diff --git a/coreapi/callbacks.c b/coreapi/callbacks.c
index cb88095c7..efd17aab6 100644
--- a/coreapi/callbacks.c
+++ b/coreapi/callbacks.c
@@ -58,14 +58,14 @@ void linphone_core_update_streams_destinations(LinphoneCore *lc, LinphoneCall *c
 		rtp_addr = (new_audiodesc->rtp_addr[0] != '\0') ? new_audiodesc->rtp_addr : new_md->addr;
 		rtcp_addr = (new_audiodesc->rtcp_addr[0] != '\0') ? new_audiodesc->rtcp_addr : new_md->addr;
 		ms_message("Change audio stream destination: RTP=%s:%d RTCP=%s:%d", rtp_addr, new_audiodesc->rtp_port, rtcp_addr, new_audiodesc->rtcp_port);
-		rtp_session_set_remote_addr_full(call->audiostream->ms.session, rtp_addr, new_audiodesc->rtp_port, rtcp_addr, new_audiodesc->rtcp_port);
+		rtp_session_set_remote_addr_full(call->audiostream->ms.sessions.rtp_session, rtp_addr, new_audiodesc->rtp_port, rtcp_addr, new_audiodesc->rtcp_port);
 	}
 #ifdef VIDEO_ENABLED
 	if (call->videostream && new_videodesc) {
 		rtp_addr = (new_videodesc->rtp_addr[0] != '\0') ? new_videodesc->rtp_addr : new_md->addr;
 		rtcp_addr = (new_videodesc->rtcp_addr[0] != '\0') ? new_videodesc->rtcp_addr : new_md->addr;
 		ms_message("Change video stream destination: RTP=%s:%d RTCP=%s:%d", rtp_addr, new_videodesc->rtp_port, rtcp_addr, new_videodesc->rtcp_port);
-		rtp_session_set_remote_addr_full(call->videostream->ms.session, rtp_addr, new_videodesc->rtp_port, rtcp_addr, new_videodesc->rtcp_port);
+		rtp_session_set_remote_addr_full(call->videostream->ms.sessions.rtp_session, rtp_addr, new_videodesc->rtp_port, rtcp_addr, new_videodesc->rtcp_port);
 	}
 #else
 	(void)new_videodesc;
@@ -97,7 +97,7 @@ void linphone_core_update_streams(LinphoneCore *lc, LinphoneCall *call, SalMedia
 	sal_media_description_ref(new_md);
 	call->expect_media_in_ack=FALSE;
 	call->resultdesc=new_md;
-	if ((call->audiostream && call->audiostream->ms.ticker) || (call->videostream && call->videostream->ms.ticker)){
+	if ((call->audiostream && call->audiostream->ms.state==MSStreamStarted) || (call->videostream && call->videostream->ms.state==MSStreamStarted)){
 		/* we already started media: check if we really need to restart it*/
 		if (oldmd){
 			int md_changed = media_parameters_changed(call, oldmd, new_md);
@@ -671,7 +671,6 @@ static void call_failure(SalOp *op){
 				!linphone_core_is_media_encryption_mandatory(lc)) {
 				int i;
 				ms_message("Outgoing call [%p] failed with SRTP (SAVP) enabled",call);
-				linphone_call_stop_media_streams(call);
 				if (call->state==LinphoneCallOutgoingInit
 						|| call->state==LinphoneCallOutgoingProgress
 						|| call->state==LinphoneCallOutgoingRinging /*push case*/
diff --git a/coreapi/linphonecall.c b/coreapi/linphonecall.c
index e4cac3a33..c5f3ef3f2 100644
--- a/coreapi/linphonecall.c
+++ b/coreapi/linphonecall.c
@@ -191,13 +191,13 @@ void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t
 	if (call->audiostream==NULL){
 		ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
 	}
-	if (call->audiostream->ms.zrtp_context==NULL){
+	if (call->audiostream->ms.sessions.zrtp_context==NULL){
 		ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
 	}
 	if (!call->auth_token_verified && verified){
-		ortp_zrtp_sas_verified(call->audiostream->ms.zrtp_context);
+		ortp_zrtp_sas_verified(call->audiostream->ms.sessions.zrtp_context);
 	}else if (call->auth_token_verified && !verified){
-		ortp_zrtp_sas_reset_verified(call->audiostream->ms.zrtp_context);
+		ortp_zrtp_sas_reset_verified(call->audiostream->ms.sessions.zrtp_context);
 	}
 	call->auth_token_verified=verified;
 	propagate_encryption_changed(call);
@@ -333,8 +333,8 @@ void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *
 	strncpy(md->streams[0].rtp_addr,local_ip,sizeof(md->streams[0].rtp_addr));
 	strncpy(md->streams[0].rtcp_addr,local_ip,sizeof(md->streams[0].rtcp_addr));
 	strncpy(md->streams[0].name,"Audio",sizeof(md->streams[0].name)-1);
-	md->streams[0].rtp_port=call->audio_port;
-	md->streams[0].rtcp_port=call->audio_port+1;
+	md->streams[0].rtp_port=call->media_ports[0].rtp_port;
+	md->streams[0].rtcp_port=call->media_ports[0].rtcp_port;
 	md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
 		SalProtoRtpSavp : SalProtoRtpAvp;
 	md->streams[0].type=SalAudio;
@@ -350,8 +350,8 @@ void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *
 	if (call->params.has_video){
 		md->n_active_streams++;
 		strncpy(md->streams[0].name,"Video",sizeof(md->streams[0].name)-1);
-		md->streams[1].rtp_port=call->video_port;
-		md->streams[1].rtcp_port=call->video_port+1;
+		md->streams[1].rtp_port=call->media_ports[1].rtp_port;
+		md->streams[1].rtcp_port=call->media_ports[1].rtcp_port;
 		md->streams[1].proto=md->streams[0].proto;
 		md->streams[1].type=SalVideo;
 		l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL,-1);
@@ -393,34 +393,19 @@ void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *
 	}
 }
 
