コード例 #1
0
ファイル: sipecho.c プロジェクト: xhook/asterisk-v11
static pj_status_t init_stack()
{
    pj_sockaddr addr;
    pjsip_inv_callback inv_cb;
    pj_status_t status;

    pj_log_set_level(5);

    status = pj_init();
    CHECK_STATUS();

    pj_log_set_level(3);

    status = pjlib_util_init();
    CHECK_STATUS();

    pj_caching_pool_init(&app.cp, NULL, 0);
    app.pool = pj_pool_create( &app.cp.factory, "sipecho", 512, 512, 0);

    status = pjsip_endpt_create(&app.cp.factory, NULL, &app.sip_endpt);
    CHECK_STATUS();

    pj_log_set_level(4);
    pj_sockaddr_init(AF, &addr, NULL, (pj_uint16_t)SIP_PORT);
    if (AF == pj_AF_INET()) {
	status = pjsip_udp_transport_start( app.sip_endpt, &addr.ipv4, NULL,
					    1, NULL);
    } else if (AF == pj_AF_INET6()) {
	status = pjsip_udp_transport_start6(app.sip_endpt, &addr.ipv6, NULL,
					    1, NULL);
    } else {
	status = PJ_EAFNOTSUP;
    }

    pj_log_set_level(3);
    CHECK_STATUS();

    status = pjsip_tsx_layer_init_module(app.sip_endpt) ||
	     pjsip_ua_init_module( app.sip_endpt, NULL );
    CHECK_STATUS();

    pj_bzero(&inv_cb, 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;
    inv_cb.on_rx_offer = &call_on_rx_offer;

    status = pjsip_inv_usage_init(app.sip_endpt, &inv_cb) ||
	     pjsip_100rel_init_module(app.sip_endpt) ||
	     pjsip_endpt_register_module( app.sip_endpt, &mod_sipecho) ||
	     pjsip_endpt_register_module( app.sip_endpt, &msg_logger) ||
	     //pjmedia_endpt_create(&app.cp.factory,
		//		  pjsip_endpt_get_ioqueue(app.sip_endpt),
		//		  0, &app.med_endpt) ||
             pj_thread_create(app.pool, "sipecho", &worker_proc, NULL, 0, 0,
                              &app.worker_thread);
    CHECK_STATUS();

    return PJ_SUCCESS;
}
コード例 #2
0
int inv_offer_answer_test(void)
{
    unsigned i;
    int rc = 0;

    /* Init UA layer */
    if (pjsip_ua_instance()->id == -1) {
	pjsip_ua_init_param ua_param;
	pj_bzero(&ua_param, sizeof(ua_param));
	ua_param.on_dlg_forked = &on_dlg_forked;
	pjsip_ua_init_module(endpt, &ua_param);
    }

    /* Init inv-usage */
    if (pjsip_inv_usage_instance()->id == -1) {
	pjsip_inv_callback inv_cb;
	pj_bzero(&inv_cb, sizeof(inv_cb));
	inv_cb.on_media_update = &on_media_update;
	inv_cb.on_rx_offer = &on_rx_offer;
	inv_cb.on_create_offer = &on_create_offer;
	inv_cb.on_state_changed = &on_state_changed;
	inv_cb.on_new_session = &on_new_session;
	pjsip_inv_usage_init(endpt, &inv_cb);
    }

    /* 100rel module */
    pjsip_100rel_init_module(endpt);

    /* Our module */
    pjsip_endpt_register_module(endpt, &mod_inv_oa_test);
    pjsip_endpt_register_module(endpt, &mod_msg_logger);

    /* Create SIP UDP transport */
    {
	pj_sockaddr_in addr;
	pjsip_transport *tp;
	pj_status_t status;

	pj_sockaddr_in_init(&addr, NULL, PORT);
	status = pjsip_udp_transport_start(endpt, &addr, NULL, 1, &tp);
	pj_assert(status == PJ_SUCCESS);
    }

    /* Do tests */
    for (i=0; i<PJ_ARRAY_SIZE(test_params); ++i) {
	rc = perform_test(&test_params[i]);
	if (rc != 0)
	    goto on_return;
    }


on_return:
    return rc;
}
コード例 #3
0
ファイル: simpleua.c プロジェクト: Archipov/android-client
/*
 * main()
 *
 * If called with argument, treat argument as SIP URL to be called.
 * Otherwise wait for incoming calls.
 */
int main(int argc, char *argv[])
{
    pj_pool_t *pool = NULL;
    pj_status_t status;
    unsigned i;

