pjsip-apps/src/samples/siprtp.c

/* $Id$ */
/* 
 * Copyright (C) 2003-2006 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 
 */





/* Usage */
static const char *USAGE = 
" PURPOSE:								    \n"
"   This program establishes SIP INVITE session and media, and calculate    \n"
"   the media quality (packet lost, jitter, rtt, etc.). Unlike normal	    \n"
"   pjmedia applications, this program bypasses all pjmedia stream	    \n"
"   framework and transmit encoded RTP packets manually using own thread.   \n"
"\n"
" USAGE:\n"
"   siprtp [options]        => to start in server mode\n"
"   siprtp [options] URL    => to start in client mode\n"
"\n"
" Program options:\n"
"   --count=N,        -c    Set number of calls to create (default:1) \n"
"   --duration=SEC,   -d    Set maximum call duration (default:unlimited) \n"
"   --auto-quit,      -q    Quit when calls have been completed (default:no)\n"
"\n"
" Address and ports options:\n"
"   --local-port=PORT,-p    Set local SIP port (default: 5060)\n"
"   --rtp-port=PORT,  -r    Set start of RTP port (default: 4000)\n"
"   --ip-addr=IP,     -i    Set local IP address to use (otherwise it will\n"
"                           try to determine local IP address from hostname)\n"
"\n"
" Logging Options:\n"
"   --log-level=N,    -l    Set log verbosity level (default=5)\n"
"   --app-log-level=N       Set app screen log verbosity (default=3)\n"
"   --log-file=FILE         Write log to file FILE\n"
"   --report-file=FILE      Write report to file FILE\n"
"\n"
/* Don't support this anymore, because codec is properly examined in
   pjmedia_session_info_from_sdp() function.

" Codec Options:\n"
"   --a-pt=PT               Set audio payload type to PT (default=0)\n"
"   --a-name=NAME           Set audio codec name to NAME (default=pcmu)\n"
"   --a-clock=RATE          Set audio codec rate to RATE Hz (default=8000Hz)\n"
"   --a-bitrate=BPS         Set audio codec bitrate to BPS (default=64000bps)\n"
"   --a-ptime=MS            Set audio frame time to MS msec (default=20ms)\n"
*/
;


/* Include all headers. */
#include <pjsip.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjsip_ua.h>
#include <pjsip_simple.h>
#include <pjlib-util.h>
#include <pjlib.h>

#include <stdlib.h>


#if PJ_HAS_HIGH_RES_TIMER==0
#   error "High resolution timer is needed for this sample"
#endif

#define THIS_FILE	"siprtp.c"
#define MAX_CALLS	1024
#define RTP_START_PORT	4000


/* Codec descriptor: */
struct codec
{
    unsigned	pt;
    char*	name;
    unsigned	clock_rate;
    unsigned	bit_rate;
    unsigned	ptime;
    char*	description;
};


/* A bidirectional media stream created when the call is active. */
struct media_stream
{
    /* Static: */
    unsigned		 call_index;	    /* Call owner.		*/
    unsigned		 media_index;	    /* Media index in call.	*/
    pjmedia_transport   *transport;	    /* To send/recv RTP/RTCP	*/

    /* Active? */
    pj_bool_t		 active;	    /* Non-zero if is in call.	*/

    /* Current stream info: */
    pjmedia_stream_info	 si;		    /* Current stream info.	*/

    /* More info: */
    unsigned		 clock_rate;	    /* clock rate		*/
    unsigned		 samples_per_frame; /* samples per frame	*/
    unsigned		 bytes_per_frame;   /* frame size.		*/

    /* RTP session: */
    pjmedia_rtp_session	 out_sess;	    /* outgoing RTP session	*/
    pjmedia_rtp_session	 in_sess;	    /* incoming RTP session	*/

    /* RTCP stats: */
    pjmedia_rtcp_session rtcp;		    /* incoming RTCP session.	*/

    /* Timer to send RTP and RTCP: */
    pj_timer_entry	 rtp_timer;	    /* timer to send RTP pkt.	*/
    pj_timer_entry	 rtcp_timer;	    /* timer to send RTCP pkt.	*/
};


/* This is a call structure that is created when the application starts
 * and only destroyed when the application quits.
 */
struct call
{
    unsigned		 index;
    pjsip_inv_session	*inv;
    unsigned		 media_count;
    struct media_stream	 media[1];
    pj_time_val		 start_time;
    pj_time_val		 response_time;
    pj_time_val		 connect_time;

    pj_timer_entry	 d_timer;	    /**< Disconnect timer.	*/
};


/* Application's global variables */
static struct app
{
    unsigned		 max_calls;
    unsigned		 uac_calls;
    unsigned		 duration;
    pj_bool_t		 auto_quit;
    unsigned		 thread_count;
    int			 sip_port;
    int			 rtp_start_port;
    pj_str_t		 local_addr;
    pj_str_t		 local_uri;
    pj_str_t		 local_contact;
    
    int			 app_log_level;
    int			 log_level;
    char		*log_filename;
    char		*report_filename;

    struct codec	 audio_codec;

    pj_str_t		 uri_to_call;

    pj_caching_pool	 cp;
    pj_pool_t		*pool;

    pjsip_endpoint	*sip_endpt;
    pj_bool_t		 thread_quit;
    pj_thread_t		*sip_thread[1];

    pjmedia_endpt	*med_endpt;
    struct call		 call[MAX_CALLS];
} app;



/*
 * Prototypes:
 */

/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
				  pj_status_t status);

/* Callback to be called when invite session's state has changed: */
static void call_on_state_changed( pjsip_inv_session *inv, 
				   pjsip_event *e);

/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);

/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata );

/* Worker thread prototype */
static int sip_worker_thread(void *arg);

/* Create SDP for call */
static pj_status_t create_sdp( pj_pool_t *pool,
			       struct call *call,
			       pjmedia_sdp_session **p_sdp);

