예제 #1
0
static RtpProfile *make_dummy_profile(int samplerate){
	RtpProfile *prof=rtp_profile_new("dummy");
	PayloadType *pt=payload_type_clone(&payload_type_l16_mono);
	pt->clock_rate=samplerate;
	rtp_profile_set_payload(prof,0,pt);
	return prof;
}
static void basic_audio_stream() {
	AudioStream * 	marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
	stats_t marielle_stats;
	AudioStream * 	margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE);
	stats_t margaux_stats;
	RtpProfile* profile = rtp_profile_new("default profile");

	reset_stats(&marielle_stats);
	reset_stats(&margaux_stats);

	rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);

	CU_ASSERT_EQUAL(audio_stream_start_full(margaux
											, profile
											, MARIELLE_IP
											, MARIELLE_RTP_PORT
											, MARIELLE_IP
											, MARIELLE_RTCP_PORT
											, 0
											, 50
											, NULL
											, RECORDED_8K_1S_FILE
											, NULL
											, NULL
											, 0),0);

	CU_ASSERT_EQUAL(audio_stream_start_full(marielle
											, profile
											, MARGAUX_IP
											, MARGAUX_RTP_PORT
											, MARGAUX_IP
											, MARGAUX_RTCP_PORT
											, 0
											, 50
											, HELLO_8K_1S_FILE
											, NULL
											, NULL
											, NULL
											, 0),0);

	ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);

	CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000));

	audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
	audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

	/* No packet loss is assumed */
	CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);

	audio_stream_stop(marielle);
	audio_stream_stop(margaux);

	unlink(RECORDED_8K_1S_FILE);
}
예제 #3
0
static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
	int bw;
	const MSList *elem;
	RtpProfile *prof=rtp_profile_new("Call profile");
	bool_t first=TRUE;
	int remote_bw=0;
	LinphoneCore *lc=call->core;
	int up_ptime=0;
	*used_pt=-1;
	
	for(elem=desc->payloads;elem!=NULL;elem=elem->next){
		PayloadType *pt=(PayloadType*)elem->data;
		int number;
		
		if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
			if (desc->type==SalAudio){
				linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
				up_ptime=linphone_core_get_upload_ptime(lc);
			}
			*used_pt=payload_type_get_number(pt);
			first=FALSE;
		}
		if (desc->bandwidth>0) remote_bw=desc->bandwidth;
		else if (md->bandwidth>0) {
			/*case where b=AS is given globally, not per stream*/
			remote_bw=md->bandwidth;
			if (desc->type==SalVideo){
				remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
			}
		}
		
		if (desc->type==SalAudio){			
				bw=get_min_bandwidth(call->audio_bw,remote_bw);
		}else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
		if (bw>0) pt->normal_bitrate=bw*1000;
		else if (desc->type==SalAudio){
			pt->normal_bitrate=-1;
		}
		if (desc->ptime>0){
			up_ptime=desc->ptime;
		}
		if (up_ptime>0){
			char tmp[40];
			snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
			payload_type_append_send_fmtp(pt,tmp);
		}
		number=payload_type_get_number(pt);
		if (rtp_profile_get_payload(prof,number)!=NULL){
			ms_warning("A payload type with number %i already exists in profile !",number);
		}else
			rtp_profile_set_payload(prof,number,pt);
	}
	return prof;
}
예제 #4
0
/*clone a profile and its payloads */
RtpProfile * rtp_profile_clone_full(RtpProfile *prof)
{
    int i;
    PayloadType *pt;
    RtpProfile *newprof=rtp_profile_new(prof->name);
    for (i=0; i<RTP_PROFILE_MAX_PAYLOADS; i++) {
        pt=rtp_profile_get_payload(prof,i);
        if (pt!=NULL) {
            rtp_profile_set_payload(newprof,i,payload_type_clone(pt));
        }
    }
    return newprof;
}
예제 #5
0
bool myAudioStream::init_stream()
{
    /** Init stream **/
    stream = (AudioStream *)ms_new0(AudioStream,1);
    if (stream == 0) {
        ms_error("Failed to create new stream");
        return false;
    }