    /* Must init PJLIB first: */
    status = pj_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

    pj_log_set_level(5);

    /* Then init PJLIB-UTIL: */
    status = pjlib_util_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


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


    /* Create global endpoint: */
    {
	const pj_str_t *hostname;
	const char *endpt_name;

	/* Endpoint MUST be assigned a globally unique name.
	 * The name will be used as the hostname in Warning header.
	 */

	/* For this implementation, we'll use hostname for simplicity */
	hostname = pj_gethostname();
	endpt_name = hostname->ptr;

	/* Create the endpoint: */

	status = pjsip_endpt_create(&cp.factory, endpt_name, 
				    &g_endpt);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }


    /* 
     * Add UDP transport, with hard-coded port 
     * Alternatively, application can use pjsip_udp_transport_attach() to
     * start UDP transport, if it already has an UDP socket (e.g. after it
     * resolves the address with STUN).
     */
    {
	pj_sockaddr addr;

	pj_sockaddr_init(AF, &addr, NULL, (pj_uint16_t)SIP_PORT);
	
	if (AF == pj_AF_INET()) {
	    status = pjsip_udp_transport_start( g_endpt, &addr.ipv4, NULL, 
						1, NULL);
	} else if (AF == pj_AF_INET6()) {
	    status = pjsip_udp_transport_start6(g_endpt, &addr.ipv6, NULL,
						1, NULL);
	} else {
	    status = PJ_EAFNOTSUP;
	}

	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to start UDP transport", status);
	    return 1;
	}
    }


    /* 
     * Init transaction layer.
     * This will create/initialize transaction hash tables etc.
     */
    status = pjsip_tsx_layer_init_module(g_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    /* 
     * Initialize UA layer module.
     * This will create/initialize dialog hash tables etc.
     */
    status = pjsip_ua_init_module( g_endpt, NULL );
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    /* 
     * Init invite session module.
     * The invite session module initialization takes additional argument,
     * i.e. a structure containing callbacks to be called on specific
     * occurence of events.
     *
     * The on_state_changed and on_new_session callbacks are mandatory.
     * Application must supply the callback function.
     *
     * We use on_media_update() callback in this application to start
     * media transmission.
     */
    {
	pjsip_inv_callback inv_cb;

	/* Init the callback for INVITE session: */
	pj_bzero(&inv_cb, 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(g_endpt, &inv_cb);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }

    /* Initialize 100rel support */
    status = pjsip_100rel_init_module(g_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /*
     * Register our module to receive incoming requests.
     */
    status = pjsip_endpt_register_module( g_endpt, &mod_simpleua);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

    /*
     * Register message logger module.
     */
    status = pjsip_endpt_register_module( g_endpt, &msg_logger);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    /* 
     * Initialize media endpoint.
     * This will implicitly initialize PJMEDIA too.
     */
#if PJ_HAS_THREADS
    status = pjmedia_endpt_create(&cp.factory, NULL, 1, &g_med_endpt);
#else
    status = pjmedia_endpt_create(&cp.factory, 
				  pjsip_endpt_get_ioqueue(g_endpt), 
				  0, &g_med_endpt);
#endif
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

    /* 
     * Add PCMA/PCMU codec to the media endpoint. 
     */
#if defined(PJMEDIA_HAS_G711_CODEC) && PJMEDIA_HAS_G711_CODEC!=0
    status = pjmedia_codec_g711_init(g_med_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
#endif


#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
    /* Init video subsystem */
    pool = pjmedia_endpt_create_pool(g_med_endpt, "Video subsystem", 512, 512);
    status = pjmedia_video_format_mgr_create(pool, 64, 0, NULL);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    status = pjmedia_converter_mgr_create(pool, NULL);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    status = pjmedia_vid_codec_mgr_create(pool, NULL);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    status = pjmedia_vid_dev_subsys_init(&cp.factory);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

#  if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) && PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
    /* Init ffmpeg video codecs */
    status = pjmedia_codec_ffmpeg_vid_init(NULL, &cp.factory);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
#  endif  /* PJMEDIA_HAS_FFMPEG_VID_CODEC */