/* Hangup call */
static void hangup_call(unsigned index);

/* Destroy the call's media */
static void destroy_call_media(unsigned call_index);

/* Destroy media. */
static void destroy_media();

/* This callback is called by media transport on receipt of RTP packet. */
static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size);

/* This callback is called by media transport on receipt of RTCP packet. */
static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size);

/* This callback is called when it's time to send RTP packet */
static void on_tx_rtp( pj_timer_heap_t *timer_heap,
		       struct pj_timer_entry *entry);

/* This callback is called when it's time to send RTCP packet. */
static void on_tx_rtcp(pj_timer_heap_t *timer_heap,
		       struct pj_timer_entry *entry);


/* Display error */
static void app_perror(const char *sender, const char *title, 
		       pj_status_t status);

/* Print call */
static void print_call(int call_index);


/* This is a PJSIP module to be registered by application to handle
 * incoming requests outside any dialogs/transactions. The main purpose
 * here is to handle incoming INVITE request message, where we will
 * create a dialog and INVITE session for it.
 */
static pjsip_module mod_siprtp =
{
    NULL, NULL,			    /* prev, next.		*/
    { "mod-siprtpapp", 13 },	    /* Name.			*/
    -1,				    /* Id			*/
    PJSIP_MOD_PRIORITY_APPLICATION, /* Priority			*/
    NULL,			    /* load()			*/
    NULL,			    /* start()			*/
    NULL,			    /* stop()			*/
    NULL,			    /* unload()			*/
    &on_rx_request,		    /* on_rx_request()		*/
    NULL,			    /* on_rx_response()		*/
    NULL,			    /* on_tx_request.		*/
    NULL,			    /* on_tx_response()		*/
    NULL,			    /* on_tsx_state()		*/
};


/* Codec constants */
struct codec audio_codecs[] = 
{
    { 0,  "PCMU", 8000, 64000, 20, "G.711 ULaw" },
    { 3,  "GSM",  8000, 13200, 20, "GSM" },
    { 4,  "G723", 8000, 6400,  30, "G.723.1" },
    { 8,  "PCMA", 8000, 64000, 20, "G.711 ALaw" },
    { 18, "G729", 8000, 8000,  20, "G.729" },
};


/*
 * Init SIP stack
 */
static pj_status_t init_sip()
{
    unsigned i;
    pj_status_t status;

    /* init PJLIB-UTIL: */
    status = pjlib_util_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /* Must create a pool factory before we can allocate any memory. */
    pj_caching_pool_init(&app.cp, &pj_pool_factory_default_policy, 0);

    /* Create application pool for misc. */
    app.pool = pj_pool_create(&app.cp.factory, "app", 1000, 1000, NULL);

    /* Create the endpoint: */
    status = pjsip_endpt_create(&app.cp.factory, pj_gethostname()->ptr, 
				&app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Add UDP transport. */
    {
	pj_sockaddr_in addr;
	pjsip_host_port addrname;

	pj_memset(&addr, 0, sizeof(addr));
	addr.sin_family = PJ_AF_INET;
	addr.sin_addr.s_addr = 0;
	addr.sin_port = pj_htons((pj_uint16_t)app.sip_port);

	if (app.local_addr.slen) {
	    addrname.host = app.local_addr;
	    addrname.port = app.sip_port;
	}

	status = pjsip_udp_transport_start( app.sip_endpt, &addr, 
					    (app.local_addr.slen ? &addrname:NULL),
					    1, NULL);
	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to start UDP transport", status);
	    return status;
	}
    }

    /* 
     * Init transaction layer.
     * This will create/initialize transaction hash tables etc.
     */
    status = pjsip_tsx_layer_init_module(app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /*  Initialize UA layer. */
    status = pjsip_ua_init_module( app.sip_endpt, NULL );
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /*  Init invite session module. */
    {
	pjsip_inv_callback inv_cb;

	/* Init the callback for INVITE session: */
	pj_memset(&inv_cb, 0, sizeof(inv_cb));
	inv_cb.on_state_changed = &call_on_state_changed;
	inv_cb.on_new_session = &call_on_forked;
	inv_cb.on_media_update = &call_on_media_update;

	/* Initialize invite session module:  */
	status = pjsip_inv_usage_init(app.sip_endpt, &inv_cb);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }

    /* Register our module to receive incoming requests. */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_siprtp);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /* Init calls */
    for (i=0; i<app.max_calls; ++i)
	app.call[i].index = i;

    /* Done */
    return PJ_SUCCESS;
}


/*
 * Destroy SIP
 */
static void destroy_sip()
{
    unsigned i;

    app.thread_quit = 1;
    for (i=0; i<app.thread_count; ++i) {
	if (app.sip_thread[i]) {
	    pj_thread_join(app.sip_thread[i]);
	    pj_thread_destroy(app.sip_thread[i]);
	    app.sip_thread[i] = NULL;
	}
    }

    if (app.sip_endpt) {
	pjsip_endpt_destroy(app.sip_endpt);
	app.sip_endpt = NULL;
    }
}


/*
 * Init media stack.
 */
static pj_status_t init_media()
{
    unsigned	i, count;
    pj_uint16_t	rtp_port;
    pj_status_t	status;


    /* Initialize media endpoint so that at least error subsystem is properly
     * initialized.
     */
    status = pjmedia_endpt_create(&app.cp.factory, 
				  pjsip_endpt_get_ioqueue(app.sip_endpt), 0, 
				  &app.med_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Must register codecs to be supported */
    pjmedia_codec_g711_init(app.med_endpt);

    /* RTP port counter */
    rtp_port = (pj_uint16_t)(app.rtp_start_port & 0xFFFE);

    /* Init media transport for all calls. */
    for (i=0, count=0; i<app.max_calls; ++i, ++count) {

	unsigned j;

