diff --git a/tester/complex_sip_call_tester.c b/tester/complex_sip_call_tester.c
index 16e4417da..0a454848a 100644
--- a/tester/complex_sip_call_tester.c
+++ b/tester/complex_sip_call_tester.c
@@ -70,6 +70,24 @@ static FILE *sip_start(const char *senario, const char* dest_username, LinphoneA
 	return NULL;
 #endif
 }
+
+static FILE *sip_start_recv(const char *senario) {
+#if HAVE_SIPP
+	char *command;
+	FILE *file;
+
+	//until errors logs are handled correctly and stop breaks output, they will be DISABLED
+	command = ms_strdup_printf(SIPP_COMMAND" -sf %s -trace_err -trace_msg -rtp_echo -m 1 -d 1000",senario);
+
+	ms_message("Starting sipp commad [%s]",command);
+	file = popen(command, "r");
+	ms_free(command);
+	return file;
+#else
+	return NULL;
+#endif
+}
+
 /*static void dest_server_server_resolved(void *data, const char *name, struct addrinfo *ai_list) {
 	*(struct addrinfo **)data =ai_list;
 }*/
@@ -286,12 +304,49 @@ static void call_with_multiple_video_mline_in_sdp() {
 	linphone_core_manager_destroy(mgr);
 }
 
+static void call_invite_200ok_without_contact_header() {
+	LinphoneCoreManager *mgr;
+	char *identity_char;
+	char *scen;
+	FILE * sipp_out;
+	LinphoneCall *call = NULL;
+
+	/*currently we use direct connection because sipp do not properly set ACK request uri*/
+	mgr= linphone_core_manager_new2("empty_rc", FALSE);
+	mgr->identity = linphone_core_get_primary_contact_parsed(mgr->lc);
+	linphone_address_set_username(mgr->identity,"marie");
+	identity_char = linphone_address_as_string(mgr->identity);
+	linphone_core_set_primary_contact(mgr->lc,identity_char);
+
+	linphone_core_iterate(mgr->lc);
+
+	scen = bc_tester_res("sipp/call_invite_200ok_without_contact_header.xml");
+
+	sipp_out = sip_start_recv(scen);
+
+	if (sipp_out) {
+		call = linphone_core_invite(mgr->lc, "sipp@127.0.0.1");
+		BC_ASSERT_PTR_NOT_NULL(call);
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingInit, 1));
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingProgress, 1));
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallOutgoingRinging, 1));
+		if (call) {
+			BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallStreamsRunning, 1));
+			check_rtcp(call);
+			linphone_core_terminate_call(mgr->lc, call);
+		}
+		pclose(sipp_out);
+	}
+	linphone_core_manager_destroy(mgr);
+}
+
 static test_t tests[] = {
 	{ "SIP UPDATE within incoming reinvite without sdp", sip_update_within_icoming_reinvite_with_no_sdp },
 	{ "Call with audio mline before video in sdp", call_with_audio_mline_before_video_in_sdp },
 	{ "Call with video mline before audio in sdp", call_with_video_mline_before_audio_in_sdp },
 	{ "Call with multiple audio mline in sdp", call_with_multiple_audio_mline_in_sdp },
 	{ "Call with multiple video mline in sdp", call_with_multiple_video_mline_in_sdp },
+	{ "Call invite 200ok without contact header", call_invite_200ok_without_contact_header },
 };
 
 test_suite_t complex_sip_call_test_suite = {
diff --git a/tester/sipp/call_invite_200ok_without_contact_header.xml b/tester/sipp/call_invite_200ok_without_contact_header.xml
new file mode 100644
index 000000000..2a19c27ab
--- /dev/null
+++ b/tester/sipp/call_invite_200ok_without_contact_header.xml
@@ -0,0 +1,118 @@
+<?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="Basic UAS responder">
+  <!-- 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">
+  </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>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      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 96 97 0 8 101 98
+      a=rtpmap:96 speex/16000
+      a=fmtp:96 vbr=on
+      a=rtpmap:97 speex/8000
+      a=fmtp:97 vbr=on
+      a=rtpmap:101 telephone-event/16000
+      a=rtpmap:98 telephone-event/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+        optional="true"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <recv request="BYE">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+      Content-Length: 0
+
+    ]]>
+  </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="1000"/>
+
+
+  <!-- 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>
\ No newline at end of file