#endif	/* PJMEDIA_HAS_VIDEO */
    
    /* 
     * Create media transport used to send/receive RTP/RTCP socket.
     * One media transport is needed for each call. Application may
     * opt to re-use the same media transport for subsequent calls.
     */
    for (i = 0; i < PJ_ARRAY_SIZE(g_med_transport); ++i) {
	status = pjmedia_transport_udp_create3(g_med_endpt, AF, NULL, NULL, 
					       RTP_PORT + i*2, 0, 
					       &g_med_transport[i]);
	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to create media transport", status);
	    return 1;
	}

	/* 
	 * Get socket info (address, port) of the media transport. We will
	 * need this info to create SDP (i.e. the address and port info in
	 * the SDP).
	 */
	pjmedia_transport_info_init(&g_med_tpinfo[i]);
	pjmedia_transport_get_info(g_med_transport[i], &g_med_tpinfo[i]);

	pj_memcpy(&g_sock_info[i], &g_med_tpinfo[i].sock_info,
		  sizeof(pjmedia_sock_info));
    }

    /*
     * If URL is specified, then make call immediately.
     */
    if (argc > 1) {
	pj_sockaddr hostaddr;
	char hostip[PJ_INET6_ADDRSTRLEN+2];
	char temp[80];
	pj_str_t dst_uri = pj_str(argv[1]);
	pj_str_t local_uri;
	pjsip_dialog *dlg;
	pjmedia_sdp_session *local_sdp;
	pjsip_tx_data *tdata;

	if (pj_gethostip(AF, &hostaddr) != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to retrieve local host IP", status);
	    return 1;
	}
	pj_sockaddr_print(&hostaddr, hostip, sizeof(hostip), 2);

	pj_ansi_sprintf(temp, "<sip:simpleuac@%s:%d>", 
			hostip, SIP_PORT);
	local_uri = pj_str(temp);

	/* Create UAC dialog */
	status = pjsip_dlg_create_uac( pjsip_ua_instance(), 
				       &local_uri,  /* local URI */
				       &local_uri,  /* local Contact */
				       &dst_uri,    /* remote URI */
				       &dst_uri,    /* remote target */
				       &dlg);	    /* dialog */
	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to create UAC dialog", status);
	    return 1;
	}

	/* If we expect the outgoing INVITE to be challenged, then we should
	 * put the credentials in the dialog here, with something like this:
	 *
	    {
		pjsip_cred_info	cred[1];

		cred[0].realm	  = pj_str("sip.server.realm");
		cred[0].scheme    = pj_str("digest");
		cred[0].username  = pj_str("theuser");
		cred[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
		cred[0].data      = pj_str("thepassword");

		pjsip_auth_clt_set_credentials( &dlg->auth_sess, 1, cred);
	    }
	 *
	 */


	/* Get the SDP body to be put in the outgoing INVITE, by asking
	 * media endpoint to create one for us.
	 */
	status = pjmedia_endpt_create_sdp( g_med_endpt,	    /* the media endpt	*/
					   dlg->pool,	    /* pool.		*/
					   MAX_MEDIA_CNT,   /* # of streams	*/
					   g_sock_info,     /* RTP sock info	*/
					   &local_sdp);	    /* the SDP result	*/
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);



	/* Create the INVITE session, and pass the SDP returned earlier
	 * as the session's initial capability.
	 */
	status = pjsip_inv_create_uac( dlg, local_sdp, 0, &g_inv);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

	/* If we want the initial INVITE to travel to specific SIP proxies,
	 * then we should put the initial dialog's route set here. The final
	 * route set will be updated once a dialog has been established.
	 * To set the dialog's initial route set, we do it with something
	 * like this:
	 *
	    {
		pjsip_route_hdr route_set;
		pjsip_route_hdr *route;
		const pj_str_t hname = { "Route", 5 };
		char *uri = "sip:proxy.server;lr";

		pj_list_init(&route_set);

		route = pjsip_parse_hdr( dlg->pool, &hname, 
					 uri, strlen(uri),
					 NULL);
		PJ_ASSERT_RETURN(route != NULL, 1);
		pj_list_push_back(&route_set, route);

		pjsip_dlg_set_route_set(dlg, &route_set);
	    }
	 *
	 * Note that Route URI SHOULD have an ";lr" parameter!
	 */