	/* Create transport for each media in the call */
	for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
	    /* Repeat binding media socket to next port when fails to bind
	     * to current port number.
	     */
	    struct media_stream *m = &app.call[i].media[j];
	    int retry;

	    m->call_index = i;
	    m->media_index = j;

	    m->rtp_timer.user_data = m;
	    m->rtp_timer.cb = &on_tx_rtp;

	    m->rtcp_timer.user_data = m;
	    m->rtcp_timer.cb = &on_tx_rtcp;


	    status = -1;
	    for (retry=0; retry<100; ++retry,rtp_port+=2)  {
		struct media_stream *m = &app.call[i].media[j];
		
		status = pjmedia_transport_udp_create2(app.med_endpt, 
						       "siprtp",
						       &app.local_addr,
						       rtp_port, 0, 
						       &m->transport);
		if (status == PJ_SUCCESS) {
		    rtp_port += 2;
		    break;
		}
	    }
	}

	if (status != PJ_SUCCESS)
	    goto on_error;
    }

    /* Done */
    return PJ_SUCCESS;

on_error:
    destroy_media();
    return status;
}


/*
 * Destroy media.
 */
static void destroy_media()
{
    unsigned i;

    for (i=0; i<app.max_calls; ++i) {
	unsigned j;
	for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
	    struct media_stream *m = &app.call[i].media[j];

	    if (m->transport) {
		pjmedia_transport_close(m->transport);
		m->transport = NULL;
	    }
	}
    }

    if (app.med_endpt) {
	pjmedia_endpt_destroy(app.med_endpt);
	app.med_endpt = NULL;
    }
}


/*
 * Make outgoing call.
 */
static pj_status_t make_call(const pj_str_t *dst_uri)
{
    unsigned i;
    struct call *call;
    pjsip_dialog *dlg;
    pjmedia_sdp_session *sdp;
    pjsip_tx_data *tdata;
    pj_status_t status;


    /* Find unused call slot */
    for (i=0; i<app.max_calls; ++i) {
	if (app.call[i].inv == NULL)
	    break;
    }

    if (i == app.max_calls)
	return PJ_ETOOMANY;

    call = &app.call[i];

    /* Create UAC dialog */
    status = pjsip_dlg_create_uac( pjsip_ua_instance(), 
				   &app.local_uri,	/* local URI	    */
				   &app.local_contact,	/* local Contact    */
				   dst_uri,		/* remote URI	    */
				   dst_uri,		/* remote target    */
				   &dlg);		/* dialog	    */
    if (status != PJ_SUCCESS) {
	++app.uac_calls;
	return status;
    }

    /* Create SDP */
    create_sdp( dlg->pool, call, &sdp);

    /* Create the INVITE session. */
    status = pjsip_inv_create_uac( dlg, sdp, 0, &call->inv);
    if (status != PJ_SUCCESS) {
	pjsip_dlg_terminate(dlg);
	++app.uac_calls;
	return status;
    }


    /* Attach call data to invite session */
    call->inv->mod_data[mod_siprtp.id] = call;

    /* Mark start of call */
    pj_gettimeofday(&call->start_time);


    /* Create initial INVITE request.
     * This INVITE request will contain a perfectly good request and 
     * an SDP body as well.
     */
    status = pjsip_inv_invite(call->inv, &tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Send initial INVITE request. 
     * From now on, the invite session's state will be reported to us
     * via the invite session callbacks.
     */
    status = pjsip_inv_send_msg(call->inv, tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    return PJ_SUCCESS;
}


/*
 * Receive incoming call
 */
static void process_incoming_call(pjsip_rx_data *rdata)
{
    unsigned i, options;
    struct call *call;
    pjsip_dialog *dlg;
    pjmedia_sdp_session *sdp;
    pjsip_tx_data *tdata;
    pj_status_t status;

    /* Find free call slot */
    for (i=0; i<app.max_calls; ++i) {
	if (app.call[i].inv == NULL)
	    break;
    }

    if (i == app.max_calls) {
	const pj_str_t reason = pj_str("Too many calls");
	pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
				       500, &reason,
				       NULL, NULL);
	return;
    }

    /* Verify that we can handle the request. */
    options = 0;
    status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
				      app.sip_endpt, &tdata);
    if (status != PJ_SUCCESS) {

	/*
	 * No we can't handle the incoming INVITE request.
	 */

	if (tdata) {
	    pjsip_response_addr res_addr;

	    pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
	    pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata, 
				      NULL, NULL);

	} else {

	    /* Respond with 500 (Internal Server Error) */
	    pjsip_endpt_respond_stateless(app.sip_endpt, rdata, 500, NULL,
					  NULL, NULL);
	}

	return;
    } 

    call = &app.call[i];

    /* Create UAS dialog */
    status = pjsip_dlg_create_uas( pjsip_ua_instance(), rdata,
				   &app.local_contact, &dlg);
    if (status != PJ_SUCCESS) {
	const pj_str_t reason = pj_str("Unable to create dialog");
	pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
				       500, &reason,
				       NULL, NULL);
	return;
    }

    /* Create SDP */
    create_sdp( dlg->pool, call, &sdp);

    /* Create UAS invite session */
    status = pjsip_inv_create_uas( dlg, rdata, sdp, 0, &call->inv);
    if (status != PJ_SUCCESS) {
	pjsip_dlg_create_response(dlg, rdata, 500, NULL, &tdata);
	pjsip_dlg_send_response(dlg, pjsip_rdata_get_tsx(rdata), tdata);
	return;
    }
    

    /* Attach call data to invite session */
    call->inv->mod_data[mod_siprtp.id] = call;

    /* Mark start of call */
    pj_gettimeofday(&call->start_time);



