Report disconnection event immediately when hanging up a call

pjsip-apps/src/pjsua/pjsua_app.c

@@ -205,8 +205,11 @@ static void on_call_state(pjsua_call_id call_id, pjsip_event *e)
 	    find_next_call();
 	}
 
-	/* Dump media state upon disconnected */
-	if (1) {
+	/* Dump media state upon disconnected.
+	 * Moved to on_stream_destroyed() since media has been deactivated
+	 * upon disconnection.
+	 */
+	if (0) {
 	    PJ_LOG(5,(THIS_FILE, 
 		      "Call %d disconnected, dumping media stats..", 
 		      call_id));
@@ -271,6 +274,21 @@ static void on_call_state(pjsua_call_id call_id, pjsip_event *e)
     }
 }
 
+/*
+ * Handler when audio stream is destroyed.
+ */
+static void on_stream_destroyed(pjsua_call_id call_id,
+                                pjmedia_stream *strm,
+				unsigned stream_idx)
+{
+    if (1) {
+	PJ_LOG(5,(THIS_FILE, 
+		  "Call %d stream %d destroyed, dumping media stats..", 
+		  call_id, stream_idx));
+	log_call_dump(call_id);
+    }
+}
+
 /**
  * Handler when there is incoming call.
  */
@@ -1317,6 +1335,7 @@ int stdout_refresh_proc(void *arg)
     return 0;
 }
 
+
 static pj_status_t app_init(void)
 {
     pjsua_transport_id transport_id = -1;
@@ -1351,6 +1370,7 @@ static pj_status_t app_init(void)
 
     /* Initialize application callbacks */
     app_config.cfg.cb.on_call_state = &on_call_state;
+    app_config.cfg.cb.on_stream_destroyed = &on_stream_destroyed;
     app_config.cfg.cb.on_call_media_state = &on_call_media_state;
     app_config.cfg.cb.on_incoming_call = &on_incoming_call;
     app_config.cfg.cb.on_dtmf_digit2 = &call_on_dtmf_callback2;

pjsip/include/pjsua-lib/pjsua.h

@@ -5277,9 +5277,11 @@ pjsua_call_send_dtmf_param_default(pjsua_call_send_dtmf_param *param);
 PJ_DECL(unsigned) pjsua_call_get_max_count(void);
 
 /**
- * Get number of currently active calls.
+ * Get the number of current calls. The number includes active calls
+ * (pjsua_call_is_active(call_id) == PJ_TRUE), as well as calls that
+ * are no longer active but still in the process of hanging up.
  *
- * @return		Number of currently active calls.
+ * @return		Number of current calls.
  */
 PJ_DECL(unsigned) pjsua_call_get_count(void);
 
@@ -5557,6 +5559,19 @@ pjsua_call_answer_with_sdp(pjsua_call_id call_id,
  * while #pjsua_call_answer() only works with incoming calls on EARLY
  * state.
  *
+ * After calling this function, media will be deinitialized (call media
+ * callbacks, if any, will still be received) and then, on_call_state()
+ * will be immediately called with state DISCONNECTED. No further
+ * call callbacks will be received after this. The call hangup process
+ * itself (sending BYE, waiting for the response, and resource cleanup)
+ * will continue in the background and the call slot can be reused only
+ * after this process is completed. If application has limited call slots
+ * and would like to check if there are any free slots remaining, it can
+ * query the number of free slots using the APIs:
+ * pjsua_call_get_max_count()-pjsua_call_get_count()
+ *
+ * Note that on_call_tsx_state() will not be called when using this API.
+ *
  * @param call_id	Call identification.
  * @param code		Optional status code to be sent when we're rejecting
  *			incoming call. If the value is zero, "603/Decline"

pjsip/include/pjsua-lib/pjsua_internal.h

@@ -205,6 +205,12 @@ struct pjsua_call
 					    created yet. This temporary 
 					    variable is used to handle such 
 					    case, see ticket #1916.	    */
+
+    pj_timer_entry	 hangup_timer;	/**< Hangup retry timer.	    */
+    unsigned		 hangup_retry;	/**< Number of hangup retries.	    */
+    unsigned		 hangup_code;	/**< Hangup code.	    	    */
+    pj_str_t		 hangup_reason;	/**< Hangup reason.	    	    */
+    pjsua_msg_data	*hangup_msg_data;/**< Hangup message data.	    */
 };
 
 

pjsip/src/pjsua-lib/pjsua_aud.c

@@ -534,7 +534,9 @@ void pjsua_aud_stop_stream(pjsua_call_media *call_med)
 	    call_med->rtp_tx_ts = stat.rtp_tx_last_ts;
 	}
 
