Re #1864: fixed assert and crash when remote remove and add back media lines in SDP:

 - reverted back r5124 (of this ticket) and r4373 (of #1626)
 - applied another fix for #1626 as in described in the ticket desc
 - also fixed a related add/remove media issue:
    - we send an offer with 2 audio and 1 video in SDP
    - remote answers with only 1 audio in SDP
    - when we send a re-offer/re-INVITE, we have 5 media lines in our SDP (3 audio + 2 video, note 1 audio and 1 video are disabled)
    - call settings are actually maintained (2 active audio + 1 active video), just better to avoid the additional disabled media.



git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@5135 74dad513-b988-da41-8d7b-12977e46ad98

5 files changed, 567 insertions, 33 deletions

diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c
index 9c70f264..2bd856a4 100644
--- a/pjsip/src/pjsua-lib/pjsua_media.c
+++ b/pjsip/src/pjsua-lib/pjsua_media.c
@@ -1139,6 +1139,72 @@ static void sort_media(const pjmedia_sdp_session *sdp,
     }
 }
 
+
+/* Go through the list of media in the call, find acceptable media, and
+ * sort them based on the "quality" of the media, and store the indexes
+ * in the specified array. Media with the best quality will be listed
+ * first in the array.
+ */
+static void sort_media2(const pjsua_call_media *call_med,
+			unsigned call_med_cnt,
+			pjmedia_type type,
+			pj_uint8_t midx[],
+			unsigned *p_count,
+			unsigned *p_total_count)
+{
+    unsigned i;
+    unsigned count = 0;
+    int score[PJSUA_MAX_CALL_MEDIA];
+
+    pj_assert(*p_count >= PJSUA_MAX_CALL_MEDIA);
+    pj_assert(*p_total_count >= PJSUA_MAX_CALL_MEDIA);
+
+    *p_count = 0;
+    *p_total_count = 0;
+    for (i=0; i<PJSUA_MAX_CALL_MEDIA; ++i)
+	score[i] = 1;
+
+    /* Score each media */
+    for (i=0; i<call_med_cnt && count<PJSUA_MAX_CALL_MEDIA; ++i) {
+
+	/* Skip different media */
+	if (call_med[i].type != type) {
+	    score[count++] = -22000;
+	    continue;
+	}
+
+	/* Is it active? */
+	if (!call_med[i].tp) {
+	    score[i] -= 10;
+	}
+
+	++count;
+    }
+
+    /* Created sorted list based on quality */
+    for (i=0; i<count; ++i) {
+	unsigned j;
+	int best = 0;
+
+	for (j=1; j<count; ++j) {
+	    if (score[j] > score[best])
+		best = j;
+	}
+	/* Don't put media with negative score, that media is unacceptable
+	 * for us.
+	 */
+	midx[i] = (pj_uint8_t)best;
+	if (score[best] >= 0)
+	    (*p_count)++;
+	if (score[best] > -22000)
+	    (*p_total_count)++;
+
+	score[best] = -22000;
+
+    }
+}
+
+
 /* Callback to receive media events */
 pj_status_t call_media_on_event(pjmedia_event *event,
                                 void *user_data)
@@ -1528,23 +1594,21 @@ on_return:
 }
 
 
-/* If idx == 0, clean up media transports in provisional media that
- * is not used by call media, else clean up media transports starting
- * from index idx that have been removed by remote.
+/* Clean up media transports in provisional media that is not used by
+ * call media.
  */
-static void media_prov_clean_up(pjsua_call_id call_id, int idx)
+void pjsua_media_prov_clean_up(pjsua_call_id call_id)
 {
     pjsua_call *call = &pjsua_var.calls[call_id];
     unsigned i;
 
-    if (idx > 0 || call->med_prov_cnt > call->med_cnt) {
+    if (call->med_prov_cnt > call->med_cnt) {
         PJ_LOG(4,(THIS_FILE, "Call %d: cleaning up provisional media, "
         		     "prov_med_cnt=%d, med_cnt=%d",
-			     call_id, (idx == 0? call->med_prov_cnt: idx),
-			     call->med_cnt));
+			     call_id, call->med_prov_cnt, call->med_cnt));
     }
 