    /* Create 200 response .*/
    status = pjsip_inv_initial_answer(call->inv, rdata, 200, 
				      NULL, NULL, &tdata);
    if (status != PJ_SUCCESS) {
	status = pjsip_inv_initial_answer(call->inv, rdata, 
					  PJSIP_SC_NOT_ACCEPTABLE,
					  NULL, NULL, &tdata);
	if (status == PJ_SUCCESS)
	    pjsip_inv_send_msg(call->inv, tdata); 
	else
	    pjsip_inv_terminate(call->inv, 500, PJ_FALSE);
	return;
    }


    /* Send the 200 response. */  
    status = pjsip_inv_send_msg(call->inv, tdata); 
    PJ_ASSERT_ON_FAIL(status == PJ_SUCCESS, return);


    /* Done */
}


/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
{
    PJ_UNUSED_ARG(inv);
    PJ_UNUSED_ARG(e);

    PJ_TODO( HANDLE_FORKING );
}


/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata )
{
    /* Ignore strandled ACKs (must not send respone */
    if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD)
	return PJ_FALSE;

    /* Respond (statelessly) any non-INVITE requests with 500  */
    if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
	pj_str_t reason = pj_str("Unsupported Operation");
	pjsip_endpt_respond_stateless( app.sip_endpt, rdata, 
				       500, &reason,
				       NULL, NULL);
	return PJ_TRUE;
    }

    /* Handle incoming INVITE */
    process_incoming_call(rdata);

    /* Done */
    return PJ_TRUE;
}


/* Callback timer to disconnect call (limiting call duration) */
static void timer_disconnect_call( pj_timer_heap_t *timer_heap,
				   struct pj_timer_entry *entry)
{
    struct call *call = entry->user_data;

    PJ_UNUSED_ARG(timer_heap);

    entry->id = 0;
    hangup_call(call->index);
}


/* Callback to be called when invite session's state has changed: */
static void call_on_state_changed( pjsip_inv_session *inv, 
				   pjsip_event *e)
{
    struct call *call = inv->mod_data[mod_siprtp.id];

    PJ_UNUSED_ARG(e);

    if (!call)
	return;

    if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
	
	pj_time_val null_time = {0, 0};

	if (call->d_timer.id != 0) {
	    pjsip_endpt_cancel_timer(app.sip_endpt, &call->d_timer);
	    call->d_timer.id = 0;
	}

	PJ_LOG(3,(THIS_FILE, "Call #%d disconnected. Reason=%s",
		  call->index,
		  pjsip_get_status_text(inv->cause)->ptr));
	PJ_LOG(3,(THIS_FILE, "Call #%d statistics:", call->index));
	print_call(call->index);


	call->inv = NULL;
	inv->mod_data[mod_siprtp.id] = NULL;

	destroy_call_media(call->index);

	call->start_time = null_time;
	call->response_time = null_time;
	call->connect_time = null_time;

	++app.uac_calls;

    } else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {

	pj_time_val t;

	pj_gettimeofday(&call->connect_time);
	if (call->response_time.sec == 0)
	    call->response_time = call->connect_time;

	t = call->connect_time;
	PJ_TIME_VAL_SUB(t, call->start_time);

	PJ_LOG(3,(THIS_FILE, "Call #%d connected in %d ms", call->index,
		  PJ_TIME_VAL_MSEC(t)));

	if (app.duration != 0) {
	    call->d_timer.id = 1;
	    call->d_timer.user_data = call;
	    call->d_timer.cb = &timer_disconnect_call;

	    t.sec = app.duration;
	    t.msec = 0;

	    pjsip_endpt_schedule_timer(app.sip_endpt, &call->d_timer, &t);
	}

    } else if (	inv->state == PJSIP_INV_STATE_EARLY ||
		inv->state == PJSIP_INV_STATE_CONNECTING) {

	if (call->response_time.sec == 0)
	    pj_gettimeofday(&call->response_time);

    }
}


/* Utility */
static void app_perror(const char *sender, const char *title, 
		       pj_status_t status)
{
    char errmsg[PJ_ERR_MSG_SIZE];

    pj_strerror(status, errmsg, sizeof(errmsg));
    PJ_LOG(3,(sender, "%s: %s [status=%d]", title, errmsg, status));
}


#if defined(PJ_WIN32) && PJ_WIN32 != 0
#include <windows.h>
static void boost_priority(void)
{
    SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS);
    SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
}

#else
#  define boost_priority()
#endif


/* Worker thread for SIP */
static int sip_worker_thread(void *arg)
{
    PJ_UNUSED_ARG(arg);

    boost_priority();

    while (!app.thread_quit) {
	pj_time_val timeout = {0, 1};
	pjsip_endpt_handle_events(app.sip_endpt, &timeout);
    }

    return 0;
}


/* Init application options */
static pj_status_t init_options(int argc, char *argv[])
{
    static char ip_addr[32];
    static char local_uri[64];

    enum { OPT_START,
	   OPT_APP_LOG_LEVEL, OPT_LOG_FILE, 
	   OPT_A_PT, OPT_A_NAME, OPT_A_CLOCK, OPT_A_BITRATE, OPT_A_PTIME,
	   OPT_REPORT_FILE };

    struct pj_getopt_option long_options[] = {
	{ "count",	    1, 0, 'c' },
	{ "duration",	    1, 0, 'd' },
	{ "auto-quit",	    0, 0, 'q' },
	{ "local-port",	    1, 0, 'p' },
	{ "rtp-port",	    1, 0, 'r' },
	{ "ip-addr",	    1, 0, 'i' },

	{ "log-level",	    1, 0, 'l' },
	{ "app-log-level",  1, 0, OPT_APP_LOG_LEVEL },
	{ "log-file",	    1, 0, OPT_LOG_FILE },

	{ "report-file",    1, 0, OPT_REPORT_FILE },

	/* Don't support this anymore, see comments in USAGE above.
	{ "a-pt",	    1, 0, OPT_A_PT },
	{ "a-name",	    1, 0, OPT_A_NAME },
	{ "a-clock",	    1, 0, OPT_A_CLOCK },
	{ "a-bitrate",	    1, 0, OPT_A_BITRATE },
	{ "a-ptime",	    1, 0, OPT_A_PTIME },
	*/

