* #39226: Switch back to pjsip rev 4710

Rev 4716 introduces errors when building for android (miltiple definitions)
diff --git a/jni/pjproject-android/.svn/pristine/b1/b10001eafe80092647311ff5d9e9e8ee2e012616.svn-base b/jni/pjproject-android/.svn/pristine/b1/b10001eafe80092647311ff5d9e9e8ee2e012616.svn-base
new file mode 100644
index 0000000..5a26c5e
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/b1/b10001eafe80092647311ff5d9e9e8ee2e012616.svn-base
@@ -0,0 +1,94 @@
+/*
+ * replay-database.h
+ *
+ * interface for a replay database for packet security
+ *
+ * David A. McGrew
+ * Cisco Systems, Inc.
+ */
+
+
+#ifndef REPLAY_DB_H
+#define REPLAY_DB_H
+
+#include "integers.h"         /* for uint32_t     */
+#include "datatypes.h"        /* for v128_t       */
+#include "err.h"              /* for err_status_t */
+
+/*
+ * if the ith least significant bit is one, then the packet index
+ * window_end-i is in the database
+ */
+
+typedef struct {
+  uint32_t window_start;   /* packet index of the first bit in bitmask */
+  v128_t bitmask;  
+} rdb_t;
+
+#define rdb_bits_in_bitmask (8*sizeof(v128_t))   
+
+/*
+ * rdb init
+ *
+ * initalizes rdb
+ *
+ * returns err_status_ok on success, err_status_t_fail otherwise
+ */
+
+err_status_t
+rdb_init(rdb_t *rdb);
+
+
+/*
+ * rdb_check
+ *
+ * checks to see if index appears in rdb
+ *
+ * returns err_status_fail if the index already appears in rdb,
+ * returns err_status_ok otherwise
+ */
+
+err_status_t
+rdb_check(const rdb_t *rdb, uint32_t index);  
+
+/*
+ * rdb_add_index
+ *
+ * adds index to rdb_t (and does *not* check if index appears in db)
+ *
+ * returns err_status_ok on success, err_status_fail otherwise
+ *
+ */
+
+err_status_t
+rdb_add_index(rdb_t *rdb, uint32_t index);
+
+/*
+ * the functions rdb_increment() and rdb_get_value() are for use by 
+ * senders, not receivers - DO NOT use these functions on the same
+ * rdb_t upon which rdb_add_index is used!
+ */
+
+
+/*
+ * rdb_increment(db) increments the sequence number in db, if it is 
+ * not too high
+ *
+ * return values:
+ * 
+ *    err_status_ok            no problem
+ *    err_status_key_expired   sequence number too high
+ *
+ */
+err_status_t
+rdb_increment(rdb_t *rdb);
+
+/*
+ * rdb_get_value(db) returns the current sequence number of db
+ */
+
+uint32_t
+rdb_get_value(const rdb_t *rdb);
+
+
+#endif /* REPLAY_DB_H */ 
diff --git a/jni/pjproject-android/.svn/pristine/b1/b1042410eaf91b9477d5c016f344ca352dc788a8.svn-base b/jni/pjproject-android/.svn/pristine/b1/b1042410eaf91b9477d5c016f344ca352dc788a8.svn-base
new file mode 100644
index 0000000..8a4b2d6
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/b1/b1042410eaf91b9477d5c016f344ca352dc788a8.svn-base
@@ -0,0 +1,44 @@
+/* $Id$ */
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * 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 
+ */
+#ifndef __PJ_FIFOBUF_H__
+#define __PJ_FIFOBUF_H__
+
+#include <pj/types.h>
+
+PJ_BEGIN_DECL
+
+typedef struct pj_fifobuf_t pj_fifobuf_t;
+struct pj_fifobuf_t
+{
+    char *first, *last;
+    char *ubegin, *uend;
+    int full;
+};
+
+PJ_DECL(void)	     pj_fifobuf_init (pj_fifobuf_t *fb, void *buffer, unsigned size);
+PJ_DECL(unsigned)    pj_fifobuf_max_size (pj_fifobuf_t *fb);
+PJ_DECL(void*)	     pj_fifobuf_alloc (pj_fifobuf_t *fb, unsigned size);
+PJ_DECL(pj_status_t) pj_fifobuf_unalloc (pj_fifobuf_t *fb, void *buf);
+PJ_DECL(pj_status_t) pj_fifobuf_free (pj_fifobuf_t *fb, void *buf);
+
+PJ_END_DECL
+
+#endif	/* __PJ_FIFOBUF_H__ */
+
diff --git a/jni/pjproject-android/.svn/pristine/b1/b13eac70e231c45431729a6e2c829398c4a936f0.svn-base b/jni/pjproject-android/.svn/pristine/b1/b13eac70e231c45431729a6e2c829398c4a936f0.svn-base
new file mode 100644
index 0000000..6debd13
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/b1/b13eac70e231c45431729a6e2c829398c4a936f0.svn-base
@@ -0,0 +1,106 @@
+<?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="Authorization retry after 1xx response test">
+  <!-- Wait for INVITE request -->
+  <recv request="INVITE" crlf="true">
+  </recv>
+
+  <!-- Send 100 Trying -->
+  <send>
+    <![CDATA[
+
+      SIP/2.0 100 Trying
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+    ]]>
+  </send>
+
+  <!-- Send 180 Ringing (with tag) -->
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+    ]]>
+  </send>
+
+  <!-- Send 401 Unauthorized -->
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 401 Unauthorized
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      WWW-Authenticate: Digest realm="sipp", nonce="1234"
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <!-- Wait for ACK -->
+  <recv request="ACK"
+        optional="false"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <!-- Wait for INVITE retransmission -->
+  <recv request="INVITE" crlf="true">
+  </recv>
+
+  <!-- Send 500 Test Success to terminate the call -->
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 500 Test Success
+      [last_Via:]
+      [last_From:]
+      [last_To:];tag=[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <!-- Wait for ACK -->
+  <recv request="ACK"
+        optional="false"
+        rtd="true"
+        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>
+
diff --git a/jni/pjproject-android/.svn/pristine/b1/b1b377e912ac702726733c387a0730bbf8721e9f.svn-base b/jni/pjproject-android/.svn/pristine/b1/b1b377e912ac702726733c387a0730bbf8721e9f.svn-base
new file mode 100644
index 0000000..57aba6f
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/b1/b1b377e912ac702726733c387a0730bbf8721e9f.svn-base
@@ -0,0 +1,2116 @@
+/* $Id$ */
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * 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 
+ */
+#include "pjsua_app.h"
+
+#define THIS_FILE	"pjsua_app.c"
+
+//#define STEREO_DEMO
+//#define TRANSPORT_ADAPTER_SAMPLE
+//#define HAVE_MULTIPART_TEST
+
+/* Ringtones		    US	       UK  */
+#define RINGBACK_FREQ1	    440	    /* 400 */
+#define RINGBACK_FREQ2	    480	    /* 450 */
+#define RINGBACK_ON	    2000    /* 400 */
+#define RINGBACK_OFF	    4000    /* 200 */
+#define RINGBACK_CNT	    1	    /* 2   */
+#define RINGBACK_INTERVAL   4000    /* 2000 */
+
+#define RING_FREQ1	    800
+#define RING_FREQ2	    640
+#define RING_ON		    200
+#define RING_OFF	    100
+#define RING_CNT	    3
+#define RING_INTERVAL	    3000
+
+#define current_acc	pjsua_acc_get_default()
+
+#ifdef STEREO_DEMO
+static void stereo_demo();
+#endif
+
+static void ringback_start(pjsua_call_id call_id);
+static void ring_start(pjsua_call_id call_id);
+static void ring_stop(pjsua_call_id call_id);
+static pj_status_t app_init();
+static pj_status_t app_destroy();
+
+static pjsua_app_cfg_t app_cfg;
+pj_str_t		    uri_arg;
+pj_bool_t		    app_running	= PJ_FALSE;
+
+/*****************************************************************************
+ * Configuration manipulation
+ */
+
+/*****************************************************************************
+ * Callback 
+ */
+static void ringback_start(pjsua_call_id call_id)
+{
+    if (app_config.no_tones)
+	return;
+
+    if (app_config.call_data[call_id].ringback_on)
+	return;
+
+    app_config.call_data[call_id].ringback_on = PJ_TRUE;
+
+    if (++app_config.ringback_cnt==1 && 
+	app_config.ringback_slot!=PJSUA_INVALID_ID) 
+    {
+	pjsua_conf_connect(app_config.ringback_slot, 0);
+    }
+}
+
+static void ring_stop(pjsua_call_id call_id)
+{
+    if (app_config.no_tones)
+	return;
+
+    if (app_config.call_data[call_id].ringback_on) {
+	app_config.call_data[call_id].ringback_on = PJ_FALSE;
+
+	pj_assert(app_config.ringback_cnt>0);
+	if (--app_config.ringback_cnt == 0 && 
+	    app_config.ringback_slot!=PJSUA_INVALID_ID) 
+	{
+	    pjsua_conf_disconnect(app_config.ringback_slot, 0);
+	    pjmedia_tonegen_rewind(app_config.ringback_port);
+	}
+    }
+
+    if (app_config.call_data[call_id].ring_on) {
+	app_config.call_data[call_id].ring_on = PJ_FALSE;
+
+	pj_assert(app_config.ring_cnt>0);
+	if (--app_config.ring_cnt == 0 && 
+	    app_config.ring_slot!=PJSUA_INVALID_ID) 
+	{
+	    pjsua_conf_disconnect(app_config.ring_slot, 0);
+	    pjmedia_tonegen_rewind(app_config.ring_port);
+	}
+    }
+}
+
+static void ring_start(pjsua_call_id call_id)
+{
+    if (app_config.no_tones)
+	return;
+
+    if (app_config.call_data[call_id].ring_on)
+	return;
+
+    app_config.call_data[call_id].ring_on = PJ_TRUE;
+
+    if (++app_config.ring_cnt==1 && 
+	app_config.ring_slot!=PJSUA_INVALID_ID) 
+    {
+	pjsua_conf_connect(app_config.ring_slot, 0);
+    }
+}
+
+/* Callback from timer when the maximum call duration has been
+ * exceeded.
