sip-lab 1.2.4

package/src/pjmedia/src/pjmedia/simpleua_dtmfdet.c

@@ -0,0 +1,753 @@
+/* $Id: simpleua.c 2394 2008-12-23 17:27:53Z bennylp $ */
+/* 
+ * Copyright (C) 2008-2009 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 
+ */
+
+
+/**
+ * simpleua.c
+ *
+ * This is a very simple SIP user agent complete with media. The user
+ * agent should do a proper SDP negotiation and start RTP media once
+ * SDP negotiation has completed.
+ *
+ * This program does not register to SIP server.
+ *
+ * Capabilities to be demonstrated here:
+ *  - Basic call
+ *  - Should support IPv6 (not tested)
+ *  - UDP transport at port 5060 (hard coded)
+ *  - RTP socket at port 4000 (hard coded)
+ *  - proper SDP negotiation
+ *  - PCMA/PCMU codec only.
+ *  - Audio/media to sound device.
+ *
+ *
+ * Usage:
+ *  - To make outgoing call, start simpleua with the URL of remote
+ *    destination to contact.
+ *    E.g.:
+ *	 simpleua sip:user@remote
+ *
+ *  - Incoming calls will automatically be answered with 180, then 200.
+ *
+ * This program does not disconnect call.
+ *
+ * This program will quit once it has completed a single call.
+ */
+
+/* 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>
+
+/* For logging purpose. */
+#define THIS_FILE   "simpleua.c"
+
+#include "util.h"
+
+
+/* Settings */
+#define AF	    pj_AF_INET()     /* Change to pj_AF_INET6() for IPv6.
+				      * PJ_HAS_IPV6 must be enabled and
+				      * your system must support IPv6.  */
+#define SIP_PORT    5060	     /* Listening SIP port		*/
+#define RTP_PORT    4000	     /* RTP port			*/
+
+/*
+ * Static variables.
+ */
+
+static pj_bool_t	     g_complete;    /* Quit flag.		*/
+static pjsip_endpoint	    *g_endpt;	    /* SIP endpoint.		*/
+static pj_caching_pool	     cp;	    /* Global pool factory.	*/
+
+static pjmedia_endpt	    *g_med_endpt;   /* Media endpoint.		*/
+static pjmedia_transport_info g_med_tpinfo; /* Socket info for media	*/
+static pjmedia_transport    *g_med_transport;/* Media stream transport	*/
+
+/* Call variables: */
+static pjsip_inv_session    *g_inv;	    /* Current invite session.	*/
+static pjmedia_session	    *g_med_session; /* Call's media session.	*/
+static pjmedia_snd_port	    *g_snd_player;  /* Call's sound player	*/
+static pjmedia_snd_port	    *g_snd_rec;	    /* Call's sound recorder.	*/
+
+static pjmedia_port         *g_master_port;
+static pjmedia_port	    *g_dtmfdet;
+
+
+/*
+ * 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 );
+
+
+
+
+/* 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_simpleua =
+{
+    NULL, NULL,			    /* prev, next.		*/
+    { "mod-simpleua", 12 },	    /* 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()		*/
+};
+
+
+
+/*
+ * 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_status_t status;
+
+    /* Must init PJLIB first: */
+    status = pj_init();
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+
+    /* 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);
+
+
+    /* 
+     * 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
+
+    
+    /* 
+     * 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.
+     */
+    status = pjmedia_transport_udp_create3(g_med_endpt, AF, NULL, NULL, 
+				           RTP_PORT, 0, &g_med_transport);
+    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);
+    pjmedia_transport_get_info(g_med_transport, &g_med_tpinfo);
+
+
+    /*
+     * 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].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. The SDP will contain all
+	 * codecs that have been registered to it (in this case, only
+	 * PCMA and PCMU), plus telephony event.
+	 */
+	status = pjmedia_endpt_create_sdp( g_med_endpt,	    /* the media endpt	*/
+					   dlg->pool,	    /* pool.		*/
+					   1,		    /* # of streams	*/
+					   &g_med_tpinfo.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);
+
+    return 0;
+}
+
+
+
+/*
+ * Callback when INVITE session state has changed.
+ * This callback is registered when the invite session module is initialized.
+ * We mostly want to know when the invite session has been disconnected,
+ * so that we can quit the application.