	{ NULL, 0, 0, 0 },
    };
    int c;
    int option_index;

    /* Get local IP address for the default IP address */
    {
	const pj_str_t *hostname;
	pj_sockaddr_in tmp_addr;
	char *addr;

	hostname = pj_gethostname();
	pj_sockaddr_in_init(&tmp_addr, hostname, 0);
	addr = pj_inet_ntoa(tmp_addr.sin_addr);
	pj_ansi_strcpy(ip_addr, addr);
    }

    /* Init defaults */
    app.max_calls = 1;
    app.thread_count = 1;
    app.sip_port = 5060;
    app.rtp_start_port = RTP_START_PORT;
    app.local_addr = pj_str(ip_addr);
    app.log_level = 5;
    app.app_log_level = 3;
    app.log_filename = NULL;

    /* Default codecs: */
    app.audio_codec = audio_codecs[0];

    /* Parse options */
    pj_optind = 0;
    while((c=pj_getopt_long(argc,argv, "c:d:p:r:i:l:q", 
			    long_options, &option_index))!=-1) 
    {
	switch (c) {
	case 'c':
	    app.max_calls = atoi(pj_optarg);
	    if (app.max_calls < 0 || app.max_calls > MAX_CALLS) {
		PJ_LOG(3,(THIS_FILE, "Invalid max calls value %s", pj_optarg));
		return 1;
	    }
	    break;
	case 'd':
	    app.duration = atoi(pj_optarg);
	    break;
	case 'q':
	    app.auto_quit = 1;
	    break;

	case 'p':
	    app.sip_port = atoi(pj_optarg);
	    break;
	case 'r':
	    app.rtp_start_port = atoi(pj_optarg);
	    break;
	case 'i':
	    app.local_addr = pj_str(pj_optarg);
	    break;

	case 'l':
	    app.log_level = atoi(pj_optarg);
	    break;
	case OPT_APP_LOG_LEVEL:
	    app.app_log_level = atoi(pj_optarg);
	    break;
	case OPT_LOG_FILE:
	    app.log_filename = pj_optarg;
	    break;

	case OPT_A_PT:
	    app.audio_codec.pt = atoi(pj_optarg);
	    break;
	case OPT_A_NAME:
	    app.audio_codec.name = pj_optarg;
	    break;
	case OPT_A_CLOCK:
	    app.audio_codec.clock_rate = atoi(pj_optarg);
	    break;
	case OPT_A_BITRATE:
	    app.audio_codec.bit_rate = atoi(pj_optarg);
	    break;
	case OPT_A_PTIME:
	    app.audio_codec.ptime = atoi(pj_optarg);
	    break;
	case OPT_REPORT_FILE:
	    app.report_filename = pj_optarg;
	    break;

	default:
	    puts(USAGE);
	    return 1;
	}
    }

    /* Check if URL is specified */
    if (pj_optind < argc)
	app.uri_to_call = pj_str(argv[pj_optind]);

    /* Build local URI and contact */
    pj_ansi_sprintf( local_uri, "sip:%s:%d", app.local_addr.ptr, app.sip_port);
    app.local_uri = pj_str(local_uri);
    app.local_contact = app.local_uri;


    return PJ_SUCCESS;
}


/*****************************************************************************
 * MEDIA STUFFS
 */

/*
 * Create SDP session for a call.
 */
static pj_status_t create_sdp( pj_pool_t *pool,
			       struct call *call,
			       pjmedia_sdp_session **p_sdp)
{
    pj_time_val tv;
    pjmedia_sdp_session *sdp;
    pjmedia_sdp_media *m;
    pjmedia_sdp_attr *attr;
    pjmedia_transport_udp_info tpinfo;
    struct media_stream *audio = &call->media[0];

    PJ_ASSERT_RETURN(pool && p_sdp, PJ_EINVAL);


    /* Get transport info */
    pjmedia_transport_udp_get_info(audio->transport, &tpinfo);

    /* Create and initialize basic SDP session */
    sdp = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_session));

    pj_gettimeofday(&tv);
    sdp->origin.user = pj_str("pjsip-siprtp");
    sdp->origin.version = sdp->origin.id = tv.sec + 2208988800UL;
    sdp->origin.net_type = pj_str("IN");
    sdp->origin.addr_type = pj_str("IP4");
    sdp->origin.addr = *pj_gethostname();
    sdp->name = pj_str("pjsip");

    /* Since we only support one media stream at present, put the
     * SDP connection line in the session level.
     */
    sdp->conn = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_conn));
    sdp->conn->net_type = pj_str("IN");
    sdp->conn->addr_type = pj_str("IP4");
    sdp->conn->addr = app.local_addr;


    /* SDP time and attributes. */
    sdp->time.start = sdp->time.stop = 0;
    sdp->attr_count = 0;

    /* Create media stream 0: */

    sdp->media_count = 1;
    m = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_media));
    sdp->media[0] = m;

    /* Standard media info: */
    m->desc.media = pj_str("audio");
    m->desc.port = pj_ntohs(tpinfo.skinfo.rtp_addr_name.sin_port);
    m->desc.port_count = 1;
    m->desc.transport = pj_str("RTP/AVP");

    /* Add format and rtpmap for each codec. */
    m->desc.fmt_count = 1;
    m->attr_count = 0;

    {
	pjmedia_sdp_rtpmap rtpmap;
	pjmedia_sdp_attr *attr;
	char ptstr[10];

	sprintf(ptstr, "%d", app.audio_codec.pt);
	pj_strdup2(pool, &m->desc.fmt[0], ptstr);
	rtpmap.pt = m->desc.fmt[0];
	rtpmap.clock_rate = app.audio_codec.clock_rate;
	rtpmap.enc_name = pj_str(app.audio_codec.name);
	rtpmap.param.slen = 0;

	pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
	m->attr[m->attr_count++] = attr;
    }