+ */
+static void call_timeout_callback(pj_timer_heap_t *timer_heap,
+				  struct pj_timer_entry *entry)
+{
+    pjsua_call_id call_id = entry->id;
+    pjsua_msg_data msg_data;
+    pjsip_generic_string_hdr warn;
+    pj_str_t hname = pj_str("Warning");
+    pj_str_t hvalue = pj_str("399 pjsua \"Call duration exceeded\"");
+
+    PJ_UNUSED_ARG(timer_heap);
+
+    if (call_id == PJSUA_INVALID_ID) {
+	PJ_LOG(1,(THIS_FILE, "Invalid call ID in timer callback"));
+	return;
+    }
+    
+    /* Add warning header */
+    pjsua_msg_data_init(&msg_data);
+    pjsip_generic_string_hdr_init2(&warn, &hname, &hvalue);
+    pj_list_push_back(&msg_data.hdr_list, &warn);
+
+    /* Call duration has been exceeded; disconnect the call */
+    PJ_LOG(3,(THIS_FILE, "Duration (%d seconds) has been exceeded "
+			 "for call %d, disconnecting the call",
+			 app_config.duration, call_id));
+    entry->id = PJSUA_INVALID_ID;
+    pjsua_call_hangup(call_id, 200, NULL, &msg_data);
+}
+
+/*
+ * Handler when invite state has changed.
+ */
+static void on_call_state(pjsua_call_id call_id, pjsip_event *e)
+{
+    pjsua_call_info call_info;
+
+    PJ_UNUSED_ARG(e);
+
+    pjsua_call_get_info(call_id, &call_info);
+
+    if (call_info.state == PJSIP_INV_STATE_DISCONNECTED) {
+
+	/* Stop all ringback for this call */
+	ring_stop(call_id);
+
+	/* Cancel duration timer, if any */
+	if (app_config.call_data[call_id].timer.id != PJSUA_INVALID_ID) {
+	    app_call_data *cd = &app_config.call_data[call_id];
+	    pjsip_endpoint *endpt = pjsua_get_pjsip_endpt();
+
+	    cd->timer.id = PJSUA_INVALID_ID;
+	    pjsip_endpt_cancel_timer(endpt, &cd->timer);
+	}
+
+	/* Rewind play file when hangup automatically, 
+	 * since file is not looped
+	 */
+	if (app_config.auto_play_hangup)
+	    pjsua_player_set_pos(app_config.wav_id, 0);
+
+
+	PJ_LOG(3,(THIS_FILE, "Call %d is DISCONNECTED [reason=%d (%s)]", 
+		  call_id,
+		  call_info.last_status,
+		  call_info.last_status_text.ptr));
+
+	if (call_id == current_call) {
+	    find_next_call();
+	}
+
+	/* Dump media state upon disconnected */
+	if (1) {
+	    PJ_LOG(5,(THIS_FILE, 
+		      "Call %d disconnected, dumping media stats..", 
+		      call_id));
+	    log_call_dump(call_id);
+	}
+
+    } else {
+
+	if (app_config.duration != PJSUA_APP_NO_LIMIT_DURATION && 
+	    call_info.state == PJSIP_INV_STATE_CONFIRMED) 
+	{
+	    /* Schedule timer to hangup call after the specified duration */
+	    app_call_data *cd = &app_config.call_data[call_id];
+	    pjsip_endpoint *endpt = pjsua_get_pjsip_endpt();
+	    pj_time_val delay;
+
+	    cd->timer.id = call_id;
+	    delay.sec = app_config.duration;
+	    delay.msec = 0;
+	    pjsip_endpt_schedule_timer(endpt, &cd->timer, &delay);
+	}
+
+	if (call_info.state == PJSIP_INV_STATE_EARLY) {
+	    int code;
+	    pj_str_t reason;
+	    pjsip_msg *msg;
+
+	    /* This can only occur because of TX or RX message */
+	    pj_assert(e->type == PJSIP_EVENT_TSX_STATE);
+
+	    if (e->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
+		msg = e->body.tsx_state.src.rdata->msg_info.msg;
+	    } else {
+		msg = e->body.tsx_state.src.tdata->msg;
+	    }
+
+	    code = msg->line.status.code;
+	    reason = msg->line.status.reason;
+
+	    /* Start ringback for 180 for UAC unless there's SDP in 180 */
+	    if (call_info.role==PJSIP_ROLE_UAC && code==180 && 
+		msg->body == NULL && 
+		call_info.media_status==PJSUA_CALL_MEDIA_NONE) 
+	    {
+		ringback_start(call_id);
+	    }
+
+	    PJ_LOG(3,(THIS_FILE, "Call %d state changed to %s (%d %.*s)", 
+		      call_id, call_info.state_text.ptr,
+		      code, (int)reason.slen, reason.ptr));
+	} else {
+	    PJ_LOG(3,(THIS_FILE, "Call %d state changed to %s", 
+		      call_id,
+		      call_info.state_text.ptr));
+	}
+
+	if (current_call==PJSUA_INVALID_ID)
+	    current_call = call_id;
+
+    }
+}
+
+/**
+ * Handler when there is incoming call.
+ */
+static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
+			     pjsip_rx_data *rdata)
+{
+    pjsua_call_info call_info;
+
+    PJ_UNUSED_ARG(acc_id);
+    PJ_UNUSED_ARG(rdata);
+
+    pjsua_call_get_info(call_id, &call_info);
+
+    if (current_call==PJSUA_INVALID_ID)
+	current_call = call_id;
+
+#ifdef USE_GUI
+    if (!showNotification(call_id))
+	return;
+#endif
+
+    /* Start ringback */
+    ring_start(call_id);
+    
+    if (app_config.auto_answer > 0) {
+	pjsua_call_setting call_opt;
+
+	pjsua_call_setting_default(&call_opt);
+	call_opt.aud_cnt = app_config.aud_cnt;
+	call_opt.vid_cnt = app_config.vid.vid_cnt;
+
+	pjsua_call_answer2(call_id, &call_opt, app_config.auto_answer, NULL,
+			   NULL);
+    }
+    
+    if (app_config.auto_answer < 200) {
+	char notif_st[80] = {0};
+
+#if PJSUA_HAS_VIDEO
+	if (call_info.rem_offerer && call_info.rem_vid_cnt) {
+	    snprintf(notif_st, sizeof(notif_st), 
+		     "To %s the video, type \"vid %s\" first, "
+		     "before answering the call!\n",
+		     (app_config.vid.vid_cnt? "reject":"accept"),
+		     (app_config.vid.vid_cnt? "disable":"enable"));
+	}
+#endif
+
+	PJ_LOG(3,(THIS_FILE,
+		  "Incoming call for account %d!\n"
+		  "Media count: %d audio & %d video\n"
+		  "%s"
+		  "From: %s\n"
+		  "To: %s\n"
+		  "Press %s to answer or %s to reject call",
+		  acc_id,
+		  call_info.rem_aud_cnt,
+		  call_info.rem_vid_cnt,
+		  notif_st,
+		  call_info.remote_info.ptr,
+		  call_info.local_info.ptr,
+		  (app_config.use_cli?"ca a":"a"),
+		  (app_config.use_cli?"g":"h")));
+    }
+}
+
+/*
+ * Handler when a transaction within a call has changed state.
+ */
+static void on_call_tsx_state(pjsua_call_id call_id,
+			      pjsip_transaction *tsx,
+			      pjsip_event *e)
+{
+    const pjsip_method info_method = 
+    {
+	PJSIP_OTHER_METHOD,
+	{ "INFO", 4 }
+    };
+
+    if (pjsip_method_cmp(&tsx->method, &info_method)==0) {
+	/*
+	 * Handle INFO method.
+	 */
+	const pj_str_t STR_APPLICATION = { "application", 11};
+	const pj_str_t STR_DTMF_RELAY  = { "dtmf-relay", 10 };
+	pjsip_msg_body *body = NULL;
+	pj_bool_t dtmf_info = PJ_FALSE;
+	
+	if (tsx->role == PJSIP_ROLE_UAC) {
+	    if (e->body.tsx_state.type == PJSIP_EVENT_TX_MSG)
+		body = e->body.tsx_state.src.tdata->msg->body;
+	    else
+		body = e->body.tsx_state.tsx->last_tx->msg->body;
+	} else {
+	    if (e->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+		body = e->body.tsx_state.src.rdata->msg_info.msg->body;
+	}
+	
+	/* Check DTMF content in the INFO message */
+	if (body && body->len &&
+	    pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
+	    pj_stricmp(&body->content_type.subtype, &STR_DTMF_RELAY)==0)
+	{
+	    dtmf_info = PJ_TRUE;
+	}
+
+	if (dtmf_info && tsx->role == PJSIP_ROLE_UAC && 
+	    (tsx->state == PJSIP_TSX_STATE_COMPLETED ||
+	       (tsx->state == PJSIP_TSX_STATE_TERMINATED &&
+	        e->body.tsx_state.prev_state != PJSIP_TSX_STATE_COMPLETED))) 
+	{
+	    /* Status of outgoing INFO request */
+	    if (tsx->status_code >= 200 && tsx->status_code < 300) {
+		PJ_LOG(4,(THIS_FILE, 
+			  "Call %d: DTMF sent successfully with INFO",
+			  call_id));
+	    } else if (tsx->status_code >= 300) {
+		PJ_LOG(4,(THIS_FILE, 
+			  "Call %d: Failed to send DTMF with INFO: %d/%.*s",
+			  call_id,
+		          tsx->status_code,
+			  (int)tsx->status_text.slen,
+			  tsx->status_text.ptr));
+	    }
+	} else if (dtmf_info && tsx->role == PJSIP_ROLE_UAS &&
+		   tsx->state == PJSIP_TSX_STATE_TRYING)
+	{
+	    /* Answer incoming INFO with 200/OK */
+	    pjsip_rx_data *rdata;
+	    pjsip_tx_data *tdata;
+	    pj_status_t status;
+
+	    rdata = e->body.tsx_state.src.rdata;
+
+	    if (rdata->msg_info.msg->body) {
+		status = pjsip_endpt_create_response(tsx->endpt, rdata,
+						     200, NULL, &tdata);
+		if (status == PJ_SUCCESS)
+		    status = pjsip_tsx_send_msg(tsx, tdata);
+
+		PJ_LOG(3,(THIS_FILE, "Call %d: incoming INFO:\n%.*s", 
+			  call_id,
+			  (int)rdata->msg_info.msg->body->len,
+			  rdata->msg_info.msg->body->data));
+	    } else {
+		status = pjsip_endpt_create_response(tsx->endpt, rdata,
+						     400, NULL, &tdata);
+		if (status == PJ_SUCCESS)
+		    status = pjsip_tsx_send_msg(tsx, tdata);
+	    }
+	}
+    }
+}
+
+/* General processing for media state. "mi" is the media index */
+static void on_call_generic_media_state(pjsua_call_info *ci, unsigned mi,
+                                        pj_bool_t *has_error)
+{
+    const char *status_name[] = {
+        "None",
+        "Active",
+        "Local hold",
+        "Remote hold",
+        "Error"
+    };
+
+    PJ_UNUSED_ARG(has_error);
+
+    pj_assert(ci->media[mi].status <= PJ_ARRAY_SIZE(status_name));
+    pj_assert(PJSUA_CALL_MEDIA_ERROR == 4);
+
+    PJ_LOG(4,(THIS_FILE, "Call %d media %d [type=%s], status is %s",
+	      ci->id, mi, pjmedia_type_name(ci->media[mi].type),
+	      status_name[ci->media[mi].status]));
+}
+
+/* Process audio media state. "mi" is the media index. */
+static void on_call_audio_state(pjsua_call_info *ci, unsigned mi,
+                                pj_bool_t *has_error)
+{
+    PJ_UNUSED_ARG(has_error);
+
+    /* Stop ringback */
+    ring_stop(ci->id);
+
+    /* Connect ports appropriately when media status is ACTIVE or REMOTE HOLD,
+     * otherwise we should NOT connect the ports.