+ */
+static void call_on_state_changed( pjsip_inv_session *inv, 
+				   pjsip_event *e)
+{
+    PJ_UNUSED_ARG(e);
+
+    if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
+
+	PJ_LOG(3,(THIS_FILE, "Call DISCONNECTED [reason=%d (%s)]", 
+		  inv->cause,
+		  pjsip_get_status_text(inv->cause)->ptr));
+
+	PJ_LOG(3,(THIS_FILE, "One call completed, application quitting..."));
+	g_complete = 1;
+
+    } else {
+
+	PJ_LOG(3,(THIS_FILE, "Call state changed to %s", 
+		  pjsip_inv_state_name(inv->state)));
+
+    }
+}
+
+
+/* This callback is called when dialog has forked. */
+static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
+{
+    /* To be done... */
+    PJ_UNUSED_ARG(inv);
+    PJ_UNUSED_ARG(e);
+}
+
+
+/*
+ * Callback when incoming requests outside any transactions and any
+ * dialogs are received. We're only interested to hande incoming INVITE
+ * request, and we'll reject any other requests with 500 response.
+ */
+static pj_bool_t on_rx_request( pjsip_rx_data *rdata )
+{
+    pj_sockaddr hostaddr;
+    char temp[80], hostip[PJ_INET6_ADDRSTRLEN];
+    pj_str_t local_uri;
+    pjsip_dialog *dlg;
+    pjmedia_sdp_session *local_sdp;
+    pjsip_tx_data *tdata;
+    unsigned options = 0;
+    pj_status_t status;
+
+
+    /* 
+     * Respond (statelessly) any non-INVITE requests with 500 
+     */
+    if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
+
+	if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
+	    pj_str_t reason = pj_str("Simple UA unable to handle "
+				     "this request");
+
+	    pjsip_endpt_respond_stateless( g_endpt, rdata, 
+					   500, &reason,
+					   NULL, NULL);
+	}
+	return PJ_TRUE;
+    }
+
+
+    /*
+     * Reject INVITE if we already have an INVITE session in progress.
+     */
+    if (g_inv) {
+
+	pj_str_t reason = pj_str("Another call is in progress");
+
+	pjsip_endpt_respond_stateless( g_endpt, rdata, 
+				       500, &reason,
+				       NULL, NULL);
+	return PJ_TRUE;
+
+    }
+
+    /* Verify that we can handle the request. */
+    status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
+				      g_endpt, NULL);
+    if (status != PJ_SUCCESS) {
+
+	pj_str_t reason = pj_str("Sorry Simple UA can not handle this INVITE");
+
+	pjsip_endpt_respond_stateless( g_endpt, rdata, 
+				       500, &reason,
+				       NULL, NULL);
+	return PJ_TRUE;
+    } 
+
+    /*
+     * Generate Contact URI
+     */
+    if (pj_gethostip(AF, &hostaddr) != PJ_SUCCESS) {
+	app_perror(THIS_FILE, "Unable to retrieve local host IP", status);
+	return PJ_TRUE;
+    }
+    pj_sockaddr_print(&hostaddr, hostip, sizeof(hostip), 2);
+
+    pj_ansi_sprintf(temp, "<sip:simpleuas@%s:%d>", 
+		    hostip, SIP_PORT);
+    local_uri = pj_str(temp);
+
+    /*
+     * Create UAS dialog.
+     */
+    status = pjsip_dlg_create_uas( pjsip_ua_instance(), 
+				   rdata,
+				   &local_uri, /* contact */
+				   &dlg);
+    if (status != PJ_SUCCESS) {
+	pjsip_endpt_respond_stateless(g_endpt, rdata, 500, NULL,
+				      NULL, NULL);
+	return PJ_TRUE;
+    }
+
+    /* 
+     * Get media capability from media endpoint: 
+     */
+
+    status = pjmedia_endpt_create_sdp( g_med_endpt, rdata->tp_info.pool, 1,
+				       &g_med_tpinfo.sock_info, 
+				       &local_sdp);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+
+    /* 
+     * Create invite session, and pass both the UAS dialog and the SDP
+     * capability to the session.
+     */
+    status = pjsip_inv_create_uas( dlg, rdata, local_sdp, 0, &g_inv);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+
+    /*
+     * Initially send 180 response.
+     *
+     * The very first response to an INVITE must be created with
+     * pjsip_inv_initial_answer(). Subsequent responses to the same
+     * transaction MUST use pjsip_inv_answer().
+     */
+    status = pjsip_inv_initial_answer(g_inv, rdata, 
+				      180, 
+				      NULL, NULL, &tdata);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+
+    /* Send the 180 response. */  
+    status = pjsip_inv_send_msg(g_inv, tdata); 
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+
+    /*
+     * Now create 200 response.