    /* Add sendrecv attribute. */
    attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
    attr->name = pj_str("sendrecv");
    m->attr[m->attr_count++] = attr;

#if 1
    /*
     * Add support telephony event
     */
    m->desc.fmt[m->desc.fmt_count++] = pj_str("121");
    /* Add rtpmap. */
    attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
    attr->name = pj_str("rtpmap");
    attr->value = pj_str(":121 telephone-event/8000");
    m->attr[m->attr_count++] = attr;
    /* Add fmtp */
    attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
    attr->name = pj_str("fmtp");
    attr->value = pj_str(":121 0-15");
    m->attr[m->attr_count++] = attr;
#endif

    /* Done */
    *p_sdp = sdp;

    return PJ_SUCCESS;
}


/*
 * This callback is called by media transport on receipt of RTP packet.
 */
static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size)
{
    struct media_stream *strm;
    pj_status_t status;
    const pjmedia_rtp_hdr *hdr;
    const void *payload;
    unsigned payload_len;

    strm = user_data;

    /* Discard packet if media is inactive */
    if (!strm->active)
	return;

    /* Check for errors */
    if (size < 0) {
	app_perror(THIS_FILE, "RTP recv() error", -size);
	return;
    }

    /* Decode RTP packet. */
    status = pjmedia_rtp_decode_rtp(&strm->in_sess, 
				    pkt, size, 
				    &hdr, &payload, &payload_len);
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "RTP decode error", status);
	return;
    }

    //PJ_LOG(4,(THIS_FILE, "Rx seq=%d", pj_ntohs(hdr->seq)));

    /* Update the RTCP session. */
    pjmedia_rtcp_rx_rtp(&strm->rtcp, pj_ntohs(hdr->seq),
			pj_ntohl(hdr->ts), payload_len);

    /* Update RTP session */
    pjmedia_rtp_session_update(&strm->in_sess, hdr, NULL);

}

/* This callback is called when it's time to send RTP packet */
static void on_tx_rtp( pj_timer_heap_t *timer_heap,
		       struct pj_timer_entry *entry)
{
    pj_status_t status;
    const pjmedia_rtp_hdr *hdr;
    pj_ssize_t size;
    int hdrlen;
    pj_time_val interval;
    char packet[512];
    struct media_stream *strm = entry->user_data;

    PJ_UNUSED_ARG(timer_heap);

    if (!strm->active)
	return;

    /* Format RTP header */
    status = pjmedia_rtp_encode_rtp( &strm->out_sess, strm->si.tx_pt,
				     0, /* marker bit */
				     strm->bytes_per_frame, 
				     strm->samples_per_frame,
				     (const void**)&hdr, &hdrlen);
    if (status == PJ_SUCCESS) {

	//PJ_LOG(4,(THIS_FILE, "\t\tTx seq=%d", pj_ntohs(hdr->seq)));

	/* Copy RTP header to packet */
	pj_memcpy(packet, hdr, hdrlen);

	/* Zero the payload */
	pj_memset(packet+hdrlen, 0, strm->bytes_per_frame);

	/* Send RTP packet */
	size = hdrlen + strm->bytes_per_frame;
	status = pjmedia_transport_send_rtp(strm->transport, 
					    packet, size);
	if (status != PJ_SUCCESS)
	    app_perror(THIS_FILE, "Error sending RTP packet", status);

    } else {
	pj_assert(!"RTP encode() error");
    }

    /* Update RTCP SR */
    pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame);

    /* Schedule next send */
    interval.sec = 0;
    interval.msec = strm->samples_per_frame * 1000 / strm->clock_rate;
    pj_time_val_normalize(&interval);

    pjsip_endpt_schedule_timer(app.sip_endpt, &strm->rtp_timer, &interval);
}


/*
 * This callback is called by media transport on receipt of RTCP packet.
 */
static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size)
{
    struct media_stream *strm;

    strm = user_data;

    /* Discard packet if media is inactive */
    if (!strm->active)
	return;

    /* Check for errors */
    if (size < 0) {
	app_perror(THIS_FILE, "Error receiving RTCP packet", -size);
	return;
    }

    /* Update RTCP session */
    pjmedia_rtcp_rx_rtcp(&strm->rtcp, pkt, size);
}


/* This callback is called when it's time to send RTCP packet. */
static void on_tx_rtcp(pj_timer_heap_t *timer_heap,
		       struct pj_timer_entry *entry)
{
    pjmedia_rtcp_pkt *rtcp_pkt;
    int rtcp_len;
    pj_ssize_t size;
    pj_status_t status;
    pj_time_val interval;
    struct media_stream *strm = entry->user_data;

    PJ_UNUSED_ARG(timer_heap);

    if (!strm->active)
	return;

    /* Build RTCP packet */
    pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len);

    /* Send packet */
    size = rtcp_len;
    status = pjmedia_transport_send_rtcp(strm->transport,
					 rtcp_pkt, size);
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "Error sending RTCP packet", status);
    }
    
    /* Schedule next send */
    interval.sec = 5;
    interval.msec = (pj_rand() % 500);
    pjsip_endpt_schedule_timer(app.sip_endpt, &strm->rtcp_timer, &interval);

}

/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
				  pj_status_t status)
{
    struct call *call;
    pj_pool_t *pool;
    struct media_stream *audio;
    const pjmedia_sdp_session *local_sdp, *remote_sdp;
    struct codec *codec_desc = NULL;
    pj_time_val interval;
    unsigned i;

    call = inv->mod_data[mod_siprtp.id];
    pool = inv->dlg->pool;
    audio = &call->media[0];