+     */
+    if (ci->media[mi].status == PJSUA_CALL_MEDIA_ACTIVE ||
+	ci->media[mi].status == PJSUA_CALL_MEDIA_REMOTE_HOLD)
+    {
+	pj_bool_t connect_sound = PJ_TRUE;
+	pj_bool_t disconnect_mic = PJ_FALSE;
+	pjsua_conf_port_id call_conf_slot;
+
+	call_conf_slot = ci->media[mi].stream.aud.conf_slot;
+
+	/* Loopback sound, if desired */
+	if (app_config.auto_loop) {
+	    pjsua_conf_connect(call_conf_slot, call_conf_slot);
+	    connect_sound = PJ_FALSE;
+	}
+
+	/* Automatically record conversation, if desired */
+	if (app_config.auto_rec && app_config.rec_port != PJSUA_INVALID_ID) {
+	    pjsua_conf_connect(call_conf_slot, app_config.rec_port);
+	}
+
+	/* Stream a file, if desired */
+	if ((app_config.auto_play || app_config.auto_play_hangup) && 
+	    app_config.wav_port != PJSUA_INVALID_ID)
+	{
+	    pjsua_conf_connect(app_config.wav_port, call_conf_slot);
+	    connect_sound = PJ_FALSE;
+	}
+
+	/* Stream AVI, if desired */
+	if (app_config.avi_auto_play &&
+	    app_config.avi_def_idx != PJSUA_INVALID_ID &&
+	    app_config.avi[app_config.avi_def_idx].slot != PJSUA_INVALID_ID)
+	{
+	    pjsua_conf_connect(app_config.avi[app_config.avi_def_idx].slot,
+			       call_conf_slot);
+	    disconnect_mic = PJ_TRUE;
+	}
+
+	/* Put call in conference with other calls, if desired */
+	if (app_config.auto_conf) {
+	    pjsua_call_id call_ids[PJSUA_MAX_CALLS];
+	    unsigned call_cnt=PJ_ARRAY_SIZE(call_ids);
+	    unsigned i;
+
+	    /* Get all calls, and establish media connection between
+	     * this call and other calls.
+	     */
+	    pjsua_enum_calls(call_ids, &call_cnt);
+
+	    for (i=0; i<call_cnt; ++i) {
+		if (call_ids[i] == ci->id)
+		    continue;
+		
+		if (!pjsua_call_has_media(call_ids[i]))
+		    continue;
+
+		pjsua_conf_connect(call_conf_slot,
+				   pjsua_call_get_conf_port(call_ids[i]));
+		pjsua_conf_connect(pjsua_call_get_conf_port(call_ids[i]),
+		                   call_conf_slot);
+
+		/* Automatically record conversation, if desired */
+		if (app_config.auto_rec && app_config.rec_port !=
+					   PJSUA_INVALID_ID)
+		{
+		    pjsua_conf_connect(pjsua_call_get_conf_port(call_ids[i]), 
+				       app_config.rec_port);
+		}
+
+	    }
+
+	    /* Also connect call to local sound device */
+	    connect_sound = PJ_TRUE;
+	}
+
+	/* Otherwise connect to sound device */
+	if (connect_sound) {
+	    pjsua_conf_connect(call_conf_slot, 0);
+	    if (!disconnect_mic)
+		pjsua_conf_connect(0, call_conf_slot);
+
+	    /* Automatically record conversation, if desired */
+	    if (app_config.auto_rec && app_config.rec_port != PJSUA_INVALID_ID)
+	    {
+		pjsua_conf_connect(call_conf_slot, app_config.rec_port);
+		pjsua_conf_connect(0, app_config.rec_port);
+	    }
+	}
+    }
+}
+
+/* Process video media state. "mi" is the media index. */
+static void on_call_video_state(pjsua_call_info *ci, unsigned mi,
+                                pj_bool_t *has_error)
+{
+    if (ci->media_status != PJSUA_CALL_MEDIA_ACTIVE)
+	return;
+
+    arrange_window(ci->media[mi].stream.vid.win_in);
+
+    PJ_UNUSED_ARG(has_error);
+}
+
+/*
+ * Callback on media state changed event.
+ * The action may connect the call to sound device, to file, or
+ * to loop the call.
+ */
+static void on_call_media_state(pjsua_call_id call_id)
+{
+    pjsua_call_info call_info;
+    unsigned mi;
+    pj_bool_t has_error = PJ_FALSE;
+
+    pjsua_call_get_info(call_id, &call_info);
+
+    for (mi=0; mi<call_info.media_cnt; ++mi) {
+	on_call_generic_media_state(&call_info, mi, &has_error);
+
+	switch (call_info.media[mi].type) {
+	case PJMEDIA_TYPE_AUDIO:
+	    on_call_audio_state(&call_info, mi, &has_error);
+	    break;
+	case PJMEDIA_TYPE_VIDEO:
+	    on_call_video_state(&call_info, mi, &has_error);
+	    break;
+	default:
+	    /* Make gcc happy about enum not handled by switch/case */
+	    break;
+	}
+    }
+
+    if (has_error) {
+	pj_str_t reason = pj_str("Media failed");
+	pjsua_call_hangup(call_id, 500, &reason, NULL);
+    }
+
+#if PJSUA_HAS_VIDEO
+    /* Check if remote has just tried to enable video */
+    if (call_info.rem_offerer && call_info.rem_vid_cnt)
+    {
+	int vid_idx;
+
+	/* Check if there is active video */
+	vid_idx = pjsua_call_get_vid_stream_idx(call_id);
+	if (vid_idx == -1 || call_info.media[vid_idx].dir == PJMEDIA_DIR_NONE) {
+	    PJ_LOG(3,(THIS_FILE,
+		      "Just rejected incoming video offer on call %d, "
+		      "use \"vid call enable %d\" or \"vid call add\" to "
+		      "enable video!", call_id, vid_idx));
+	}
+    }
+#endif
+}
+
+/*
+ * DTMF callback.
+ */
+static void call_on_dtmf_callback(pjsua_call_id call_id, int dtmf)
+{
+    PJ_LOG(3,(THIS_FILE, "Incoming DTMF on call %d: %c", call_id, dtmf));
+}
+
+/*
+ * Redirection handler.
+ */
+static pjsip_redirect_op call_on_redirected(pjsua_call_id call_id, 
+					    const pjsip_uri *target,
+					    const pjsip_event *e)
+{
+    PJ_UNUSED_ARG(e);
+
+    if (app_config.redir_op == PJSIP_REDIRECT_PENDING) {
+	char uristr[PJSIP_MAX_URL_SIZE];
+	int len;
+
+	len = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, target, uristr, 
+			      sizeof(uristr));
+	if (len < 1) {
+	    pj_ansi_strcpy(uristr, "--URI too long--");
+	}
+
+	PJ_LOG(3,(THIS_FILE, "Call %d is being redirected to %.*s. "
+		  "Press 'Ra' to accept+replace To header, 'RA' to accept, "
+		  "'Rr' to reject, or 'Rd' to disconnect.",
+		  call_id, len, uristr));
+    }
+
+    return app_config.redir_op;
+}
+
+/*
+ * Handler registration status has changed.
+ */
+static void on_reg_state(pjsua_acc_id acc_id)
+{
+    PJ_UNUSED_ARG(acc_id);
+
+    // Log already written.
+}
+
+/*
+ * Handler for incoming presence subscription request
+ */
+static void on_incoming_subscribe(pjsua_acc_id acc_id,
+				  pjsua_srv_pres *srv_pres,
+				  pjsua_buddy_id buddy_id,
+				  const pj_str_t *from,
+				  pjsip_rx_data *rdata,
+				  pjsip_status_code *code,
+				  pj_str_t *reason,
+				  pjsua_msg_data *msg_data)
+{
+    /* Just accept the request (the default behavior) */
+    PJ_UNUSED_ARG(acc_id);
+    PJ_UNUSED_ARG(srv_pres);
+    PJ_UNUSED_ARG(buddy_id);
+    PJ_UNUSED_ARG(from);
+    PJ_UNUSED_ARG(rdata);
+    PJ_UNUSED_ARG(code);
+    PJ_UNUSED_ARG(reason);
+    PJ_UNUSED_ARG(msg_data);
+}
+
+
+/*
+ * Handler on buddy state changed.
