
+  </recv>
+
+
+  <recv response="200" rtd="true">
+  </recv>
+ 
+
+<send retrans="500">
+    <![CDATA[
+  
+      UPDATE sip:[service]@sip.example.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 UPDATE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+  <recv response="200" optionnal="true" rtd="true">
+  </recv>
+ <send >
+    <![CDATA[
+      ACK sip:[service]@sip.example.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 ACK
+      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>
+
+  <pause/>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[service]@sip.example.org SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@sip.example.org>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@sip.example.org>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 4 BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+<!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <!-- 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>
+