    /* If this is a mid-call media update, then destroy existing media */
    if (audio->active)
	destroy_call_media(call->index);


    /* Do nothing if media negotiation has failed */
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "SDP negotiation failed", status);
	return;
    }

    
    /* Capture stream definition from the SDP */
    pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
    pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);

    status = pjmedia_stream_info_from_sdp(&audio->si, inv->pool, app.med_endpt,
					  local_sdp, remote_sdp, 0);
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "Error creating stream info from SDP", status);
	return;
    }

    /* Get the remainder of codec information from codec descriptor */
    if (audio->si.fmt.pt == app.audio_codec.pt)
	codec_desc = &app.audio_codec;
    else {
	/* Find the codec description in codec array */
	for (i=0; i<PJ_ARRAY_SIZE(audio_codecs); ++i) {
	    if (audio_codecs[i].pt == audio->si.fmt.pt) {
		codec_desc = &audio_codecs[i];
		break;
	    }
	}

	if (codec_desc == NULL) {
	    PJ_LOG(3, (THIS_FILE, "Error: Invalid codec payload type"));
	    return;
	}
    }

    audio->clock_rate = audio->si.fmt.clock_rate;
    audio->samples_per_frame = audio->clock_rate * codec_desc->ptime / 1000;
    audio->bytes_per_frame = codec_desc->bit_rate * codec_desc->ptime / 1000 / 8;


    pjmedia_rtp_session_init(&audio->out_sess, audio->si.tx_pt, 
			     pj_rand());
    pjmedia_rtp_session_init(&audio->in_sess, audio->si.fmt.pt, 0);
    pjmedia_rtcp_init(&audio->rtcp, "rtcp", audio->clock_rate, 
		      audio->samples_per_frame, 0);


    /* Attach media to transport */
    status = pjmedia_transport_attach(audio->transport, audio, 
				      &audio->si.rem_addr, 
				      &audio->si.rem_rtcp,
				      sizeof(pj_sockaddr_in),
				      &on_rx_rtp,
				      &on_rx_rtcp);
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "Error on pjmedia_transport_attach()", status);
	return;
    }

    /* Set the media as active */
    audio->active = PJ_TRUE;

    /* Immediately schedule to send the first RTP packet. */
    audio->rtp_timer.id = 1;
    interval.sec = interval.msec = 0;
    pjsip_endpt_schedule_timer(app.sip_endpt, &audio->rtp_timer, &interval);

    /* And schedule the first RTCP packet */
    audio->rtcp_timer.id = 1;
    interval.sec = 4;
    interval.msec = (pj_rand() % 1000);
    pjsip_endpt_schedule_timer(app.sip_endpt, &audio->rtcp_timer, &interval);
}



/* Destroy call's media */
static void destroy_call_media(unsigned call_index)
{
    struct media_stream *audio = &app.call[call_index].media[0];

    if (audio->active) {

	audio->active = PJ_FALSE;

	if (audio->rtp_timer.id) {
	    audio->rtp_timer.id = 0;
	    pjsip_endpt_cancel_timer(app.sip_endpt, &audio->rtp_timer);
	}

	if (audio->rtcp_timer.id) {
	    audio->rtcp_timer.id = 0;
	    pjsip_endpt_cancel_timer(app.sip_endpt, &audio->rtcp_timer);
	}

	pjmedia_transport_detach(audio->transport, audio);
    }
}


/*****************************************************************************
 * USER INTERFACE STUFFS
 */
#include "siprtp_report.c"


static void list_calls()
{
    unsigned i;
    puts("List all calls:");
    for (i=0; i<app.max_calls; ++i) {
	if (!app.call[i].inv)
	    continue;
	print_call(i);
    }
}

static void hangup_call(unsigned index)
{
    pjsip_tx_data *tdata;
    pj_status_t status;

    if (app.call[index].inv == NULL)
	return;

    status = pjsip_inv_end_session(app.call[index].inv, 603, NULL, &tdata);
    if (status==PJ_SUCCESS && tdata!=NULL)
	pjsip_inv_send_msg(app.call[index].inv, tdata);
}

static void hangup_all_calls()
{
    unsigned i;
    for (i=0; i<app.max_calls; ++i) {
	if (!app.call[i].inv)
	    continue;
	hangup_call(i);
    }
    
    /* Wait until all calls are terminated */
    for (i=0; i<app.max_calls; ++i) {
	while (app.call[i].inv)
	    pj_thread_sleep(10);
    }
}

static pj_bool_t simple_input(const char *title, char *buf, pj_size_t len)
{
    char *p;

    printf("%s (empty to cancel): ", title); fflush(stdout);
    fgets(buf, len, stdin);

    /* Remove trailing newlines. */
    for (p=buf; ; ++p) {
	if (*p=='\r' || *p=='\n') *p='\0';
	else if (!*p) break;
    }

    if (!*buf)
	return PJ_FALSE;
    
    return PJ_TRUE;
}


static const char *MENU =
"\n"
"Enter menu character:\n"
"  l    List all calls\n"
"  h    Hangup a call\n"
"  H    Hangup all calls\n"
"  q    Quit\n"
"\n";


/* Main screen menu */
static void console_main()
{
    char input1[10];
    unsigned i;

    printf("%s", MENU);

    for (;;) {
	printf(">>> "); fflush(stdout);
	fgets(input1, sizeof(input1), stdin);

	switch (input1[0]) {
	case 'l':
	    list_calls();
	    break;

	case 'h':
	    if (!simple_input("Call number to hangup", input1, sizeof(input1)))
		break;

	    i = atoi(input1);
	    hangup_call(i);
	    break;

	case 'H':
	    hangup_all_calls();
	    break;

	case 'q':
	    goto on_exit;

	default:
	    puts("Invalid command");
	    printf("%s", MENU);
	    break;
	}

	fflush(stdout);
    }

on_exit:
    hangup_all_calls();
}


/*****************************************************************************
 * Below is a simple module to log all incoming and outgoing SIP messages
 */