+ */
+static void on_buddy_state(pjsua_buddy_id buddy_id)
+{
+    pjsua_buddy_info info;
+    pjsua_buddy_get_info(buddy_id, &info);
+
+    PJ_LOG(3,(THIS_FILE, "%.*s status is %.*s, subscription state is %s "
+			 "(last termination reason code=%d %.*s)",
+	      (int)info.uri.slen,
+	      info.uri.ptr,
+	      (int)info.status_text.slen,
+	      info.status_text.ptr,
+	      info.sub_state_name,
+	      info.sub_term_code,
+	      (int)info.sub_term_reason.slen,
+	      info.sub_term_reason.ptr));
+}
+
+
+/*
+ * Subscription state has changed.
+ */
+static void on_buddy_evsub_state(pjsua_buddy_id buddy_id,
+				 pjsip_evsub *sub,
+				 pjsip_event *event)
+{
+    char event_info[80];
+
+    PJ_UNUSED_ARG(sub);
+
+    event_info[0] = '\0';
+
+    if (event->type == PJSIP_EVENT_TSX_STATE &&
+	    event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+    {
+	pjsip_rx_data *rdata = event->body.tsx_state.src.rdata;
+	snprintf(event_info, sizeof(event_info),
+		 " (RX %s)",
+		 pjsip_rx_data_get_info(rdata));
+    }
+
+    PJ_LOG(4,(THIS_FILE,
+	      "Buddy %d: subscription state: %s (event: %s%s)",
+	      buddy_id, pjsip_evsub_get_state_name(sub),
+	      pjsip_event_str(event->type),
+	      event_info));
+
+}
+
+
+/**
+ * Incoming IM message (i.e. MESSAGE request)!
+ */
+static void on_pager(pjsua_call_id call_id, const pj_str_t *from, 
+		     const pj_str_t *to, const pj_str_t *contact,
+		     const pj_str_t *mime_type, const pj_str_t *text)
+{
+    /* Note: call index may be -1 */
+    PJ_UNUSED_ARG(call_id);
+    PJ_UNUSED_ARG(to);
+    PJ_UNUSED_ARG(contact);
+    PJ_UNUSED_ARG(mime_type);
+
+    PJ_LOG(3,(THIS_FILE,"MESSAGE from %.*s: %.*s (%.*s)",
+	      (int)from->slen, from->ptr,
+	      (int)text->slen, text->ptr,
+	      (int)mime_type->slen, mime_type->ptr));
+}
+
+
+/**
+ * Received typing indication
+ */
+static void on_typing(pjsua_call_id call_id, const pj_str_t *from,
+		      const pj_str_t *to, const pj_str_t *contact,
+		      pj_bool_t is_typing)
+{
+    PJ_UNUSED_ARG(call_id);
+    PJ_UNUSED_ARG(to);
+    PJ_UNUSED_ARG(contact);
+
+    PJ_LOG(3,(THIS_FILE, "IM indication: %.*s %s",
+	      (int)from->slen, from->ptr,
+	      (is_typing?"is typing..":"has stopped typing")));
+}
+
+
+/**
+ * Call transfer request status.
+ */
+static void on_call_transfer_status(pjsua_call_id call_id,
+				    int status_code,
+				    const pj_str_t *status_text,
+				    pj_bool_t final,
+				    pj_bool_t *p_cont)
+{
+    PJ_LOG(3,(THIS_FILE, "Call %d: transfer status=%d (%.*s) %s",
+	      call_id, status_code,
+	      (int)status_text->slen, status_text->ptr,
+	      (final ? "[final]" : "")));
+
+    if (status_code/100 == 2) {
+	PJ_LOG(3,(THIS_FILE, 
+	          "Call %d: call transfered successfully, disconnecting call",
+		  call_id));
+	pjsua_call_hangup(call_id, PJSIP_SC_GONE, NULL, NULL);
+	*p_cont = PJ_FALSE;
+    }
+}
+
+
+/*
+ * Notification that call is being replaced.
+ */
+static void on_call_replaced(pjsua_call_id old_call_id,
+			     pjsua_call_id new_call_id)
+{
+    pjsua_call_info old_ci, new_ci;
+
+    pjsua_call_get_info(old_call_id, &old_ci);
+    pjsua_call_get_info(new_call_id, &new_ci);
+
+    PJ_LOG(3,(THIS_FILE, "Call %d with %.*s is being replaced by "
+			 "call %d with %.*s",
+			 old_call_id, 
+			 (int)old_ci.remote_info.slen, old_ci.remote_info.ptr,
+			 new_call_id,
+			 (int)new_ci.remote_info.slen, new_ci.remote_info.ptr));
+}
+
+
+/*
+ * NAT type detection callback.
+ */
+static void on_nat_detect(const pj_stun_nat_detect_result *res)
+{
+    if (res->status != PJ_SUCCESS) {
+	pjsua_perror(THIS_FILE, "NAT detection failed", res->status);
+    } else {
+	PJ_LOG(3, (THIS_FILE, "NAT detected as %s", res->nat_type_name));
+    }
+}
+
+
+/*
+ * MWI indication
+ */
+static void on_mwi_info(pjsua_acc_id acc_id, pjsua_mwi_info *mwi_info)
+{
+    pj_str_t body;
+    
+    PJ_LOG(3,(THIS_FILE, "Received MWI for acc %d:", acc_id));
+
+    if (mwi_info->rdata->msg_info.ctype) {
+	const pjsip_ctype_hdr *ctype = mwi_info->rdata->msg_info.ctype;
+
+	PJ_LOG(3,(THIS_FILE, " Content-Type: %.*s/%.*s",
+	          (int)ctype->media.type.slen,
+		  ctype->media.type.ptr,
+		  (int)ctype->media.subtype.slen,
+		  ctype->media.subtype.ptr));
+    }
+
+    if (!mwi_info->rdata->msg_info.msg->body) {
+	PJ_LOG(3,(THIS_FILE, "  no message body"));
+	return;
+    }
+
+    body.ptr = (char *)mwi_info->rdata->msg_info.msg->body->data;
+    body.slen = mwi_info->rdata->msg_info.msg->body->len;
+
+    PJ_LOG(3,(THIS_FILE, " Body:\n%.*s", (int)body.slen, body.ptr));
+}
+
+
+/*
+ * Transport status notification
+ */
+static void on_transport_state(pjsip_transport *tp, 
+			       pjsip_transport_state state,
+			       const pjsip_transport_state_info *info)
+{
+    char host_port[128];
+
+    pj_ansi_snprintf(host_port, sizeof(host_port), "[%.*s:%d]",
+		     (int)tp->remote_name.host.slen,
+		     tp->remote_name.host.ptr,
+		     tp->remote_name.port);
+
+    switch (state) {
+    case PJSIP_TP_STATE_CONNECTED:
+	{
+	    PJ_LOG(3,(THIS_FILE, "SIP %s transport is connected to %s",
+		     tp->type_name, host_port));
+	}
+	break;
+
+    case PJSIP_TP_STATE_DISCONNECTED:
+	{
+	    char buf[100];
+
+	    snprintf(buf, sizeof(buf), "SIP %s transport is disconnected "
+		    "from %s", tp->type_name, host_port);
+	    pjsua_perror(THIS_FILE, buf, info->status);
+	}
+	break;
+
+    default:
+	break;
+    }
+
+#if defined(PJSIP_HAS_TLS_TRANSPORT) && PJSIP_HAS_TLS_TRANSPORT!=0
+
+    if (!pj_ansi_stricmp(tp->type_name, "tls") && info->ext_info &&
+	(state == PJSIP_TP_STATE_CONNECTED || 
+	 ((pjsip_tls_state_info*)info->ext_info)->
+			         ssl_sock_info->verify_status != PJ_SUCCESS))
+    {
+	pjsip_tls_state_info *tls_info = (pjsip_tls_state_info*)info->ext_info;
+	pj_ssl_sock_info *ssl_sock_info = tls_info->ssl_sock_info;
+	char buf[2048];
+	const char *verif_msgs[32];
+	unsigned verif_msg_cnt;
+
+	/* Dump server TLS cipher */
+	PJ_LOG(4,(THIS_FILE, "TLS cipher used: 0x%06X/%s",
+		  ssl_sock_info->cipher,
+		  pj_ssl_cipher_name(ssl_sock_info->cipher) ));
+
+	/* Dump server TLS certificate */
+	pj_ssl_cert_info_dump(ssl_sock_info->remote_cert_info, "  ",
+			      buf, sizeof(buf));
+	PJ_LOG(4,(THIS_FILE, "TLS cert info of %s:\n%s", host_port, buf));
+
+	/* Dump server TLS certificate verification result */
+	verif_msg_cnt = PJ_ARRAY_SIZE(verif_msgs);
+	pj_ssl_cert_get_verify_status_strings(ssl_sock_info->verify_status,
+					      verif_msgs, &verif_msg_cnt);
+	PJ_LOG(3,(THIS_FILE, "TLS cert verification result of %s : %s",
+			     host_port,
+			     (verif_msg_cnt == 1? verif_msgs[0]:"")));
+	if (verif_msg_cnt > 1) {
+	    unsigned i;
+	    for (i = 0; i < verif_msg_cnt; ++i)
+		PJ_LOG(3,(THIS_FILE, "- %s", verif_msgs[i]));
+	}
+
+	if (ssl_sock_info->verify_status &&
+	    !app_config.udp_cfg.tls_setting.verify_server) 
+	{
+	    PJ_LOG(3,(THIS_FILE, "PJSUA is configured to ignore TLS cert "
+				 "verification errors"));
+	}
+    }
+
+#endif
+
+}
+
+/*
+ * Notification on ICE error.
+ */
+static void on_ice_transport_error(int index, pj_ice_strans_op op,
+				   pj_status_t status, void *param)
+{
+    PJ_UNUSED_ARG(op);
+    PJ_UNUSED_ARG(param);
+    PJ_PERROR(1,(THIS_FILE, status,
+	         "ICE keep alive failure for transport %d", index));
+}
+
+/*
+ * Notification on sound device operation.
