Ticket #799: UAC disconnect call when receiving BYE in early state (thanks Gang Liu for the suggestion)

 - backported changes from #798


git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.0@2800 74dad513-b988-da41-8d7b-12977e46ad98

4 files changed, 246 insertions, 4 deletions

diff --git a/pjsip-apps/src/pjsua/pjsua_app.c b/pjsip-apps/src/pjsua/pjsua_app.c
index 95aa3bba..7caf8c2a 100644
--- a/pjsip-apps/src/pjsua/pjsua_app.c
+++ b/pjsip-apps/src/pjsua/pjsua_app.c
@@ -3759,9 +3759,15 @@ void console_app_main(const pj_str_t *uri_to_call)
 	    ui_input_url("Destination URI", buf, sizeof(buf), &result);
 	    if (result.nb_result != NO_NB) {
 
-		if (result.nb_result == -1 || result.nb_result == 0) {
+		if (result.nb_result == -1) {
 		    puts("Sorry you can't do that!");
 		    continue;
+		} else if (result.nb_result == 0) {
+		    uri = NULL;
+		    if (current_call == PJSUA_INVALID_ID) {
+			puts("No current call");
+			continue;
+		    }
 		} else {
 		    pjsua_buddy_info binfo;
 		    pjsua_buddy_get_info(result.nb_result-1, &binfo);
@@ -3772,9 +3778,18 @@ void console_app_main(const pj_str_t *uri_to_call)
 		uri = result.uri_result;
 	    }
 	    
-	    tmp = pj_str(uri);
-
-	    send_request(text, &tmp);
+	    if (uri) {
+		tmp = pj_str(uri);
+		send_request(text, &tmp);
+	    } else {
+		/* If you send call control request using this method
+		 * (such requests includes BYE, CANCEL, etc.), it will
+		 * not go well with the call state, so don't do it
+		 * unless it's for testing.
+		 */
+		pj_str_t method = pj_str(text);
+		pjsua_call_send_request(current_call, &method, NULL);
+	    }
 	    break;
 
 	case 'e':
diff --git a/pjsip/src/pjsip-ua/sip_inv.c b/pjsip/src/pjsip-ua/sip_inv.c
index 0b3ba5ff..e2062ef4 100644
--- a/pjsip/src/pjsip-ua/sip_inv.c
+++ b/pjsip/src/pjsip-ua/sip_inv.c
@@ -3359,6 +3359,22 @@ static void inv_on_state_early( pjsip_inv_session *inv, pjsip_event *e)
 	
 	/* Generic handling for UAC tsx completion */
 	handle_uac_tsx_response(inv, e);
+
+    } else if (tsx->role == PJSIP_ROLE_UAS &&
+	       tsx->method.id == PJSIP_BYE_METHOD &&
+	       tsx->status_code < 200 &&
+	       e->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+    {
+	/* Received BYE before the 2xx/OK response to INVITE.
+	 * Assume that the 2xx/OK response is lost and the BYE
+	 * arrives earlier.
+	 */
+	inv_respond_incoming_bye(inv, tsx, e->body.tsx_state.src.rdata, e);
+
+	/* Set timer just in case we will never get the final response
+	 * for INVITE.
+	 */
+	pjsip_tsx_set_timeout(inv->invite_tsx, 64*pjsip_cfg()->tsx.t1);
     }
 }
 
diff --git a/tests/pjsua/scripts-sipp/uas-early-bye.xml b/tests/pjsua/scripts-sipp/uas-early-bye.xml
new file mode 100644
index 00000000..216b8b45
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-early-bye.xml
@@ -0,0 +1,68 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="Early BYE">
+  <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>
+    <![CDATA[
+      SIP/2.0 100 Trying
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:[$5] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port]
+      From: sipp  <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+      To[$3]
+      Call-ID: [call_id]
+      Cseq: 1 BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+   <recv response="200">
+   </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/uas-forked-200.xml b/tests/pjsua/scripts-sipp/uas-forked-200.xml
new file mode 100644
index 00000000..229f7d89
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-forked-200.xml
@@ -0,0 +1,143 @@
+<?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="Forked INVITE, one of them require PRACK">
+  <recv request="INVITE" crlf="true">
+   <action>
+    <ereg regexp="branch=([0-9a-zA-Z]*)"
+          search_in="hdr"
+	  header="Via"
+          assign_to="1,2"/>
+    <assign assign_to="1" variable="2"/>
+    </action>
+  </recv>
+
+  <send>
+    <![CDATA[
+      SIP/2.0 100 Trying
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+    ]]>
+  </send>
+
+  <!-- Call leg 1: 200/OK -->
+  <send>
+    <![CDATA[
+      SIP/2.0 200 OK
+      Via: SIP/2.0/UDP 127.0.0.1;received=127.0.0.1;rport=5080;branch=[$2]
+      [last_From:]
+      [last_To:];tag=UA_1
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:UA_1@[local_ip]:[local_port]>
+      Content-Type: application/sdp
+
+      v=0
+      o=- 3442013205 3442013205 IN IP4 192.168.0.13
+      s=pjsip
+      c=IN IP4 192.168.0.13
+      t=0 0
+      m=audio 4002 RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+    ]]>
+  </send>
+
+  <!-- Call leg 2: 200/OK -->
+  <send>
+    <![CDATA[
+      SIP/2.0 200 OK
+      Via: SIP/2.0/UDP 127.0.0.1;received=127.0.0.1;rport=5080;branch=[$2]
+      [last_From:]
+      [last_To:];tag=UA_2
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: <sip:UA_2@[local_ip]:[local_port]>
+      Content-Type: application/sdp
+
+      v=0
+      o=- 3442013205 3442013205 IN IP4 192.168.0.13
+      s=pjsip
+      c=IN IP4 192.168.0.13
+      t=0 0
+      m=audio 4002 RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+    ]]>
+  </send>
+
+  <!-- Receive ACK -->
+  <recv request="ACK"
+        optional="false"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <!-- Receive BYE -->
+  <recv request="BYE" crlf="true">
+  </recv>
+
+  <!-- Send 200/OK to BYE -->
+  <send>
+    <![CDATA[
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+    ]]>
+  </send>
+
+
+
+  <!-- Receive ACK -->
+  <recv request="ACK"
+        optional="false"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <!-- Receive BYE -->
+  <recv request="BYE" crlf="true">
+  </recv>
+
+  <!-- Send 200/OK to BYE -->
+  <send>
+    <![CDATA[
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+    ]]>
+  </send>
+
+  <!-- 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>
+