From 52a5ab76e751c9f1c08bcdd9082d104223264f90 Mon Sep 17 00:00:00 2001
From: Sylvain Berfini <sylvain.berfini@belledonne-communications.com>
Date: Tue, 22 Sep 2015 10:55:43 +0200
Subject: [PATCH] Added test for multiple video streams in sdp

---
 tester/complex_sip_call.c                     |  39 ++++++
 .../call_with_multiple_video_mline_in_sdp.xml | 129 ++++++++++++++++++
 2 files changed, 168 insertions(+)
 create mode 100644 tester/sipp/call_with_multiple_video_mline_in_sdp.xml

diff --git a/tester/complex_sip_call.c b/tester/complex_sip_call.c
index da9a68165..d82841f28 100644
--- a/tester/complex_sip_call.c
+++ b/tester/complex_sip_call.c
@@ -228,11 +228,50 @@ static void call_with_multiple_audio_mline_in_sdp() {
 	linphone_core_manager_destroy(mgr);
 }
 
+static void call_with_multiple_video_mline_in_sdp() {
+	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_with_multiple_video_mline_in_sdp.xml");
+	
+	sipp_out = sip_start(scen, linphone_address_get_username(mgr->identity), mgr->identity);
+
+	if (sipp_out) {
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallIncomingReceived, 1));
+		call = linphone_core_get_current_call(mgr->lc);
+		linphone_core_accept_call(mgr->lc, call);
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallStreamsRunning, 1));
+		BC_ASSERT_EQUAL(call->main_audio_stream_index, 0, int, "%d");
+		BC_ASSERT_EQUAL(call->main_video_stream_index, 1, int, "%d");
+		BC_ASSERT_TRUE(call->main_text_stream_index > 3);
+		BC_ASSERT_TRUE(linphone_call_log_video_enabled(linphone_call_get_call_log(call)));
+		
+		check_rtcp(call);
+	
+		BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallEnd, 1));
+		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 },
 };
 
 test_suite_t complex_sip_call_test_suite = {
diff --git a/tester/sipp/call_with_multiple_video_mline_in_sdp.xml b/tester/sipp/call_with_multiple_video_mline_in_sdp.xml
new file mode 100644
index 000000000..77fd615e9
--- /dev/null
+++ b/tester/sipp/call_with_multiple_video_mline_in_sdp.xml
@@ -0,0 +1,129 @@
+<?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]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>
+      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 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
+      m=video [media_port+2] RTP/AVP 96
+      a=rtpmap:96 VP8/90000
+      m=video [media_port+3] RTP/AVP 96
+      a=rtpmap:96 VP8/90000
+      m=video [media_port+4] RTP/AVP 96
+      a=rtpmap:96 VP8/90000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <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@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[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>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </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>
+