+ */
+static pj_status_t on_snd_dev_operation(int operation)
+{
+    PJ_LOG(3,(THIS_FILE, "Turning sound device %s", (operation? "ON":"OFF")));
+    return PJ_SUCCESS;
+}
+
+/* Callback on media events */
+static void on_call_media_event(pjsua_call_id call_id,
+                                unsigned med_idx,
+                                pjmedia_event *event)
+{
+    char event_name[5];
+
+    PJ_LOG(5,(THIS_FILE, "Event %s",
+	      pjmedia_fourcc_name(event->type, event_name)));
+
+#if PJSUA_HAS_VIDEO
+    if (event->type == PJMEDIA_EVENT_FMT_CHANGED) {
+	/* Adjust renderer window size to original video size */
+	pjsua_call_info ci;
+	pjsua_vid_win_id wid;
+	pjmedia_rect_size size;
+
+	pjsua_call_get_info(call_id, &ci);
+
+	if ((ci.media[med_idx].type == PJMEDIA_TYPE_VIDEO) &&
+	    (ci.media[med_idx].dir & PJMEDIA_DIR_DECODING))
+	{
+	    wid = ci.media[med_idx].stream.vid.win_in;
+	    size = event->data.fmt_changed.new_fmt.det.vid.size;
+	    pjsua_vid_win_set_size(wid, &size);
+	}
+
+	/* Re-arrange video windows */
+	arrange_window(PJSUA_INVALID_ID);
+    }
+#else
+    PJ_UNUSED_ARG(call_id);
+    PJ_UNUSED_ARG(med_idx);
+    PJ_UNUSED_ARG(event);
+#endif
+}
+
+#ifdef TRANSPORT_ADAPTER_SAMPLE
+/*
+ * This callback is called when media transport needs to be created.
+ */
+static pjmedia_transport* on_create_media_transport(pjsua_call_id call_id,
+						    unsigned media_idx,
+						    pjmedia_transport *base_tp,
+						    unsigned flags)
+{
+    pjmedia_transport *adapter;
+    pj_status_t status;
+
+    /* Create the adapter */
+    status = pjmedia_tp_adapter_create(pjsua_get_pjmedia_endpt(),
+                                       NULL, base_tp,
+                                       (flags & PJSUA_MED_TP_CLOSE_MEMBER),
+                                       &adapter);
+    if (status != PJ_SUCCESS) {
+	PJ_PERROR(1,(THIS_FILE, status, "Error creating adapter"));
+	return NULL;
+    }
+
+    PJ_LOG(3,(THIS_FILE, "Media transport is created for call %d media %d",
+	      call_id, media_idx));
+
+    return adapter;
+}
+#endif
+
+/* Playfile done notification, set timer to hangup calls */
+pj_status_t on_playfile_done(pjmedia_port *port, void *usr_data)
+{
+    pj_time_val delay;
+
+    PJ_UNUSED_ARG(port);
+    PJ_UNUSED_ARG(usr_data);
+
+    /* Just rewind WAV when it is played outside of call */
+    if (pjsua_call_get_count() == 0) {
+	pjsua_player_set_pos(app_config.wav_id, 0);
+	return PJ_SUCCESS;
+    }
+
+    /* Timer is already active */
+    if (app_config.auto_hangup_timer.id == 1)
+	return PJ_SUCCESS;
+
+    app_config.auto_hangup_timer.id = 1;
+    delay.sec = 0;
+    delay.msec = 200; /* Give 200 ms before hangup */
+    pjsip_endpt_schedule_timer(pjsua_get_pjsip_endpt(), 
+			       &app_config.auto_hangup_timer, 
+			       &delay);
+
+    return PJ_SUCCESS;
+}
+
+/* Auto hangup timer callback */
+static void hangup_timeout_callback(pj_timer_heap_t *timer_heap,
+				    struct pj_timer_entry *entry)
+{
+    PJ_UNUSED_ARG(timer_heap);
+    PJ_UNUSED_ARG(entry);
+
+    app_config.auto_hangup_timer.id = 0;
+    pjsua_call_hangup_all();
+}
+
+/*
+ * A simple registrar, invoked by default_mod_on_rx_request()
+ */
+static void simple_registrar(pjsip_rx_data *rdata)
+{
+    pjsip_tx_data *tdata;
+    const pjsip_expires_hdr *exp;
+    const pjsip_hdr *h;
+    unsigned cnt = 0;
+    pjsip_generic_string_hdr *srv;
+    pj_status_t status;
+
+    status = pjsip_endpt_create_response(pjsua_get_pjsip_endpt(),
+					 rdata, 200, NULL, &tdata);
+    if (status != PJ_SUCCESS)
+    return;
+
+    exp = (pjsip_expires_hdr *)pjsip_msg_find_hdr(rdata->msg_info.msg, 
+						  PJSIP_H_EXPIRES, NULL);
+
+    h = rdata->msg_info.msg->hdr.next;
+    while (h != &rdata->msg_info.msg->hdr) {
+	if (h->type == PJSIP_H_CONTACT) {
+	    const pjsip_contact_hdr *c = (const pjsip_contact_hdr*)h;
+	    int e = c->expires;
+
+	    if (e < 0) {
+		if (exp)
+		    e = exp->ivalue;
+		else
+		    e = 3600;
+	    }
+
+	    if (e > 0) {
+		pjsip_contact_hdr *nc = (pjsip_contact_hdr *)pjsip_hdr_clone(
+								tdata->pool, h);
+		nc->expires = e;
+		pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)nc);
+		++cnt;
+	    }
+	}
+	h = h->next;
+    }
+
+    srv = pjsip_generic_string_hdr_create(tdata->pool, NULL, NULL);
+    srv->name = pj_str("Server");
+    srv->hvalue = pj_str("pjsua simple registrar");
+    pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)srv);
+
+    pjsip_endpt_send_response2(pjsua_get_pjsip_endpt(),
+		       rdata, tdata, NULL, NULL);
+}
+
+/*****************************************************************************
+ * A simple module to handle otherwise unhandled request. We will register
+ * this with the lowest priority.
+ */
+
+/* Notification on incoming request */
+static pj_bool_t default_mod_on_rx_request(pjsip_rx_data *rdata)
+{
+    pjsip_tx_data *tdata;
+    pjsip_status_code status_code;
+    pj_status_t status;
+
+    /* Don't respond to ACK! */
+    if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method,
+			 &pjsip_ack_method) == 0)
+	return PJ_TRUE;
+
+    /* Simple registrar */
+    if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method,
+                         &pjsip_register_method) == 0)
+    {
+	simple_registrar(rdata);
+	return PJ_TRUE;
+    }
+
+    /* Create basic response. */
+    if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, 
+			 &pjsip_notify_method) == 0)
+    {
+	/* Unsolicited NOTIFY's, send with Bad Request */
+	status_code = PJSIP_SC_BAD_REQUEST;
+    } else {
+	/* Probably unknown method */
+	status_code = PJSIP_SC_METHOD_NOT_ALLOWED;
+    }
+    status = pjsip_endpt_create_response(pjsua_get_pjsip_endpt(), 
+					 rdata, status_code, 
+					 NULL, &tdata);
+    if (status != PJ_SUCCESS) {
+	pjsua_perror(THIS_FILE, "Unable to create response", status);
+	return PJ_TRUE;
+    }
+
+    /* Add Allow if we're responding with 405 */
+    if (status_code == PJSIP_SC_METHOD_NOT_ALLOWED) {
+	const pjsip_hdr *cap_hdr;
+	cap_hdr = pjsip_endpt_get_capability(pjsua_get_pjsip_endpt(), 
+					     PJSIP_H_ALLOW, NULL);
+	if (cap_hdr) {
+	    pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)pjsip_hdr_clone(
+							 tdata->pool, cap_hdr));
+	}
+    }
+
+    /* Add User-Agent header */
+    {
+	pj_str_t user_agent;
+	char tmp[80];
+	const pj_str_t USER_AGENT = { "User-Agent", 10};
+	pjsip_hdr *h;
+
+	pj_ansi_snprintf(tmp, sizeof(tmp), "PJSUA v%s/%s", 
+			 pj_get_version(), PJ_OS_NAME);
+	pj_strdup2_with_null(tdata->pool, &user_agent, tmp);
+
+	h = (pjsip_hdr*) pjsip_generic_string_hdr_create(tdata->pool,
+							 &USER_AGENT,
+							 &user_agent);
+	pjsip_msg_add_hdr(tdata->msg, h);
+    }
+
+    pjsip_endpt_send_response2(pjsua_get_pjsip_endpt(), rdata, tdata, 
+			       NULL, NULL);
+
+    return PJ_TRUE;
+}
+
+/* The module instance. */
+static pjsip_module mod_default_handler = 
+{
+    NULL, NULL,				/* prev, next.		*/
+    { "mod-default-handler", 19 },	/* Name.		*/
+    -1,					/* Id			*/
+    PJSIP_MOD_PRIORITY_APPLICATION+99,	/* Priority	        */
+    NULL,				/* load()		*/
+    NULL,				/* start()		*/
+    NULL,				/* stop()		*/
+    NULL,				/* unload()		*/
+    &default_mod_on_rx_request,		/* on_rx_request()	*/
+    NULL,				/* on_rx_response()	*/
+    NULL,				/* on_tx_request.	*/
+    NULL,				/* on_tx_response()	*/
+    NULL,				/* on_tsx_state()	*/
+
+};
+
+/** CLI callback **/
+
+/* Called on CLI (re)started, e.g: initial start, after iOS bg */
+void cli_on_started(pj_status_t status)
+{
+    /* Notify app */
+    if (app_cfg.on_started) {
+	if (status == PJ_SUCCESS) {
+	    char info[128];
+	    cli_get_info(info, sizeof(info));
+	    if (app_cfg.on_started) {
+		(*app_cfg.on_started)(status, info);		
+	    } 
+	} else {
+	    if (app_cfg.on_started) {
+		(*app_cfg.on_started)(status, NULL);
+	    } 		
+	}
+    }
+}
+
+/* Called on CLI quit */
+void cli_on_stopped(pj_bool_t restart, int argc, char* argv[])
+{
+    /* Notify app */
+    if (app_cfg.on_stopped)
+	(*app_cfg.on_stopped)(restart, argc, argv);
+}
+
+
+/* Called on pjsua legacy quit */
+void legacy_on_stopped(pj_bool_t restart)