    /** Configure stream **/
    stream->play_dtmfs = false;
    stream->use_gc = false;
    stream->use_agc = false;
    stream->use_ng = false;

    /** Init RTP session **/
    stream->session = rtp_session_new(RTP_SESSION_RECVONLY);
    if (stream->session == 0) {
        ms_error("Failed to create new RTP session");
        return false;
    }

    /** Configure RTP session **/
    /* Create profile to use in session */
    RtpProfile *rtp_profile = rtp_profile_new("My profile");
    if (rtp_profile == 0) {
        ms_error("Failed to create new RTP profile");
        return false;
    }
    rtp_session_set_profile(stream->session, rtp_profile);

    /* Define some payloads */
    rtp_profile_set_payload(rtp_profile,110,&payload_type_speex_nb);
    rtp_profile_set_payload(rtp_profile,111,&payload_type_speex_wb);
    rtp_profile_set_payload(rtp_profile,112,&payload_type_speex_uwb);

    /* Set local address and port */
    rtp_session_set_local_addr(stream->session, "0.0.0.0", 1337);

    return true;
}
예제 #6
0
Call * call_new(Sipomatic *root, eXosip_event_t *ev)
{
	Call *obj;
	char *sdpans;
	int status;
	sdp_message_t *sdp;
	sdp_context_t *sdpc;
	
	sdp=eXosip_get_sdp_info(ev->request);
	sdpc=sdp_handler_create_context(&sipomatic_sdp_handler,NULL,"sipomatic");
	obj=ms_new0(Call,1);
	obj->profile=rtp_profile_new("remote");
	eXosip_call_send_answer(ev->tid,100,NULL);
	sdp_context_set_user_pointer(sdpc,obj);
	sdpans=sdp_context_get_answer(sdpc,sdp);
	if (sdpans!=NULL){
		eXosip_call_send_answer(ev->tid,180,NULL);
		
	}else{
		status=sdp_context_get_status(sdpc);
		eXosip_call_send_answer(ev->tid,status,NULL);
		sdp_context_free(sdpc);
		rtp_profile_destroy(obj->profile);
		ms_free(obj);
		return NULL;
	}
	obj->sdpc=sdpc;
	obj->did=ev->did;
	obj->tid=ev->tid;
	obj->time=time(NULL);
	obj->audio_stream=NULL;
	obj->state=CALL_STATE_INIT;
	obj->eof=0;
	obj->root=root;
	root->calls=ms_list_append(root->calls,obj);
	return obj;
}
static void basic_audio_stream_base(	const char* marielle_local_ip
									, 	int marielle_local_rtp_port
									, 	int marielle_local_rtcp_port
									, 	const char*  margaux_local_ip
									, 	int margaux_local_rtp_port
									, 	int margaux_local_rtcp_port) {
	AudioStream * 	marielle = audio_stream_new2 (marielle_local_ip, marielle_local_rtp_port, marielle_local_rtcp_port);
	stats_t marielle_stats;
	AudioStream * 	margaux = audio_stream_new2 (margaux_local_ip, margaux_local_rtp_port,margaux_local_rtcp_port);
	stats_t margaux_stats;
	RtpProfile* profile = rtp_profile_new("default profile");
	char* hello_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_file_root(), HELLO_8K_1S_FILE);
	char* recorded_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_writable_dir(), RECORDED_8K_1S_FILE);
	int dummy=0;
	rtp_session_set_multicast_loopback(marielle->ms.sessions.rtp_session,TRUE);
    rtp_session_set_multicast_loopback(margaux->ms.sessions.rtp_session,TRUE);

    reset_stats(&marielle_stats);
	reset_stats(&margaux_stats);

	rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);


	CU_ASSERT_EQUAL(audio_stream_start_full(margaux
											, profile
											, ms_is_multicast(margaux_local_ip)?margaux_local_ip:marielle_local_ip
											, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:marielle_local_rtp_port
											, marielle_local_ip
											, marielle_local_rtcp_port
											, 0
											, 50
											, NULL
											, recorded_file
											, NULL
											, NULL
											, 0),0);