	/* Create initial INVITE request.
	 * This INVITE request will contain a perfectly good request and 
	 * an SDP body as well.
	 */
	status = pjsip_inv_invite(g_inv, &tdata);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);



	/* 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(g_inv, tdata);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    } else {

	/* No URL to make call to */

	PJ_LOG(3,(THIS_FILE, "Ready to accept incoming calls..."));
    }


    /* Loop until one call is completed */
    for (;!g_complete;) {
	pj_time_val timeout = {0, 10};
	pjsip_endpt_handle_events(g_endpt, &timeout);
    }

    /* On exit, dump current memory usage: */
    dump_pool_usage(THIS_FILE, &cp);

    /* Destroy audio ports. Destroy the audio port first
     * before the stream since the audio port has threads
     * that get/put frames to the stream.
     */
    if (g_snd_port)
	pjmedia_snd_port_destroy(g_snd_port);

#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
    /* Destroy video ports */
    if (g_vid_capturer)
	pjmedia_vid_port_destroy(g_vid_capturer);
    if (g_vid_renderer)
	pjmedia_vid_port_destroy(g_vid_renderer);
#endif

    /* Destroy streams */
    if (g_med_stream)
	pjmedia_stream_destroy(g_med_stream);
#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
    if (g_med_vstream)
	pjmedia_vid_stream_destroy(g_med_vstream);

    /* Deinit ffmpeg codec */
#   if defined(PJMEDIA_HAS_FFMPEG_VID_CODEC) && PJMEDIA_HAS_FFMPEG_VID_CODEC!=0
    pjmedia_codec_ffmpeg_vid_deinit();
#   endif

#endif

    /* Destroy media transports */
    for (i = 0; i < MAX_MEDIA_CNT; ++i) {
	if (g_med_transport[i])
	    pjmedia_transport_close(g_med_transport[i]);
    }

    /* Deinit pjmedia endpoint */
    if (g_med_endpt)
	pjmedia_endpt_destroy(g_med_endpt);

    /* Deinit pjsip endpoint */
    if (g_endpt)
	pjsip_endpt_destroy(g_endpt);

    /* Release pool */
    if (pool)
	pj_pool_release(pool);

    return 0;
}
コード例 #4
0
/*
 * Init SIP stack
 */
static pj_status_t init_sip()
{
    pj_status_t status = -1;

    /* Add UDP/TCP transport. */
    {
	pj_sockaddr_in addr;
	pjsip_host_port addrname;
	const char *transport_type = NULL;

	pj_bzero(&addr, sizeof(addr));
	addr.sin_family = pj_AF_INET();
	addr.sin_addr.s_addr = 0;
	addr.sin_port = pj_htons((pj_uint16_t)app.local_port);

	if (app.local_addr.slen) {
	    addrname.host = app.local_addr;
	    addrname.port = 5060;
	} 
	if (app.local_port != 0)
	    addrname.port = app.local_port;

	if (0) {
#if defined(PJ_HAS_TCP) && PJ_HAS_TCP!=0
	} else if (app.use_tcp) {
	    pj_sockaddr_in local_addr;
	    pjsip_tpfactory *tpfactory;
	    
	    transport_type = "tcp";
	    pj_sockaddr_in_init(&local_addr, 0, (pj_uint16_t)app.local_port);
	    status = pjsip_tcp_transport_start(app.sip_endpt, &local_addr,
					       app.thread_count, &tpfactory);
	    if (status == PJ_SUCCESS) {
		app.local_addr = tpfactory->addr_name.host;
		app.local_port = tpfactory->addr_name.port;
	    }
#endif
	} else {
	    pjsip_transport *tp;

	    transport_type = "udp";
	    status = pjsip_udp_transport_start(app.sip_endpt, &addr, 
					       (app.local_addr.slen ? &addrname:NULL),
					       app.thread_count, &tp);
	    if (status == PJ_SUCCESS) {
		app.local_addr = tp->local_name.host;
		app.local_port = tp->local_name.port;
	    }