+{
+    /* Notify app */
+    if (app_cfg.on_stopped)
+	(*app_cfg.on_stopped)(restart, 1, NULL);
+}
+
+/*****************************************************************************
+ * Public API
+ */
+
+int stdout_refresh_proc(void *arg)
+{
+    extern char *stdout_refresh_text;
+
+    PJ_UNUSED_ARG(arg);
+
+    /* Set thread to lowest priority so that it doesn't clobber
+     * stdout output
+     */
+    pj_thread_set_prio(pj_thread_this(), 
+		       pj_thread_get_prio_min(pj_thread_this()));
+
+    while (!stdout_refresh_quit) {
+	pj_thread_sleep(stdout_refresh * 1000);
+	puts(stdout_refresh_text);
+	fflush(stdout);
+    }
+
+    return 0;
+}
+
+static pj_status_t app_init()
+{
+    pjsua_transport_id transport_id = -1;
+    pjsua_transport_config tcp_cfg;
+    unsigned i;
+    pj_pool_t *tmp_pool;
+    pj_status_t status;
+
+    /** Create pjsua **/
+    status = pjsua_create();
+    if (status != PJ_SUCCESS)
+	return status;
+
+    /* Create pool for application */
+    app_config.pool = pjsua_pool_create("pjsua-app", 1000, 1000);
+    tmp_pool = pjsua_pool_create("tmp-pjsua", 1000, 1000);;
+
+    /* Init CLI & its FE settings */
+    if (!app_running) {
+	pj_cli_cfg_default(&app_config.cli_cfg.cfg);
+	pj_cli_telnet_cfg_default(&app_config.cli_cfg.telnet_cfg);
+	pj_cli_console_cfg_default(&app_config.cli_cfg.console_cfg);
+	app_config.cli_cfg.telnet_cfg.on_started = cli_on_started;
+    }
+
+    /** Parse args **/
+    status = load_config(app_cfg.argc, app_cfg.argv, &uri_arg);
+    if (status != PJ_SUCCESS) {
+	pj_pool_release(tmp_pool);
+	return status;
+    }
+
+    /* Initialize application callbacks */
+    app_config.cfg.cb.on_call_state = &on_call_state;
+    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_call_tsx_state = &on_call_tsx_state;
+    app_config.cfg.cb.on_dtmf_digit = &call_on_dtmf_callback;
+    app_config.cfg.cb.on_call_redirected = &call_on_redirected;
+    app_config.cfg.cb.on_reg_state = &on_reg_state;
+    app_config.cfg.cb.on_incoming_subscribe = &on_incoming_subscribe;
+    app_config.cfg.cb.on_buddy_state = &on_buddy_state;
+    app_config.cfg.cb.on_buddy_evsub_state = &on_buddy_evsub_state;
+    app_config.cfg.cb.on_pager = &on_pager;
+    app_config.cfg.cb.on_typing = &on_typing;
+    app_config.cfg.cb.on_call_transfer_status = &on_call_transfer_status;
+    app_config.cfg.cb.on_call_replaced = &on_call_replaced;
+    app_config.cfg.cb.on_nat_detect = &on_nat_detect;
+    app_config.cfg.cb.on_mwi_info = &on_mwi_info;
+    app_config.cfg.cb.on_transport_state = &on_transport_state;
+    app_config.cfg.cb.on_ice_transport_error = &on_ice_transport_error;
+    app_config.cfg.cb.on_snd_dev_operation = &on_snd_dev_operation;
+    app_config.cfg.cb.on_call_media_event = &on_call_media_event;
+#ifdef TRANSPORT_ADAPTER_SAMPLE
+    app_config.cfg.cb.on_create_media_transport = &on_create_media_transport;
+#endif
+
+    /* Set sound device latency */
+    if (app_config.capture_lat > 0)
+	app_config.media_cfg.snd_rec_latency = app_config.capture_lat;
+    if (app_config.playback_lat)
+	app_config.media_cfg.snd_play_latency = app_config.playback_lat;
+
+    if (app_cfg.on_config_init)
+	(*app_cfg.on_config_init)(&app_config);
+
+    /* Initialize pjsua */
+    status = pjsua_init(&app_config.cfg, &app_config.log_cfg,
+			&app_config.media_cfg);
+    if (status != PJ_SUCCESS) {
+	pj_pool_release(tmp_pool);
+	return status;
+    }
+
+    /* Initialize our module to handle otherwise unhandled request */
+    status = pjsip_endpt_register_module(pjsua_get_pjsip_endpt(),
+					 &mod_default_handler);
+    if (status != PJ_SUCCESS)
+	return status;
+
+#ifdef STEREO_DEMO
+    stereo_demo();
+#endif
+
+    /* Initialize calls data */
+    for (i=0; i<PJ_ARRAY_SIZE(app_config.call_data); ++i) {
+	app_config.call_data[i].timer.id = PJSUA_INVALID_ID;
+	app_config.call_data[i].timer.cb = &call_timeout_callback;
+    }
+
+    /* Optionally registers WAV file */
+    for (i=0; i<app_config.wav_count; ++i) {
+	pjsua_player_id wav_id;
+	unsigned play_options = 0;
+
+	if (app_config.auto_play_hangup)
+	    play_options |= PJMEDIA_FILE_NO_LOOP;
+
+	status = pjsua_player_create(&app_config.wav_files[i], play_options, 
+				     &wav_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	if (app_config.wav_id == PJSUA_INVALID_ID) {
+	    app_config.wav_id = wav_id;
+	    app_config.wav_port = pjsua_player_get_conf_port(app_config.wav_id);
+	    if (app_config.auto_play_hangup) {
+		pjmedia_port *port;
+
+		pjsua_player_get_port(app_config.wav_id, &port);
+		status = pjmedia_wav_player_set_eof_cb(port, NULL, 
+						       &on_playfile_done);
+		if (status != PJ_SUCCESS)
+		    goto on_error;
+
+		pj_timer_entry_init(&app_config.auto_hangup_timer, 0, NULL, 
+				    &hangup_timeout_callback);
+	    }
+	}
+    }
+
+    /* Optionally registers tone players */
+    for (i=0; i<app_config.tone_count; ++i) {
+	pjmedia_port *tport;
+	char name[80];
+	pj_str_t label;
+	pj_status_t status;
+
+	pj_ansi_snprintf(name, sizeof(name), "tone-%d,%d",
+			 app_config.tones[i].freq1, 
+			 app_config.tones[i].freq2);
+	label = pj_str(name);
+	status = pjmedia_tonegen_create2(app_config.pool, &label,
+					 8000, 1, 160, 16, 
+					 PJMEDIA_TONEGEN_LOOP,  &tport);
+	if (status != PJ_SUCCESS) {
+	    pjsua_perror(THIS_FILE, "Unable to create tone generator", status);
+	    goto on_error;
+	}
+
+	status = pjsua_conf_add_port(app_config.pool, tport, 
+				     &app_config.tone_slots[i]);
+	pj_assert(status == PJ_SUCCESS);
+
+	status = pjmedia_tonegen_play(tport, 1, &app_config.tones[i], 0);
+	pj_assert(status == PJ_SUCCESS);
+    }
+
+    /* Optionally create recorder file, if any. */
+    if (app_config.rec_file.slen) {
+	status = pjsua_recorder_create(&app_config.rec_file, 0, NULL, 0, 0,
+				       &app_config.rec_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	app_config.rec_port = pjsua_recorder_get_conf_port(app_config.rec_id);
+    }
+
+    pj_memcpy(&tcp_cfg, &app_config.udp_cfg, sizeof(tcp_cfg));
+
+    /* Create ringback tones */
+    if (app_config.no_tones == PJ_FALSE) {
+	unsigned i, samples_per_frame;
+	pjmedia_tone_desc tone[RING_CNT+RINGBACK_CNT];
+	pj_str_t name;
+
+	samples_per_frame = app_config.media_cfg.audio_frame_ptime * 
+			    app_config.media_cfg.clock_rate *
+			    app_config.media_cfg.channel_count / 1000;
+
+	/* Ringback tone (call is ringing) */
+	name = pj_str("ringback");
+	status = pjmedia_tonegen_create2(app_config.pool, &name, 
+					 app_config.media_cfg.clock_rate,
+					 app_config.media_cfg.channel_count, 
+					 samples_per_frame,
+					 16, PJMEDIA_TONEGEN_LOOP, 
+					 &app_config.ringback_port);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	pj_bzero(&tone, sizeof(tone));
+	for (i=0; i<RINGBACK_CNT; ++i) {
+	    tone[i].freq1 = RINGBACK_FREQ1;
+	    tone[i].freq2 = RINGBACK_FREQ2;
+	    tone[i].on_msec = RINGBACK_ON;
+	    tone[i].off_msec = RINGBACK_OFF;
+	}
+	tone[RINGBACK_CNT-1].off_msec = RINGBACK_INTERVAL;
+
+	pjmedia_tonegen_play(app_config.ringback_port, RINGBACK_CNT, tone,
+			     PJMEDIA_TONEGEN_LOOP);
+
+
+	status = pjsua_conf_add_port(app_config.pool, app_config.ringback_port,
+				     &app_config.ringback_slot);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Ring (to alert incoming call) */
+	name = pj_str("ring");
+	status = pjmedia_tonegen_create2(app_config.pool, &name, 
+					 app_config.media_cfg.clock_rate,
+					 app_config.media_cfg.channel_count, 
+					 samples_per_frame,
+					 16, PJMEDIA_TONEGEN_LOOP, 
+					 &app_config.ring_port);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	for (i=0; i<RING_CNT; ++i) {
+	    tone[i].freq1 = RING_FREQ1;
+	    tone[i].freq2 = RING_FREQ2;
+	    tone[i].on_msec = RING_ON;
+	    tone[i].off_msec = RING_OFF;
+	}
+	tone[RING_CNT-1].off_msec = RING_INTERVAL;
+
+	pjmedia_tonegen_play(app_config.ring_port, RING_CNT, 
+			     tone, PJMEDIA_TONEGEN_LOOP);
+
+	status = pjsua_conf_add_port(app_config.pool, app_config.ring_port,
+				     &app_config.ring_slot);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+    }
+