	CU_ASSERT_EQUAL(audio_stream_start_full(marielle
											, profile
											, margaux_local_ip
											, margaux_local_rtp_port
											, margaux_local_ip
											, margaux_local_rtcp_port
											, 0
											, 50
											, hello_file
											, NULL
											, NULL
											, NULL
											, 0),0);

	ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);

	CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000));

	/*make sure packets can cross from sender to receiver*/
	wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);

	audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
	audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

	/* No packet loss is assumed */
	CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);

	audio_stream_stop(marielle);
	audio_stream_stop(margaux);

	unlink(recorded_file);
	ms_free(recorded_file);
	ms_free(hello_file);
}
static void encrypted_audio_stream_base( bool_t change_ssrc,
										 bool_t change_send_key_in_the_middle
										,bool_t set_both_send_recv_key
										,bool_t send_key_first) {
	AudioStream * 	marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
	AudioStream * 	margaux = audio_stream_new (MARGAUX_RTP_PORT,MARGAUX_RTCP_PORT, FALSE);
	RtpProfile* profile = rtp_profile_new("default profile");
	char* hello_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_file_root(), HELLO_8K_1S_FILE);
	char* recorded_file = ms_strdup_printf("%s/%s", mediastreamer2_tester_get_writable_dir(), RECORDED_8K_1S_FILE);
	stats_t marielle_stats;
	stats_t margaux_stats;
	int dummy=0;

	if (ms_srtp_supported()) {
		reset_stats(&marielle_stats);
		reset_stats(&margaux_stats);

		rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);

		CU_ASSERT_EQUAL(audio_stream_start_full(margaux
				, profile
				, MARIELLE_IP
				, MARIELLE_RTP_PORT
				, MARIELLE_IP
				, MARIELLE_RTCP_PORT
				, 0
				, 50
				, NULL
				, recorded_file
				, NULL
				, NULL
				, 0),0);

		CU_ASSERT_EQUAL(audio_stream_start_full(marielle
				, profile
				, MARGAUX_IP
				, MARGAUX_RTP_PORT
				, MARGAUX_IP
				, MARGAUX_RTCP_PORT
				, 0
				, 50
				, hello_file
				, NULL
				, NULL
				, NULL
				, 0),0);

		if (send_key_first) {
			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);

			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);

		} else {
			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") ==0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") ==0);

			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);
			if (set_both_send_recv_key)
				CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "6jCLmtRkVW9E/BUuJtYj/R2z6+4iEe06/DWohQ9F") == 0);

		}

		ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);
		if (change_send_key_in_the_middle) {
			int dummy=0;
			wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,2000);
			CU_ASSERT_TRUE(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") == 0);
			CU_ASSERT_TRUE(media_stream_set_srtp_recv_key_b64(&(margaux->ms.sessions), MS_AES_128_SHA1_32, "eCYF4nYyCvmCpFWjUeDaxI2GWp2BzCRlIPfg52Te") ==0);
		}
		CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000));

		/*make sure packets can cross from sender to receiver*/
		wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);

		audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
		audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

		/* No packet loss is assumed */
		if (change_send_key_in_the_middle) {
			/*we can accept one or 2 error in such case*/
			CU_ASSERT_TRUE((marielle_stats.rtp.packet_sent-margaux_stats.rtp.packet_recv)<3);
		} else
			CU_ASSERT_EQUAL(marielle_stats.rtp.sent,margaux_stats.rtp.recv);

		if (change_ssrc) {
			audio_stream_stop(marielle);
			marielle = audio_stream_new (MARIELLE_RTP_PORT, MARIELLE_RTCP_PORT,FALSE);
			CU_ASSERT_EQUAL(audio_stream_start_full(marielle
							, profile
							, MARGAUX_IP
							, MARGAUX_RTP_PORT
							, MARGAUX_IP
							, MARGAUX_RTCP_PORT
							, 0
							, 50
							, hello_file
							, NULL
							, NULL
							, NULL
							, 0),0);
			CU_ASSERT_FATAL(media_stream_set_srtp_send_key_b64(&(marielle->ms.sessions), MS_AES_128_SHA1_32, "d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj") == 0);

			ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);

			CU_ASSERT_TRUE(wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,2,12000));