-    for (i = idx; i < call->med_prov_cnt; ++i) {
+    for (i = 0; i < call->med_prov_cnt; ++i) {
 	pjsua_call_media *call_med = &call->media_prov[i];
 	unsigned j;
 	pj_bool_t used = PJ_FALSE;
@@ -1553,7 +1617,7 @@ static void media_prov_clean_up(pjsua_call_id call_id, int idx)
 	    continue;
 
 	for (j = 0; j < call->med_cnt; ++j) {
-	    if (idx == 0 && call->media[j].tp == call_med->tp) {
+	    if (call->media[j].tp == call_med->tp) {
 		used = PJ_TRUE;
 		break;
 	    }
@@ -1573,11 +1637,6 @@ static void media_prov_clean_up(pjsua_call_id call_id, int idx)
     call->med_prov_cnt = 0;
 }
 
-void pjsua_media_prov_clean_up(pjsua_call_id call_id)
-{
-    media_prov_clean_up(call_id, 0);
-}
-
 
 pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 				     pjsip_role_e role,
@@ -1664,6 +1723,10 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 
     /* Get media count for each media type */
     if (rem_sdp) {
+
+	/* We are sending answer, check media count for each media type
+	 * from the remote SDP.
+	 */
 	sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp,
 		   maudidx, &maudcnt, &mtotaudcnt);
 	if (maudcnt==0) {
@@ -1695,22 +1758,22 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 
     } else {
 
-	/* If call already established, calculate media count from current 
-	 * local active SDP and call setting. Otherwise, calculate media
-	 * count from the call setting only.
+	/* If call is already established, adjust the existing call media list
+	 * to media count setting in call setting, e.g: re-enable/disable/add
+	 * media from existing media.
+	 * Otherwise, apply media count from the call setting directly.
 	 */
 	if (reinit) {
-	    const pjmedia_sdp_session *sdp;
-
-	    status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &sdp);
-	    pj_assert(status == PJ_SUCCESS);
 
-	    sort_media(sdp, &STR_AUDIO, acc->cfg.use_srtp,
-		       maudidx, &maudcnt, &mtotaudcnt);
+	    /* We are sending reoffer, check media count for each media type
+	     * from the existing call media list.
+	     */
+	    sort_media2(call->media_prov, call->med_prov_cnt,
+			PJMEDIA_TYPE_AUDIO, maudidx, &maudcnt, &mtotaudcnt);
 	    pj_assert(maudcnt > 0);
 
-	    sort_media(sdp, &STR_VIDEO, acc->cfg.use_srtp,
-		       mvididx, &mvidcnt, &mtotvidcnt);
+	    sort_media2(call->media_prov, call->med_prov_cnt,
+			PJMEDIA_TYPE_VIDEO, mvididx, &mvidcnt, &mtotvidcnt);
 
 	    /* Call setting may add or remove media. Adding media is done by
 	     * enabling any disabled/port-zeroed media first, then adding new
@@ -1855,8 +1918,9 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 		pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_DISABLED);
 	    }
 
-	    /* Put media type just for info */
-	    call_med->type = media_type;
+	    /* Put media type just for info if not yet defined */
+	    if (call_med->type == PJMEDIA_TYPE_NONE)
+		call_med->type = media_type;
 	}
     }
 
@@ -1991,11 +2055,12 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id,
 
 	if (rem_sdp && mi >= rem_sdp->media_count) {
 	    /* Remote might have removed some media lines. */
-	    for (i = rem_sdp->media_count; i < call->med_prov_cnt; ++i) {
-		stop_media_stream(call, i);
-	    }
-            media_prov_clean_up(call->index, rem_sdp->media_count);
-            call->med_prov_cnt = rem_sdp->media_count;
+	    /* Note that we must not modify the current active media
+	     * (e.g: stop stream, close/cleanup media transport), as if
+	     * SDP nego fails, the current active media should be maintained.
+	     * Also note that our media count should never decrease, even when
+	     * remote removed some media lines.
+	     */
 	    break;
 	}
 
@@ -2989,7 +3054,9 @@ on_check_med_status:
     pj_memcpy(call->media, call->media_prov,
 	      sizeof(call->media_prov[0]) * call->med_prov_cnt);
 