+    /* Create AVI player virtual devices */
+    if (app_config.avi_cnt) {
+#if PJMEDIA_HAS_VIDEO && PJMEDIA_VIDEO_DEV_HAS_AVI
+	pjmedia_vid_dev_factory *avi_factory;
+
+	status = pjmedia_avi_dev_create_factory(pjsua_get_pool_factory(),
+	                                        app_config.avi_cnt,
+	                                        &avi_factory);
+	if (status != PJ_SUCCESS) {
+	    PJ_PERROR(1,(THIS_FILE, status, "Error creating AVI factory"));
+	    goto on_error;
+	}
+
+	for (i=0; i<app_config.avi_cnt; ++i) {
+	    pjmedia_avi_dev_param avdp;
+	    pjmedia_vid_dev_index avid;
+	    unsigned strm_idx, strm_cnt;
+
+	    app_config.avi[i].dev_id = PJMEDIA_VID_INVALID_DEV;
+	    app_config.avi[i].slot = PJSUA_INVALID_ID;
+
+	    pjmedia_avi_dev_param_default(&avdp);
+	    avdp.path = app_config.avi[i].path;
+
+	    status =  pjmedia_avi_dev_alloc(avi_factory, &avdp, &avid);
+	    if (status != PJ_SUCCESS) {
+		PJ_PERROR(1,(THIS_FILE, status,
+			     "Error creating AVI player for %.*s",
+			     (int)avdp.path.slen, avdp.path.ptr));
+		goto on_error;
+	    }
+
+	    PJ_LOG(4,(THIS_FILE, "AVI player %.*s created, dev_id=%d",
+		      (int)avdp.title.slen, avdp.title.ptr, avid));
+
+	    app_config.avi[i].dev_id = avid;
+	    if (app_config.avi_def_idx == PJSUA_INVALID_ID)
+		app_config.avi_def_idx = i;
+
+	    strm_cnt = pjmedia_avi_streams_get_num_streams(avdp.avi_streams);
+	    for (strm_idx=0; strm_idx<strm_cnt; ++strm_idx) {
+		pjmedia_port *aud;
+		pjmedia_format *fmt;
+		pjsua_conf_port_id slot;
+		char fmt_name[5];
+
+		aud = pjmedia_avi_streams_get_stream(avdp.avi_streams,
+		                                     strm_idx);
+		fmt = &aud->info.fmt;
+
+		pjmedia_fourcc_name(fmt->id, fmt_name);
+
+		if (fmt->id == PJMEDIA_FORMAT_PCM) {
+		    status = pjsua_conf_add_port(app_config.pool, aud,
+		                                 &slot);
+		    if (status == PJ_SUCCESS) {
+			PJ_LOG(4,(THIS_FILE,
+				  "AVI %.*s: audio added to slot %d",
+				  (int)avdp.title.slen, avdp.title.ptr,
+				  slot));
+			app_config.avi[i].slot = slot;
+		    }
+		} else {
+		    PJ_LOG(4,(THIS_FILE,
+			      "AVI %.*s: audio ignored, format=%s",
+			      (int)avdp.title.slen, avdp.title.ptr,
+			      fmt_name));
+		}
+	    }
+	}
+#else
+	PJ_LOG(2,(THIS_FILE,
+		  "Warning: --play-avi is ignored because AVI is disabled"));
+#endif	/* PJMEDIA_VIDEO_DEV_HAS_AVI */
+    }
+
+    /* Add UDP transport unless it's disabled. */
+    if (!app_config.no_udp) {
+	pjsua_acc_id aid;
+	pjsip_transport_type_e type = PJSIP_TRANSPORT_UDP;
+
+	status = pjsua_transport_create(type,
+					&app_config.udp_cfg,
+					&transport_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_TRUE, &aid);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(aid, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    pjsua_acc_modify(aid, &acc_cfg);
+	}
+
+	//pjsua_acc_set_transport(aid, transport_id);
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+
+	if (app_config.udp_cfg.port == 0) {
+	    pjsua_transport_info ti;
+	    pj_sockaddr_in *a;
+
+	    pjsua_transport_get_info(transport_id, &ti);
+	    a = (pj_sockaddr_in*)&ti.local_addr;
+
+	    tcp_cfg.port = pj_ntohs(a->sin_port);
+	}
+    }
+
+    /* Add UDP IPv6 transport unless it's disabled. */
+    if (!app_config.no_udp && app_config.ipv6) {
+	pjsua_acc_id aid;
+	pjsip_transport_type_e type = PJSIP_TRANSPORT_UDP6;
+	pjsua_transport_config udp_cfg;
+
+	udp_cfg = app_config.udp_cfg;
+	if (udp_cfg.port == 0)
+	    udp_cfg.port = 5060;
+	else
+	    udp_cfg.port += 10;
+	status = pjsua_transport_create(type,
+					&udp_cfg,
+					&transport_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_TRUE, &aid);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(aid, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    acc_cfg.ipv6_media_use = PJSUA_IPV6_ENABLED;
+	    pjsua_acc_modify(aid, &acc_cfg);
+	}
+
+	//pjsua_acc_set_transport(aid, transport_id);
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+
+	if (app_config.udp_cfg.port == 0) {
+	    pjsua_transport_info ti;
+	    pj_sockaddr_in *a;
+
+	    pjsua_transport_get_info(transport_id, &ti);
+	    a = (pj_sockaddr_in*)&ti.local_addr;
+
+	    tcp_cfg.port = pj_ntohs(a->sin_port);
+	}
+    }
+
+    /* Add TCP transport unless it's disabled */
+    if (!app_config.no_tcp) {
+	pjsua_acc_id aid;
+
+	status = pjsua_transport_create(PJSIP_TRANSPORT_TCP,
+					&tcp_cfg, 
+					&transport_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_TRUE, &aid);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(aid, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    pjsua_acc_modify(aid, &acc_cfg);
+	}
+
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+
+    }
+
+    /* Add TCP IPv6 transport unless it's disabled. */
+    if (!app_config.no_tcp && app_config.ipv6) {
+	pjsua_acc_id aid;
+	pjsip_transport_type_e type = PJSIP_TRANSPORT_TCP6;
+
+	tcp_cfg.port += 10;
+
+	status = pjsua_transport_create(type,
+					&tcp_cfg,
+					&transport_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_TRUE, &aid);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(aid, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    acc_cfg.ipv6_media_use = PJSUA_IPV6_ENABLED;
+	    pjsua_acc_modify(aid, &acc_cfg);
+	}
+
+	//pjsua_acc_set_transport(aid, transport_id);
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+    }
+
+
+#if defined(PJSIP_HAS_TLS_TRANSPORT) && PJSIP_HAS_TLS_TRANSPORT!=0
+    /* Add TLS transport when application wants one */
+    if (app_config.use_tls) {
+
+	pjsua_acc_id acc_id;
+
+	/* Copy the QoS settings */
+	tcp_cfg.tls_setting.qos_type = tcp_cfg.qos_type;
+	pj_memcpy(&tcp_cfg.tls_setting.qos_params, &tcp_cfg.qos_params, 
+		  sizeof(tcp_cfg.qos_params));
+
+	/* Set TLS port as TCP port+1 */
+	tcp_cfg.port++;
+	status = pjsua_transport_create(PJSIP_TRANSPORT_TLS,
+					&tcp_cfg, 
+					&transport_id);
+	tcp_cfg.port--;
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+	
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_FALSE, &acc_id);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(acc_id, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    pjsua_acc_modify(acc_id, &acc_cfg);
+	}
+
+	pjsua_acc_set_online_status(acc_id, PJ_TRUE);
+    }
+
+    /* Add TLS IPv6 transport unless it's disabled. */
+    if (app_config.use_tls && app_config.ipv6) {
+	pjsua_acc_id aid;
+	pjsip_transport_type_e type = PJSIP_TRANSPORT_TLS6;
+
+	tcp_cfg.port += 10;
+
+	status = pjsua_transport_create(type,
+					&tcp_cfg,
+					&transport_id);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+
+	/* Add local account */
+	pjsua_acc_add_local(transport_id, PJ_TRUE, &aid);
+
+	/* Adjust local account config based on pjsua app config */
+	{
+	    pjsua_acc_config acc_cfg;
+	    pjsua_acc_get_config(aid, tmp_pool, &acc_cfg);
+
+	    app_config_init_video(&acc_cfg);
+	    acc_cfg.rtp_cfg = app_config.rtp_cfg;
+	    acc_cfg.ipv6_media_use = PJSUA_IPV6_ENABLED;
+	    pjsua_acc_modify(aid, &acc_cfg);
+	}
+
+	//pjsua_acc_set_transport(aid, transport_id);
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+    }
+
+#endif
+
+    if (transport_id == -1) {
+	PJ_LOG(1,(THIS_FILE, "Error: no transport is configured"));
+	status = -1;
+	goto on_error;
+    }
+
+
+    /* Add accounts */
+    for (i=0; i<app_config.acc_cnt; ++i) {
+	app_config.acc_cfg[i].rtp_cfg = app_config.rtp_cfg;
+	app_config.acc_cfg[i].reg_retry_interval = 300;
+	app_config.acc_cfg[i].reg_first_retry_interval = 60;
+
+	app_config_init_video(&app_config.acc_cfg[i]);
+
+	status = pjsua_acc_add(&app_config.acc_cfg[i], PJ_TRUE, NULL);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+	pjsua_acc_set_online_status(current_acc, PJ_TRUE);
+    }
+
+    /* Add buddies */
+    for (i=0; i<app_config.buddy_cnt; ++i) {
+	status = pjsua_buddy_add(&app_config.buddy_cfg[i], NULL);
+	if (status != PJ_SUCCESS) {
+	    PJ_PERROR(1,(THIS_FILE, status, "Error adding buddy"));
+	    goto on_error;
+	}
+    }
+
+    /* Optionally disable some codec */
+    for (i=0; i<app_config.codec_dis_cnt; ++i) {
+	pjsua_codec_set_priority(&app_config.codec_dis[i],