	}
	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to start transport", status);
	    return status;
	}

	app.local_uri.ptr = pj_pool_alloc(app.pool, 128);
	app.local_uri.slen = pj_ansi_sprintf(app.local_uri.ptr, 
				    	     "<sip:pjsip-perf@%.*s:%d;transport=%s>",
					     (int)app.local_addr.slen,
					     app.local_addr.ptr,
					     app.local_port,
					     transport_type);

	app.local_contact = app.local_uri;
    }

    /* 
     * 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);

    /* Initialize 100rel support */
    status = pjsip_100rel_init_module(app.sip_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

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

	/* Init the callback for INVITE session: */
	pj_bzero(&inv_cb, 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_test);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Register stateless server module */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_stateless_server);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /* Register default responder module */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_responder);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /* Register stateless server module */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_stateful_server);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Register call server module */
    status = pjsip_endpt_register_module( app.sip_endpt, &mod_call_server);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);


    /* Done */
    return PJ_SUCCESS;
}
コード例 #5
0
ファイル: sip_plugin.c プロジェクト: zetagor/multivpn
/*
	Esta función es invocada desde MAIN=>Plugin, al igual que en TCP
	Es invocada desde un thread.
	Básicamente lo que hace es bootear PJSIP y luego se queda loopeando eventos SIP (con un pseudo polling que tiene la LIB)
	Adicionalmente lanza el thread para controlar los eventos TUN 
	(el tráfico es bidireccional)
*/
int sip_start(int argc,char **argv)
{

    pj_status_t status;


    /* Must init PJLIB first: */
    status = pj_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

    pj_log_set_level(5);

    /* Then init PJLIB-UTIL: */
    status = pjlib_util_init();
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


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

    /* Create global endpoint: */
    {
	const pj_str_t *hostname;
	const char *endpt_name;

	/* Endpoint MUST be assigned a globally unique name.
	 * The name will be used as the hostname in Warning header.
	 */

	/* For this implementation, we'll use hostname for simplicity */
	hostname = pj_gethostname();
	endpt_name = hostname->ptr;

	/* Create the endpoint: */

	status = pjsip_endpt_create(&cp.factory, endpt_name, 
				    &g_endpt);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }


    /* 
     * Add UDP transport, with hard-coded port 
     * Alternatively, application can use pjsip_udp_transport_attach() to
     * start UDP transport, if it already has an UDP socket (e.g. after it
     * resolves the address with STUN).
     */
    {
	pj_sockaddr addr;

	pj_sockaddr_init(AF, &addr, NULL, (pj_uint16_t)SIP_PORT);
	

	if (status != PJ_SUCCESS) {
	    app_perror(THIS_FILE, "Unable to start UDP transport", status);
	    return 1;
	}
    }


    /* 
     * Init transaction layer.
     * This will create/initialize transaction hash tables etc.
     */
    status = pjsip_tsx_layer_init_module(g_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    /* 
     * Initialize UA layer module.
     * This will create/initialize dialog hash tables etc.
     */
    status = pjsip_ua_init_module( g_endpt, NULL );
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


	/*
		Esto seguramente no haga falta nada de esto de INVITE's ...
		
		*/
    /* 
     * Init invite session module.
     * The invite session module initialization takes additional argument,
     * i.e. a structure containing callbacks to be called on specific
     * occurence of events.
     *
     * The on_state_changed and on_new_session callbacks are mandatory.
     * Application must supply the callback function.
     *
     * We use on_media_update() callback in this application to start
     * media transmission.
     */
    {
	pjsip_inv_callback inv_cb;

	/* Init the callback for INVITE session: */
	pj_bzero(&inv_cb, 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(g_endpt, &inv_cb);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
    }

    /* Initialize 100rel support */
    status = pjsip_100rel_init_module(g_endpt);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);

    /*
     * Register our module to receive incoming requests.
     */
    status = pjsip_endpt_register_module( g_endpt, &mod_simpleua);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

    /*
     * Register message logger module.
     */
    status = pjsip_endpt_register_module( g_endpt, &msg_logger);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);

   


	debug(1,"SIP Plugin launching thread for looping events");
	pthread_create( &sip_plugin_event_thread, NULL, sip_loop_sip_events, NULL);
	    
	/*
		Aqui es donde estamos esperando a recibir datos del PIPE (que viene del TUN DRIVER)
		hacemos I/O BLOCK con select
	*/
	sip_loop_tun_events();
	
	// Aqui llegamos cuando ha muerto todo ;)

    return 0;
}