-    /* Perform SDP re-negotiation if needed. */
+    /* Perform SDP re-negotiation if some media have just got disabled
+     * in this function due to media count limit settings.
+     */
     if (got_media && need_renego_sdp) {
 	pjmedia_sdp_neg *neg = call->inv->neg;
 
diff --git a/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.py b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.py
new file mode 100644
index 00000000..6ef94ff4
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.py
@@ -0,0 +1,7 @@
+# $Id$
+#
+import inc_const as const
+
+PJSUA = ["--null-audio --max-calls=1 --auto-answer=200"]
+
+PJSUA_EXPECTS = []
diff --git a/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.xml b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.xml
new file mode 100644
index 00000000..581bd2b2
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.xml
@@ -0,0 +1,226 @@
+<?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             -->
+<!--                                                                    -->
+<!--                                                                    -->
+
+<scenario name="Ticket #1864 scenario 1">
+  <!-- UAC -->
+  
+  <!-- offer with two media lines -->
+  <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=[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=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/SAVP 0
+      c=IN IP4 172.20.73.100
+      a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
+      m=audio 7016 RTP/AVP 0
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" 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@[local_ip]:[local_port]>;tag=[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-Type: application/sdp
+      Content-Length: [len]
+
+      ]]>
+  </send>
+
+
+  <pause milliseconds="1000"/>
+
+  
+  <!-- reoffer with one media line -->
+  <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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/AVP 0
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="200" 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@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[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]
+
+      ]]>
+  </send>
+
+
+  <pause milliseconds="1000"/>
+  
+  
+  <!-- another offer with two media lines -->
+  <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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 INVITE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/SAVP 0
+      c=IN IP4 172.20.73.100
+      a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
+      m=audio 7016 RTP/AVP 0
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="200" 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@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 ACK
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      ]]>
+  </send>
+  
+
+  <pause milliseconds="1000"/>
+  
+  
+  <send>
+    <![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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 4 BYE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+      
+    ]]>
+  </send>
+
+  <recv response="200" rtd="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>
+
diff --git a/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.py b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.py
new file mode 100644
index 00000000..892499aa
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.py
@@ -0,0 +1,7 @@
+# $Id$
+#
+import inc_const as const
+
+PJSUA = ["--null-audio --max-calls=1 --dis-codec=pcma --auto-answer=200"]
+
+PJSUA_EXPECTS = []
diff --git a/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.xml b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.xml
new file mode 100644
index 00000000..be583f86
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.xml
@@ -0,0 +1,227 @@
+<?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             -->
+<!--                                                                    -->
+<!--                                                                    -->
+
+<scenario name="Ticket #1864 scenario 2">
+  <!-- UAC -->
+  
+  <!-- offer with two media lines -->
+  <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=[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=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/SAVP 0
+      c=IN IP4 172.20.73.100
+      a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
+      m=audio 7016 RTP/AVP 0
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" 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@[local_ip]:[local_port]>;tag=[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-Type: application/sdp
+      Content-Length: [len]
+
+      ]]>
+  </send>
+
+
+  <pause milliseconds="1000"/>
+
+  
+  <!-- reoffer with one media line and remote unsupported codec -->
+  <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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 INVITE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/AVP 8
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <!-- remote rejects the offer due to no matching codec -->
+  <recv response="500" 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@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[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]
+
+      ]]>
+  </send>
+
+
+  <pause milliseconds="1000"/>
+  
+  
+  <!-- reoffer with two media lines and supported codec -->
+  <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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 INVITE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
+      s=3cxPS Audio call
+      c=IN IP4 172.20.73.100
+      t=0 0
+      m=audio 7016 RTP/SAVP 0
+      c=IN IP4 172.20.73.100
+      a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
+      m=audio 7016 RTP/AVP 0
+      c=IN IP4 172.20.73.100
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="200" 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@[local_ip]:[local_port]>;tag=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 ACK
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      ]]>
+  </send>
+  
+
+  <pause milliseconds="1000"/>
+  
+  
+  <send>
+    <![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=[call_number]
+      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 4 BYE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+      
+    ]]>
+  </send>
+
+  <recv response="200" rtd="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>
+