diff --git a/linphone/mediastreamer2/tests/Makefile.am b/linphone/mediastreamer2/tests/Makefile.am
index fe6dd8584..d55c96dc6 100644
--- a/linphone/mediastreamer2/tests/Makefile.am
+++ b/linphone/mediastreamer2/tests/Makefile.am
@@ -1,4 +1,4 @@
-noinst_PROGRAMS=echo ring mtudiscover
+noinst_PROGRAMS=echo ring mtudiscover bench
 
 if BUILD_VIDEO
 noinst_PROGRAMS+=videodisplay
@@ -8,6 +8,7 @@ echo_SOURCES=echo.c
 ring_SOURCES=ring.c
 videodisplay_SOURCES=videodisplay.c
 mtudiscover_SOURCES=mtudiscover.c
+becnh_SOURCES=bench.c
 
 libexec_PROGRAMS=mediastream
 
diff --git a/linphone/mediastreamer2/tests/bench.c b/linphone/mediastreamer2/tests/bench.c
new file mode 100644
index 000000000..2a2a7e974
--- /dev/null
+++ b/linphone/mediastreamer2/tests/bench.c
@@ -0,0 +1,361 @@
+
+/*
+mediastreamer2 library - modular sound and video processing and streaming
+Copyright (C) 2006  Simon MORLAT (simon.morlat@linphone.org)
+
+This program is free software; you can redistribute it and/or
+modify it under the terms of the GNU General Public License
+as published by the Free Software Foundation; either version 2
+of the License, or (at your option) any later version.
+
+This program is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+GNU General Public License for more details.
+
+You should have received a copy of the GNU General Public License
+along with this program; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
+*/
+
+
+#include "mediastreamer2/msticker.h"
+
+#include "mediastreamer2/msrtp.h"
+#include "mediastreamer2/msfileplayer.h"
+#include "mediastreamer2/msfilerec.h"
+
+#include <signal.h>
+
+#define MAX_RTP_SIZE	1500
+
+static int run=1;
+
+static void stop(int signum){
+	run=0;
+}
+
+struct test_session {
+	RtpSession *rtps;
+	
+	MSFilter *fplayer;
+	MSFilter *encoder;
+	MSFilter *rtpsend;
+	
+	MSFilter *rtprecv;
+	MSFilter *decoder;
+	MSFilter *frecorder;
+};
+
+struct bench_config {
+	int num_session;
+	int num_session_record;
+	
+	int port_origin;
+	char *ip_destination;
+	int port_destination;
+	
+	int payload;
+	int rate;
+	int ptime;
+	char *wavfile;
+	
+	MSTicker *ticker;
+	MSList *tsessions; /* list of struct test_session */
+};
+
+#define NUM_SESSION 50
+#define NUM_SESSION_RECORD 1
+
+struct bench_config cfg[] = {
+	{	NUM_SESSION,NUM_SESSION_RECORD,
+		8000,"127.0.0.1",9000,8,8000,20,"test1.wav",NULL,NULL	},
+	{	NUM_SESSION,NUM_SESSION_RECORD,
+		9000,"127.0.0.1",8000,8,8000,20,"test1.wav",NULL,NULL	},
+	
+	{	NUM_SESSION,NUM_SESSION_RECORD,
+		10000,"127.0.0.1",11000,8,8000,20,"test1.wav",NULL,NULL	},
+	{	NUM_SESSION,NUM_SESSION_RECORD,
+		11000,"127.0.0.1",10000,8,8000,20,"test1.wav",NULL,NULL	},
+	
+	{	0,0,0,'\0',0,0,0,0,NULL,NULL,NULL	},
+};
+
+RtpSession *create_duplex_rtpsession(int locport){
+	RtpSession *rtpr;
+	rtpr=rtp_session_new(RTP_SESSION_SENDRECV);
+	rtp_session_set_recv_buf_size(rtpr,MAX_RTP_SIZE);
+	rtp_session_set_scheduling_mode(rtpr,0);
+	rtp_session_set_blocking_mode(rtpr,0);
+	rtp_session_enable_adaptive_jitter_compensation(rtpr,FALSE);
+	rtp_session_set_symmetric_rtp(rtpr,TRUE);
+	rtp_session_set_local_addr(rtpr,"0.0.0.0",locport);
+	rtp_session_signal_connect(rtpr,"timestamp_jump",(RtpCallback)rtp_session_resync,(long)NULL);
+	rtp_session_signal_connect(rtpr,"ssrc_changed",(RtpCallback)rtp_session_resync,(long)NULL);
+	return rtpr;
+}
+
+int init_bench(struct bench_config *bench)
+{
+	PayloadType *pt;
+	int pos;
+	int val;
+	int count;
+	bench->ticker=ms_ticker_new();
+
+	count = 0;
+	/* creates the couple of encoder/decoder */
+	pt=rtp_profile_get_payload(&av_profile,bench->payload);
+	if (pt==NULL){
+		ms_error("audiostream.c: undefined payload type.");
+		return count;
+	}
+	if (pt->clock_rate!=8000 && pt->clock_rate!=16000 && pt->clock_rate!=32000){
+		ms_error("audiostream.c: wrong rate.");
+		return count;
+	}
+	for (pos=0;pos<bench->num_session;pos++)
+		{
+			struct test_session *ts = (struct test_session *)ortp_malloc(sizeof(struct test_session));
+			memset(ts, 0, sizeof(struct test_session));
+			
+			ts->rtps = create_duplex_rtpsession(bench->port_origin+pos*2);