+				 PJMEDIA_CODEC_PRIO_DISABLED);
+#if PJSUA_HAS_VIDEO
+	pjsua_vid_codec_set_priority(&app_config.codec_dis[i],
+				     PJMEDIA_CODEC_PRIO_DISABLED);
+#endif
+    }
+
+    /* Optionally set codec orders */
+    for (i=0; i<app_config.codec_cnt; ++i) {
+	pjsua_codec_set_priority(&app_config.codec_arg[i],
+				 (pj_uint8_t)(PJMEDIA_CODEC_PRIO_NORMAL+i+9));
+#if PJSUA_HAS_VIDEO
+	pjsua_vid_codec_set_priority(&app_config.codec_arg[i],
+				   (pj_uint8_t)(PJMEDIA_CODEC_PRIO_NORMAL+i+9));
+#endif
+    }
+
+    /* Use null sound device? */
+#ifndef STEREO_DEMO
+    if (app_config.null_audio) {
+	status = pjsua_set_null_snd_dev();
+	if (status != PJ_SUCCESS)
+	    return status;
+    }
+#endif
+
+    if (app_config.capture_dev  != PJSUA_INVALID_ID ||
+        app_config.playback_dev != PJSUA_INVALID_ID) 
+    {
+	status = pjsua_set_snd_dev(app_config.capture_dev, 
+				   app_config.playback_dev);
+	if (status != PJ_SUCCESS)
+	    goto on_error;
+    }
+
+    /* Init call setting */
+    pjsua_call_setting_default(&call_opt);
+    call_opt.aud_cnt = app_config.aud_cnt;
+    call_opt.vid_cnt = app_config.vid.vid_cnt;
+
+    pj_pool_release(tmp_pool);
+    return PJ_SUCCESS;
+
+on_error:
+    pj_pool_release(tmp_pool);
+    app_destroy();
+    return status;
+}
+
+pj_status_t pjsua_app_init(const pjsua_app_cfg_t *cfg)
+{
+    pj_status_t status;
+    pj_memcpy(&app_cfg, cfg, sizeof(app_cfg));
+
+    status = app_init();
+    if (status != PJ_SUCCESS)
+	return status;
+
+    /* Init CLI if configured */    
+    if (app_config.use_cli) {
+	status = cli_init();
+    } 
+    return status;
+}
+
+pj_status_t pjsua_app_run(pj_bool_t wait_telnet_cli)
+{
+    pj_thread_t *stdout_refresh_thread = NULL;
+    pj_status_t status;
+
+    /* Start console refresh thread */
+    if (stdout_refresh > 0) {
+	pj_thread_create(app_config.pool, "stdout", &stdout_refresh_proc,
+			 NULL, 0, 0, &stdout_refresh_thread);
+    }
+
+    status = pjsua_start();
+    if (status != PJ_SUCCESS)
+	goto on_return;
+
+    if (app_config.use_cli && (app_config.cli_cfg.cli_fe & CLI_FE_TELNET)) {
+	char info[128];
+	cli_get_info(info, sizeof(info));
+	if (app_cfg.on_started) {
+	    (*app_cfg.on_started)(status, info);
+	}
+    } else {
+	if (app_cfg.on_started) {
+	    (*app_cfg.on_started)(status, "Ready");
+	}    
+    }
+
+    /* If user specifies URI to call, then call the URI */
+    if (uri_arg.slen) {
+	pjsua_call_setting_default(&call_opt);
+	call_opt.aud_cnt = app_config.aud_cnt;
+	call_opt.vid_cnt = app_config.vid.vid_cnt;
+
+	pjsua_call_make_call(current_acc, &uri_arg, &call_opt, NULL, 
+			     NULL, NULL);
+    }   
+
+    app_running = PJ_TRUE;
+
+    if (app_config.use_cli)
+	cli_main(wait_telnet_cli);	
+    else
+	legacy_main();
+
+    status = PJ_SUCCESS;
+
+on_return:
+    if (stdout_refresh_thread) {
+	stdout_refresh_quit = PJ_TRUE;
+	pj_thread_join(stdout_refresh_thread);
+	pj_thread_destroy(stdout_refresh_thread);
+	stdout_refresh_quit = PJ_FALSE;
+    }
+    return status;
+}
+
+static pj_status_t app_destroy()
+{
+    pj_status_t status = PJ_SUCCESS;
+    unsigned i;
+    pj_bool_t use_cli = PJ_FALSE;
+    int cli_fe = 0;
+    pj_uint16_t cli_telnet_port = 0;
+
+#ifdef STEREO_DEMO
+    if (app_config.snd) {
+	pjmedia_snd_port_destroy(app_config.snd);
+	app_config.snd = NULL;
+    }
+    if (app_config.sc_ch1) {
+	pjsua_conf_remove_port(app_config.sc_ch1_slot);
+	app_config.sc_ch1_slot = PJSUA_INVALID_ID;
+	pjmedia_port_destroy(app_config.sc_ch1);
+	app_config.sc_ch1 = NULL;
+    }
+    if (app_config.sc) {
+	pjmedia_port_destroy(app_config.sc);
+	app_config.sc = NULL;
+    }
+#endif
+
+    /* Close avi devs and ports */
+    for (i=0; i<app_config.avi_cnt; ++i) {
+	if (app_config.avi[i].slot != PJSUA_INVALID_ID)
+	    pjsua_conf_remove_port(app_config.avi[i].slot);
+#if PJMEDIA_HAS_VIDEO && PJMEDIA_VIDEO_DEV_HAS_AVI
+	if (app_config.avi[i].dev_id != PJMEDIA_VID_INVALID_DEV)
+	    pjmedia_avi_dev_free(app_config.avi[i].dev_id);
+#endif
+    }
+
+    /* Close ringback port */
+    if (app_config.ringback_port && 
+	app_config.ringback_slot != PJSUA_INVALID_ID) 
+    {
+	pjsua_conf_remove_port(app_config.ringback_slot);
+	app_config.ringback_slot = PJSUA_INVALID_ID;
+	pjmedia_port_destroy(app_config.ringback_port);
+	app_config.ringback_port = NULL;
+    }
+
+    /* Close ring port */
+    if (app_config.ring_port && app_config.ring_slot != PJSUA_INVALID_ID) {
+	pjsua_conf_remove_port(app_config.ring_slot);
+	app_config.ring_slot = PJSUA_INVALID_ID;
+	pjmedia_port_destroy(app_config.ring_port);
+	app_config.ring_port = NULL;
+    }
+
+    /* Close tone generators */
+    for (i=0; i<app_config.tone_count; ++i) {
+	pjsua_conf_remove_port(app_config.tone_slots[i]);
+    }
+
+    if (app_config.pool) {
+	pj_pool_release(app_config.pool);
+	app_config.pool = NULL;
+    }
+
+    status = pjsua_destroy();
+
+    if (app_config.use_cli) {
+	use_cli = app_config.use_cli;
+	cli_fe = app_config.cli_cfg.cli_fe;
+	cli_telnet_port = app_config.cli_cfg.telnet_cfg.port;	
+    }
+
+    /* Reset config */
+    pj_bzero(&app_config, sizeof(app_config));
+
+    if (use_cli) {    
+	app_config.use_cli = use_cli;
+	app_config.cli_cfg.cli_fe = cli_fe;
+	app_config.cli_cfg.telnet_cfg.port = cli_telnet_port;
+    }
+
+    return status;
+}
+
+pj_status_t pjsua_app_destroy()
+{
+    pj_status_t status;
+
+    status = app_destroy();
+
+    if (app_config.use_cli) {	
+	cli_destroy();
+    }
+    
+    return status;
+}
+
+/** ======================= **/
+
+#ifdef STEREO_DEMO
+/*
+ * In this stereo demo, we open the sound device in stereo mode and
+ * arrange the attachment to the PJSUA-LIB conference bridge as such
+ * so that channel0/left channel of the sound device corresponds to
+ * slot 0 in the bridge, and channel1/right channel of the sound
+ * device corresponds to slot 1 in the bridge. Then user can independently
+ * feed different media to/from the speakers/microphones channels, by
+ * connecting them to slot 0 or 1 respectively.
+ *
+ * Here's how the connection looks like:
+ *
+   +-----------+ stereo +-----------------+ 2x mono +-----------+
+   | AUDIO DEV |<------>| SPLITCOMB   left|<------->|#0  BRIDGE |
+   +-----------+        |            right|<------->|#1         |
+                        +-----------------+         +-----------+
+ */
+static void stereo_demo()
+{
+    pjmedia_port *conf;
+    pj_status_t status;
+
+    /* Disable existing sound device */
+    conf = pjsua_set_no_snd_dev();
+
+    /* Create stereo-mono splitter/combiner */
+    status = pjmedia_splitcomb_create(app_config.pool, 
+				      conf->info.clock_rate /* clock rate */,
+				      2	    /* stereo */,
+				      2 * conf->info.samples_per_frame,
+				      conf->info.bits_per_sample,
+				      0	    /* options */,
+				      &app_config.sc);
+    pj_assert(status == PJ_SUCCESS);
+
+    /* Connect channel0 (left channel?) to conference port slot0 */
+    status = pjmedia_splitcomb_set_channel(app_config.sc, 0 /* ch0 */, 
+					   0 /*options*/,
+					   conf);
+    pj_assert(status == PJ_SUCCESS);
+
+    /* Create reverse channel for channel1 (right channel?)... */
+    status = pjmedia_splitcomb_create_rev_channel(app_config.pool,
+						  app_config.sc,
+						  1  /* ch1 */,
+						  0  /* options */,
+						  &app_config.sc_ch1);
+    pj_assert(status == PJ_SUCCESS);
+
+    /* .. and register it to conference bridge (it would be slot1
+     * if there's no other devices connected to the bridge)
+     */
+    status = pjsua_conf_add_port(app_config.pool, app_config.sc_ch1, 
+				 &app_config.sc_ch1_slot);
+    pj_assert(status == PJ_SUCCESS);
+    
+    /* Create sound device */
+    status = pjmedia_snd_port_create(app_config.pool, -1, -1, 
+				     conf->info.clock_rate,
+				     2	    /* stereo */,
+				     2 * conf->info.samples_per_frame,
+				     conf->info.bits_per_sample,
+				     0, &app_config.snd);
+    pj_assert(status == PJ_SUCCESS);
+
+
+    /* Connect the splitter to the sound device */
+    status = pjmedia_snd_port_connect(app_config.snd, app_config.sc);
+    pj_assert(status == PJ_SUCCESS);
+}
+#endif