			/*make sure packets can cross from sender to receiver*/
			wait_for_until(&marielle->ms,&margaux->ms,&dummy,1,500);

			audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
			audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);

			/* No packet loss is assumed */
			CU_ASSERT_EQUAL(marielle_stats.rtp.sent*2,margaux_stats.rtp.recv);

		}

		unlink(recorded_file);
		ms_free(recorded_file);
		ms_free(hello_file);
	} else {
		ms_warning("srtp not available, skiping...");
	}
	audio_stream_stop(marielle);
	audio_stream_stop(margaux);
	rtp_profile_destroy(profile);

}
static void basic_audio_stream_base_2(	const char* marielle_local_ip
									  ,	const char* marielle_remote_ip
									  , int marielle_local_rtp_port
									  , int marielle_remote_rtp_port
									  ,	int marielle_local_rtcp_port
									  , int marielle_remote_rtcp_port
									  ,	const char*  margaux_local_ip
									  , const char*  margaux_remote_ip
									  ,	int margaux_local_rtp_port
									  , int margaux_remote_rtp_port
									  ,	int margaux_local_rtcp_port
									  , int margaux_remote_rtcp_port
									  , int lost_percentage) {
	AudioStream * 	marielle = audio_stream_new2 (_factory, marielle_local_ip, marielle_local_rtp_port, marielle_local_rtcp_port);
	stats_t marielle_stats;
	AudioStream * 	margaux = audio_stream_new2 (_factory, margaux_local_ip, margaux_local_rtp_port,margaux_local_rtcp_port);
	stats_t margaux_stats;
	RtpProfile* profile = rtp_profile_new("default profile");
	char* hello_file = bc_tester_res(HELLO_8K_1S_FILE);
	char* recorded_file = bc_tester_file(RECORDED_8K_1S_FILE);
	uint64_t marielle_rtp_sent=0;

	rtp_session_set_multicast_loopback(marielle->ms.sessions.rtp_session,TRUE);
	rtp_session_set_multicast_loopback(margaux->ms.sessions.rtp_session,TRUE);
	rtp_session_set_rtcp_report_interval(marielle->ms.sessions.rtp_session, 1000);
	rtp_session_set_rtcp_report_interval(margaux->ms.sessions.rtp_session, 1000);

	reset_stats(&marielle_stats);
	reset_stats(&margaux_stats);

	rtp_profile_set_payload (profile,0,&payload_type_pcmu8000);

	BC_ASSERT_EQUAL(audio_stream_start_full(margaux
											, profile
											, ms_is_multicast(margaux_local_ip)?margaux_local_ip:margaux_remote_ip
											, ms_is_multicast(margaux_local_ip)?margaux_local_rtp_port:margaux_remote_rtp_port
											, margaux_remote_ip
											, margaux_remote_rtcp_port
											, 0
											, 50
											, NULL
											, recorded_file
											, NULL
											, NULL
											, 0)
					,0, int, "%d");

	BC_ASSERT_EQUAL(audio_stream_start_full(marielle
											, profile
											, marielle_remote_ip
											, marielle_remote_rtp_port
											, marielle_remote_ip
											, marielle_remote_rtcp_port
											, 0
											, 50
											, hello_file
											, NULL
											, NULL
											, NULL
											, 0)
					,0, int, "%d");

	ms_filter_add_notify_callback(marielle->soundread, notify_cb, &marielle_stats,TRUE);

	wait_for_until(&marielle->ms,&margaux->ms,&marielle_stats.number_of_EndOfFile,1,12000);

	audio_stream_get_local_rtp_stats(marielle,&marielle_stats.rtp);
	audio_stream_get_local_rtp_stats(margaux,&margaux_stats.rtp);
	marielle_rtp_sent = marielle_stats.rtp.sent;