+			if (ts->rtps==NULL)
+				{
+					ms_error("bench.c: cannot create rtp_session!");
+					ortp_free(ts);
+					return count;
+				}
+			
+			rtp_session_set_payload_type(ts->rtps,bench->payload);
+			rtp_session_set_remote_addr_full(ts->rtps,
+											 bench->ip_destination,
+											 bench->port_destination+pos*2,
+											 bench->port_destination+1+pos*2);
+			
+			ts->fplayer = ms_filter_new(MS_FILE_PLAYER_ID);
+			if (strstr(bench->wavfile, ".au")==NULL)
+				ts->encoder = ms_filter_create_encoder(pt->mime_type);
+			ts->rtpsend = ms_filter_new(MS_RTP_SEND_ID);
+			
+			ts->rtprecv = ms_filter_new(MS_RTP_RECV_ID);
+			ts->decoder = ms_filter_create_decoder(pt->mime_type);
+			ts->frecorder = ms_filter_new(MS_FILE_REC_ID);
+			
+			if ((ts->encoder==NULL && strstr(bench->wavfile, ".au")==NULL)
+				|| (ts->decoder==NULL )){
+				ms_error("bench.c: No decoder available for payload %i.",bench->payload);
+				if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+				if (ts->encoder) ms_filter_destroy(ts->encoder);
+				if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+				if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+				if (ts->decoder) ms_filter_destroy(ts->decoder);
+				if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+				ortp_free(ts);
+				return count;
+			}
+			if (ts->fplayer==NULL){
+				ms_error("bench.c: missing player filter.");
+				if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+				if (ts->encoder) ms_filter_destroy(ts->encoder);
+				if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+				if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+				if (ts->decoder) ms_filter_destroy(ts->decoder);
+				if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+				ortp_free(ts);
+				return count;
+			}
+			if (ts->frecorder==NULL){
+				ms_error("bench.c: missing recorder filter.");
+				if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+				if (ts->encoder) ms_filter_destroy(ts->encoder);
+				if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+				if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+				if (ts->decoder) ms_filter_destroy(ts->decoder);
+				if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+				ortp_free(ts);
+				return count;
+			}
+			if (ts->rtpsend==NULL){
+				ms_error("bench.c: missing rtpsend filter.");
+				if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+				if (ts->encoder) ms_filter_destroy(ts->encoder);
+				if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+				if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+				if (ts->decoder) ms_filter_destroy(ts->decoder);
+				if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+				ortp_free(ts);
+				return count;
+			}
+			if (ts->rtprecv==NULL){
+				ms_error("bench.c: missing rtprecv filter.");
+				if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+				if (ts->encoder) ms_filter_destroy(ts->encoder);
+				if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+				if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+				if (ts->decoder) ms_filter_destroy(ts->decoder);
+				if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+				ortp_free(ts);
+				return count;
+			}
+			
+			ms_filter_call_method(ts->rtpsend,MS_RTP_SEND_SET_SESSION,ts->rtps);
+			ms_filter_call_method(ts->rtprecv,MS_RTP_RECV_SET_SESSION,ts->rtps);
+			
+			ms_filter_call_method (ts->rtprecv, MS_FILTER_SET_SAMPLE_RATE,
+								   &pt->clock_rate);
+			
+			ms_filter_call_method (ts->frecorder, MS_FILTER_SET_SAMPLE_RATE,
+								   &pt->clock_rate);
+			
+			val = ms_filter_call_method(ts->fplayer,MS_FILE_PLAYER_OPEN,(void*)bench->wavfile);
+			if (val!=0)
+				{
+					ms_error("bench.c: Cannot open wav file (%s)", bench->wavfile);
+					if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+					if (ts->encoder) ms_filter_destroy(ts->encoder);
+					if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+					if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+					if (ts->decoder) ms_filter_destroy(ts->decoder);