+     */
+    status = pjsip_inv_answer( g_inv, 
+			       200, NULL,	/* st_code and st_text */
+			       NULL,		/* SDP already specified */
+			       &tdata);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+    /*
+     * Send the 200 response.
+     */
+    status = pjsip_inv_send_msg(g_inv, tdata);
+    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);
+
+
+    /* Done. 
+     * When the call is disconnected, it will be reported via the callback.
+     */
+
+    return PJ_TRUE;
+}
+
+static void dtmf_callback(pjmedia_port *port, void *user_data, char digit)
+{
+	printf("Digit arrived %c\n", digit);
+} 
+
+/*
+ * Callback when SDP negotiation has completed.
+ * We are interested with this callback because we want to start media
+ * as soon as SDP negotiation is completed.
+ */
+static void call_on_media_update( pjsip_inv_session *inv,
+				  pj_status_t status)
+{
+    pjmedia_session_info sess_info;
+    const pjmedia_sdp_session *local_sdp;
+    const pjmedia_sdp_session *remote_sdp;
+    pjmedia_port *media_port;
+
+    if (status != PJ_SUCCESS) {
+
+	app_perror(THIS_FILE, "SDP negotiation has failed", status);
+
+	/* Here we should disconnect call if we're not in the middle 
+	 * of initializing an UAS dialog and if this is not a re-INVITE.
+	 */
+	return;
+    }
+
+    /* Get local and remote SDP.
+     * We need both SDPs to create a media session.
+     */
+    status = pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
+
+    status = pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
+
+
+    /* Create session info based on the two SDPs. 
+     * We only support one stream per session for now.
+     */
+    status = pjmedia_session_info_from_sdp(inv->dlg->pool, g_med_endpt, 
+					   1, &sess_info, 
+					   local_sdp, remote_sdp);
+    if (status != PJ_SUCCESS) {
+	app_perror( THIS_FILE, "Unable to create media session", status);
+	return;
+    }
+
+    /* If required, we can also change some settings in the session info,
+     * (such as jitter buffer settings, codec settings, etc) before we
+     * create the session.
+     */
+
+    /* Create new media session, passing the two SDPs, and also the
+     * media socket that we created earlier.
+     * The media session is active immediately.
+     */
+    status = pjmedia_session_create( g_med_endpt, &sess_info,
+				     &g_med_transport, NULL, &g_med_session );
+    if (status != PJ_SUCCESS) {
+	app_perror( THIS_FILE, "Unable to create media session", status);
+	return;
+    }
+
+
+    /* Get the media port interface of the first stream in the session. 
+     * Media port interface is basicly a struct containing get_frame() and
+     * put_frame() function. With this media port interface, we can attach
+     * the port interface to conference bridge, or directly to a sound
+     * player/recorder device.
+     */
+    pjmedia_session_get_port(g_med_session, 0, &media_port);
+
+    status = pjmedia_dtmfdet_create(
+			inv->pool,
+			media_port->info.clock_rate,
+			media_port->info.channel_count,
+			media_port->info.samples_per_frame,
+			media_port->info.bits_per_sample,
+			dtmf_callback,
+			1234,
+			&g_dtmfdet);
+
+    printf("%i %i %i %i %X %X\n", media_port->info.clock_rate,
+			media_port->info.channel_count,
+			media_port->info.samples_per_frame,
+			media_port->info.bits_per_sample,
+			dtmf_callback,
+			g_dtmfdet);
+
+
+    if (status != PJ_SUCCESS) {
+	app_perror( THIS_FILE, "Unable to create dtmfdet", status);
+	PJ_LOG(3,(THIS_FILE, "%d %d %d %d",
+	    	    media_port->info.clock_rate,	/* clock rate	    */
+		    media_port->info.channel_count,	/* channel count    */
+		    media_port->info.samples_per_frame, /* samples per frame*/
+		    media_port->info.bits_per_sample    /* bits per sample  */
+	    ));
+	return;
+    }
+
+    status = pjmedia_master_port_create(inv->pool,
+					media_port,
+					g_dtmfdet,
+					0,
+					&g_master_port);
+    if (status != PJ_SUCCESS) {
+	app_perror( THIS_FILE, "Unable to create master port", status);
+	return;
+    }
+
+    status = pjmedia_master_port_start(g_master_port);
+    if (status != PJ_SUCCESS) {
+	app_perror( THIS_FILE, "Unable to start master port", status);
+	return;
+    }
+
+}
+
+