-	if (pjsua_var.ua_cfg.cb.on_stream_destroyed) {
+	if (!call_med->call->hanging_up &&
+	    pjsua_var.ua_cfg.cb.on_stream_destroyed)
+	{
 	    pjsua_var.ua_cfg.cb.on_stream_destroyed(call_med->call->index,
 	                                            strm, call_med->idx);
 	}
@@ -558,15 +560,19 @@ void pjsua_aud_stop_stream(pjsua_call_media *call_med)
 static void dtmf_callback(pjmedia_stream *strm, void *user_data,
 			  int digit)
 {
+    pjsua_call_id call_id;
+
     PJ_UNUSED_ARG(strm);
 
+    call_id = (pjsua_call_id)(pj_ssize_t)user_data;
+    if (pjsua_var.calls[call_id].hanging_up)
+    	return;
+
     pj_log_push_indent();
 
     if (pjsua_var.ua_cfg.cb.on_dtmf_digit2) {
-	pjsua_call_id call_id;
 	pjsua_dtmf_info info;
 
-	call_id = (pjsua_call_id)(pj_ssize_t)user_data;
 	info.method = PJSUA_DTMF_METHOD_RFC2833;
 	info.digit = digit;
         info.duration = PJSUA_UNKNOWN_DTMF_DURATION;
@@ -576,9 +582,6 @@ static void dtmf_callback(pjmedia_stream *strm, void *user_data,
 	 * callback, please see ticket #460:
 	 *	http://trac.pjsip.org/repos/ticket/460#comment:4
 	 */    
-	pjsua_call_id call_id;
-
-	call_id = (pjsua_call_id)(pj_ssize_t)user_data;
 	(*pjsua_var.ua_cfg.cb.on_dtmf_digit)(call_id, digit);
     }
 
@@ -596,10 +599,13 @@ static void dtmf_event_callback(pjmedia_stream *strm, void *user_data,
 
     PJ_UNUSED_ARG(strm);
 
+    call_id = (pjsua_call_id)(pj_ssize_t)user_data;
+    if (pjsua_var.calls[call_id].hanging_up)
+    	return;
+
     pj_log_push_indent();
 
     if (pjsua_var.ua_cfg.cb.on_dtmf_event) {
-        call_id = (pjsua_call_id)(pj_ssize_t)user_data;
         evt.method = PJSUA_DTMF_METHOD_RFC2833;
         evt.timestamp = event->timestamp;
         evt.digit = event->digit;
@@ -668,7 +674,7 @@ pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med,
         si->ka_cfg = pjsua_var.acc[call->acc_id].cfg.stream_ka_cfg;
 #endif
 
-        if (pjsua_var.ua_cfg.cb.on_stream_precreate) {
+        if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_stream_precreate) {
             pjsua_on_stream_precreate_param prm;
             prm.stream_idx = strm_idx;
             prm.stream_info.type = PJMEDIA_TYPE_AUDIO;
@@ -707,12 +713,13 @@ pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med,
 	/* If DTMF callback is installed by application, install our
 	 * callback to the session.
 	 */
-        if (pjsua_var.ua_cfg.cb.on_dtmf_event) {
+        if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_dtmf_event) {
             pjmedia_stream_set_dtmf_event_callback(call_med->strm.a.stream,
                                               &dtmf_event_callback,
                                               (void*)(pj_ssize_t)(call->index));
-        } else if (pjsua_var.ua_cfg.cb.on_dtmf_digit || 
-	           pjsua_var.ua_cfg.cb.on_dtmf_digit2) 
+        } else if (!call->hanging_up &&
+        	   (pjsua_var.ua_cfg.cb.on_dtmf_digit || 
+	            pjsua_var.ua_cfg.cb.on_dtmf_digit2))
 	{
 	    pjmedia_stream_set_dtmf_callback(call_med->strm.a.stream,
 					     &dtmf_callback,
@@ -729,7 +736,7 @@ pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med,
 	 * Note: application may modify media_port to point to different
 	 * media port
 	 */
-	if (pjsua_var.ua_cfg.cb.on_stream_created2) {
+	if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_stream_created2) {
 	    pjsua_on_stream_created_param prm;
 
 	    prm.stream = call_med->strm.a.stream;
@@ -741,7 +748,8 @@ pj_status_t pjsua_aud_channel_update(pjsua_call_media *call_med,
 	    call_med->strm.a.destroy_port = prm.destroy_port;
 	    call_med->strm.a.media_port = prm.port;
 
-	} else if (pjsua_var.ua_cfg.cb.on_stream_created) {
+	} else if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_stream_created)
+	{
 	    (*pjsua_var.ua_cfg.cb.on_stream_created)(call->index,
 						  call_med->strm.a.stream,
 						  strm_idx,