/* Notification on incoming messages */
static pj_bool_t logger_on_rx_msg(pjsip_rx_data *rdata)
{
    PJ_LOG(4,(THIS_FILE, "RX %d bytes %s from %s:%d:\n"
			 "%s\n"
			 "--end msg--",
			 rdata->msg_info.len,
			 pjsip_rx_data_get_info(rdata),
			 rdata->pkt_info.src_name,
			 rdata->pkt_info.src_port,
			 rdata->msg_info.msg_buf));
    
    /* Always return false, otherwise messages will not get processed! */
    return PJ_FALSE;
}

/* Notification on outgoing messages */
static pj_status_t logger_on_tx_msg(pjsip_tx_data *tdata)
{
    
    /* Important note:
     *	tp_info field is only valid after outgoing messages has passed
     *	transport layer. So don't try to access tp_info when the module
     *	has lower priority than transport layer.
     */

    PJ_LOG(4,(THIS_FILE, "TX %d bytes %s to %s:%d:\n"
			 "%s\n"
			 "--end msg--",
			 (tdata->buf.cur - tdata->buf.start),
			 pjsip_tx_data_get_info(tdata),
			 tdata->tp_info.dst_name,
			 tdata->tp_info.dst_port,
			 tdata->buf.start));

    /* Always return success, otherwise message will not get sent! */
    return PJ_SUCCESS;
}

/* The module instance. */
static pjsip_module msg_logger = 
{
    NULL, NULL,				/* prev, next.		*/
    { "mod-siprtp-log", 14 },		/* Name.		*/
    -1,					/* Id			*/
    PJSIP_MOD_PRIORITY_TRANSPORT_LAYER-1,/* Priority	        */
    NULL,				/* load()		*/
    NULL,				/* start()		*/
    NULL,				/* stop()		*/
    NULL,				/* unload()		*/
    &logger_on_rx_msg,			/* on_rx_request()	*/
    &logger_on_rx_msg,			/* on_rx_response()	*/
    &logger_on_tx_msg,			/* on_tx_request.	*/
    &logger_on_tx_msg,			/* on_tx_response()	*/
    NULL,				/* on_tsx_state()	*/

};



/*****************************************************************************
 * Console application custom logging:
 */


static FILE *log_file;


static void app_log_writer(int level, const char *buffer, int len)
{
    /* Write to both stdout and file. */

    if (level <= app.app_log_level)
	pj_log_write(level, buffer, len);

    if (log_file) {
	fwrite(buffer, len, 1, log_file);
	fflush(log_file);
    }
}


pj_status_t app_logging_init(void)
{
    /* Redirect log function to ours */

    pj_log_set_log_func( &app_log_writer );

    /* If output log file is desired, create the file: */

    if (app.log_filename) {
	log_file = fopen(app.log_filename, "wt");
	if (log_file == NULL) {
	    PJ_LOG(1,(THIS_FILE, "Unable to open log file %s", 
		      app.log_filename));   
	    return -1;
	}
    }

    return PJ_SUCCESS;
}


void app_logging_shutdown(void)
{
    /* Close logging file, if any: */

    if (log_file) {
	fclose(log_file);
	log_file = NULL;
    }
}


/*
 * main()
 */
int main(int argc, char *argv[])
{
    unsigned i;
    pj_status_t status;

    /* Must init PJLIB first */
    status = pj_init();
    if (status != PJ_SUCCESS)
	return 1;

    /* Get command line options */
    status = init_options(argc, argv);
    if (status != PJ_SUCCESS)
	return 1;

    /* Verify options: */

    /* Auto-quit can not be specified for UAS */
    if (app.auto_quit && app.uri_to_call.slen == 0) {
	printf("Error: --auto-quit option only valid for outgoing "
	       "mode (UAC) only\n");
	return 1;
    }

    /* Init logging */
    status = app_logging_init();
    if (status != PJ_SUCCESS)
	return 1;

    /* Init SIP etc */
    status = init_sip();
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "Initialization has failed", status);
	destroy_sip();
	return 1;
    }

    /* Register module to log incoming/outgoing messages */
    pjsip_endpt_register_module(app.sip_endpt, &msg_logger);

    /* Init media */
    status = init_media();
    if (status != PJ_SUCCESS) {
	app_perror(THIS_FILE, "Media initialization failed", status);
	destroy_sip();
	return 1;
    }

    /* Start worker threads */
    for (i=0; i<app.thread_count; ++i) {
	pj_thread_create( app.pool, "app", &sip_worker_thread, NULL,
			  0, 0, &app.sip_thread[i]);
    }

    /* If URL is specified, then make call immediately */
    if (app.uri_to_call.slen) {
	unsigned i;

	PJ_LOG(3,(THIS_FILE, "Making %d calls to %s..", app.max_calls,
		  app.uri_to_call.ptr));

	for (i=0; i<app.max_calls; ++i) {
	    status = make_call(&app.uri_to_call);
	    if (status != PJ_SUCCESS) {
		app_perror(THIS_FILE, "Error making call", status);
		break;
	    }
	}

	if (app.auto_quit) {
	    /* Wait for calls to complete */
	    while (app.uac_calls < app.max_calls)
		pj_thread_sleep(100);
	    pj_thread_sleep(200);
	} else {
	    /* Start user interface loop */
	    console_main();
	}

    } else {

	PJ_LOG(3,(THIS_FILE, "Ready for incoming calls (max=%d)", 
		  app.max_calls));

	/* Start user interface loop */
	console_main();

    }

    
    /* Shutting down... */
    destroy_media();
    destroy_sip();

    if (app.pool) {
	pj_pool_release(app.pool);
	app.pool = NULL;
	pj_caching_pool_destroy(&app.cp);
    }

    app_logging_shutdown();


    return 0;
}