	if (rtp_session_rtcp_enabled(marielle->ms.sessions.rtp_session) && rtp_session_rtcp_enabled(margaux->ms.sessions.rtp_session)) {
		BC_ASSERT_GREATER_STRICT(rtp_session_get_round_trip_propagation(marielle->ms.sessions.rtp_session),0,float,"%f");
		BC_ASSERT_GREATER_STRICT(rtp_session_get_stats(marielle->ms.sessions.rtp_session)->recv_rtcp_packets,0,unsigned long long,"%llu");
	}
예제 #10
0
/*
 * This small program starts a video stream to either
 * - read an H264 video track from mkv file and stream it out with RTP to specified destination
 * - receive H264 RTP packets on a local port and record them into an mkv file
 */
int main(int argc, char *argv[]){
	const char *command;
	const char *file;
	const char *ip;
	int port;
	VideoStream *stream;
	RtpProfile *profile;
	PayloadType *pt;
	Mode mode = INVALID_MODE;
	int local_port = 7778;
	MSFactory *factory ;
	
	MSMediaStreamIO io = MS_MEDIA_STREAM_IO_INITIALIZER;
	int err;
	
	/*parse command line arguments*/
	
	if (argc<4) usage(argv[0]);
	
	command = argv[1];
	if (strcasecmp(command,"play")==0) mode = PLAY_MODE;
	else if (strcasecmp(command, "record")==0) mode = RECORD_MODE;
	else usage(argv[0]);
	
	file = argv[2];
	
	if (mode == PLAY_MODE){
		ip = argv[3];
		if (argc<5) usage(argv[0]);
		port = atoi(argv[4]);
	}else{
		local_port = atoi(argv[3]);
		ip = "127.0.0.1"; port = 9990; /*dummy destination address, we won't send anything here anyway*/
	}
	
	/*set a signal handler to interrupt the program cleanly*/
	signal(SIGINT,stop_handler);
	
	/*initialize mediastreamer2*/
	factory = ms_factory_new_with_voip();
	
	/*create the video stream */
	stream = video_stream_new(factory, local_port, local_port+1, FALSE);
	
	/*define its local input and outputs with the MSMediaStreamIO structure*/
	if (mode == PLAY_MODE){
		io.input.type = MSResourceFile;
		io.input.file = file; /*the file we want to stream out via rtp*/
		io.output.type = MSResourceFile;
		io.output.file = NULL; /*we don't set a record file in PLAY_MODE, we just want the received video stream to be ignored, if something is received*/
	}else{
		io.input.type = MSResourceFile;
		io.input.file = NULL; /*We don't want to send anything via RTP in RECORD_MODE*/
		io.output.type = MSResourceFile;
		io.output.file = file; /*The file to which we want to record the received video stream*/
	}
	
	/*define the RTP profile to use: in this case we just want to use H264 codec*/
	profile = rtp_profile_new("My RTP profile");
	pt = payload_type_clone(&payload_type_h264);
	rtp_profile_set_payload(profile, payload_type_number, pt); /*we assign H264 to payload type number payload_type_number*/
	
	media_stream_set_target_network_bitrate(&stream->ms, 500000); /*set a target IP bitrate in bits/second */
	
	/*By default, the VideoStream will show up a display window where the received video is played, with a local preview as well.
	 * If you don't need this, assign (void*)-1 as window id, which explicitely disable the display feature.*/
	
	/*video_stream_set_native_window_id(stream, (void*)-1);*/
	
	/*start the video stream, given the RtpProfile and "io" definition */
	err = video_stream_start_from_io(stream, profile, ip, port, ip, port+1, payload_type_number, &io);
	if (err !=0 ){
		fprintf(stderr,"Could not start video stream.");
		goto end;
	}
	/*Register an event handler on the player to be notified of end of file*/
	ms_filter_add_notify_callback(stream->source, on_end_of_play, NULL, FALSE);
	
	/*program's main loop*/
	while (active){
		/*handle video stream background activity. This is non blocking*/
		video_stream_iterate(stream);
		/*process event callbacks*/
		ms_event_queue_pump(ms_factory_get_event_queue(factory));
		ms_usleep(50000); /*pause 50ms to avoid busy loop*/
	}
	
end:
	/*stop and destroy the video stream object*/
	if (stream) video_stream_stop(stream);
	/*free the RTP profile and payload type inside*/
	if (profile) rtp_profile_destroy(profile);
	
	ms_factory_destroy(factory);
	
	return err;
}