+					if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+					ortp_free(ts);
+					return count;
+				}
+			
+			val=0;
+			ms_filter_call_method (ts->fplayer, MS_FILTER_GET_SAMPLE_RATE,
+								   &val);
+			if (val!=pt->clock_rate)
+				{
+					ms_error("bench.c: unsupported rate for wav file: codec=%i / file=%i",
+							 pt->clock_rate, val);
+					if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+					if (ts->encoder) ms_filter_destroy(ts->encoder);
+					if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+					if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+					if (ts->decoder) ms_filter_destroy(ts->decoder);
+					if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+					ortp_free(ts);
+					return count;
+				}
+			ms_filter_call_method (ts->fplayer, MS_FILTER_GET_NCHANNELS,
+								   &val);
+			
+			if (val!=1)
+				{
+					ms_error("bench.c: unsupported number of channel for wav file: codec=1 / file=%i",
+							 val);
+					if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+					if (ts->encoder) ms_filter_destroy(ts->encoder);
+					if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);				
+					if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+					if (ts->decoder) ms_filter_destroy(ts->decoder);
+					if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+					ortp_free(ts);
+					return count;
+				}
+			ms_filter_call_method_noarg(ts->fplayer,MS_FILE_PLAYER_START);
+			
+			if (strstr(bench->wavfile, ".au")==NULL)
+				{
+					ms_filter_link(ts->fplayer,0,ts->encoder,0);
+					ms_filter_link(ts->encoder,0,ts->rtpsend,0);
+				}
+			else
+				{
+					ms_filter_link(ts->fplayer,0,ts->rtpsend,0);
+				}
+			
+			ms_filter_link(ts->rtprecv,0,ts->decoder,0);
+			ms_filter_link(ts->decoder,0,ts->frecorder,0);
+			
+			ms_ticker_attach(bench->ticker,ts->fplayer);
+			ms_ticker_attach(bench->ticker,ts->rtprecv);
+			
+			if (pos < bench->num_session_record)
+			{
+				char rec_file[128];
+				snprintf(rec_file, sizeof(rec_file), "rec_%s_%i.wav",
+						 bench->ip_destination,
+						 bench->port_destination+pos*2);
+				ms_filter_call_method(ts->frecorder,MS_FILE_REC_OPEN,(void*)rec_file);
+				ms_filter_call_method_noarg(ts->frecorder,MS_FILE_REC_START);
+			}
+			
+			bench->tsessions = ms_list_append(bench->tsessions, (void*)ts);
+			count++;
+		}
+
+	return count;
+}
+
+static int uninit_bench(struct bench_config *bench)
+{
+	MSList *it;
+	for(it=bench->tsessions;it!=NULL;it=bench->tsessions){
+		struct test_session *ts = (struct test_session *)it->data;
+		bench->tsessions = ms_list_remove_link(bench->tsessions, it);
+
+		ms_ticker_detach(bench->ticker,ts->fplayer);
+		ms_ticker_detach(bench->ticker,ts->rtprecv);
+		
+		ms_filter_call_method_noarg(ts->frecorder,MS_FILE_REC_CLOSE);
+		
+		if (strstr(bench->wavfile, ".au")==NULL)
+			{
+				ms_filter_unlink(ts->fplayer,0,ts->encoder,0);
+				ms_filter_unlink(ts->encoder,0,ts->rtpsend,0);
+			}
+		else
+			{
+				ms_filter_unlink(ts->fplayer,0,ts->rtpsend,0);
+			}
+		
+		ms_filter_unlink(ts->rtprecv,0,ts->decoder,0);
+		ms_filter_unlink(ts->decoder,0,ts->frecorder,0);
+			
+		if (ts->fplayer) ms_filter_destroy(ts->fplayer);
+		if (ts->encoder) ms_filter_destroy(ts->encoder);
+		if (ts->rtpsend) ms_filter_destroy(ts->rtpsend);
+		
+		if (ts->rtprecv) ms_filter_destroy(ts->rtprecv);
+		if (ts->decoder) ms_filter_destroy(ts->decoder);
+		if (ts->frecorder) ms_filter_destroy(ts->frecorder);
+
+		ortp_free(ts);
+	}
+	
+	ms_ticker_destroy(bench->ticker);
+	return 0;
+}
+
+
+int main(int argc, char *argv[]){
+	int pos;
+	int count;
+	ortp_init();
+	ortp_set_log_level_mask(ORTP_MESSAGE|ORTP_WARNING|ORTP_ERROR|ORTP_FATAL);
+	ms_init();
+	
+	signal(SIGINT,stop);
+
+	count=0;
+	for (pos=0;cfg[pos].num_session!=0;pos++)
+		{
+			count = count + init_bench(&cfg[pos]);
+			sleep(10);
+		}
+
+	ms_message("Number of session started: %i.", count);
+	
+	while(run)
+		sleep(1);
+
+	for (pos=0;cfg[pos].num_session!=0;pos++)
+		{
+			uninit_bench(&cfg[pos]);
+		}
+
+	return 0;
+}
+