-static int find_port_offset(LinphoneCore *lc, SalStreamType type){
+static int find_port_offset(LinphoneCore *lc, int stream_index, int base_port){
 	int offset;
 	MSList *elem;
 	int tried_port;
 	int existing_port;
 	bool_t already_used=FALSE;
+	
 	for(offset=0;offset<100;offset+=2){
-		switch (type) {
-			default:
-			case SalAudio:
-				tried_port=linphone_core_get_audio_port (lc)+offset;
-				break;
-			case SalVideo:
-				tried_port=linphone_core_get_video_port (lc)+offset;
-				break;
-		}
+		tried_port=base_port+offset;
 		already_used=FALSE;
 		for(elem=lc->calls;elem!=NULL;elem=elem->next){
 			LinphoneCall *call=(LinphoneCall*)elem->data;
-			switch (type) {
-				default:
-				case SalAudio:
-					existing_port = call->audio_port;
-					break;
-				case SalVideo:
-					existing_port = call->video_port;
-					break;
-			}
+			existing_port=call->media_ports[stream_index].rtp_port;
 			if (existing_port==tried_port) {
 				already_used=TRUE;
 				break;
@@ -435,37 +420,19 @@ static int find_port_offset(LinphoneCore *lc, SalStreamType type){
 	return offset;
 }
 
-static int select_random_port(LinphoneCore *lc, SalStreamType type) {
+static int select_random_port(LinphoneCore *lc, int stream_index, int min_port, int max_port) {
 	MSList *elem;
 	int nb_tries;
 	int tried_port = 0;
 	int existing_port = 0;
-	int min_port = 0, max_port = 0;
 	bool_t already_used = FALSE;
 
-	switch (type) {
-		default:
-		case SalAudio:
-			linphone_core_get_audio_port_range(lc, &min_port, &max_port);
-			break;
-		case SalVideo:
-			linphone_core_get_video_port_range(lc, &min_port, &max_port);
-			break;
-	}
 	tried_port = (rand() % (max_port - min_port) + min_port) & ~0x1;
 	if (tried_port < min_port) tried_port = min_port + 2;
 	for (nb_tries = 0; nb_tries < 100; nb_tries++) {
 		for (elem = lc->calls; elem != NULL; elem = elem->next) {
 			LinphoneCall *call = (LinphoneCall *)elem->data;
-			switch (type) {
-				default:
-				case SalAudio:
-					existing_port = call->audio_port;
-					break;
-				case SalVideo:
-					existing_port = call->video_port;
-					break;
-			}
+			existing_port=call->media_ports[stream_index].rtp_port;
 			if (existing_port == tried_port) {
 				already_used = TRUE;
 				break;
@@ -480,8 +447,31 @@ static int select_random_port(LinphoneCore *lc, SalStreamType type) {
 	return tried_port;
 }
 
-static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
+static void port_config_set_random(LinphoneCall *call, int stream_index){
+	call->media_ports[stream_index].rtp_port=-1;
+	call->media_ports[stream_index].rtcp_port=-1;
+}
+
+static void port_config_set(LinphoneCall *call, int stream_index, int min_port, int max_port){
 	int port_offset;
+	if (min_port>0 && max_port>0){
+		if (min_port == max_port) {
+			/* Used fixed RTP audio port. */
+			port_offset=find_port_offset(call->core, stream_index, min_port);
+			if (port_offset==-1) {
+				port_config_set_random(call, stream_index);
+				return;
+			}
+			call->media_ports[stream_index].rtp_port=min_port+port_offset;
+		} else {
+			/* Select random RTP audio port in the specified range. */
+			call->media_ports[stream_index].rtp_port = select_random_port(call->core, stream_index, min_port, max_port);
+		}
+		call->media_ports[stream_index].rtcp_port=call->media_ports[stream_index].rtp_port+1;
+	}else port_config_set_random(call,stream_index);
+}
+
+static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
 	int min_port, max_port;
 
 	call->magic=linphone_call_magic;
@@ -494,25 +484,11 @@ static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from,
 	call->camera_enabled=TRUE;
 
 	linphone_core_get_audio_port_range(call->core, &min_port, &max_port);
-	if (min_port == max_port) {
-		/* Used fixed RTP audio port. */
-		port_offset=find_port_offset (call->core, SalAudio);
-		if (port_offset==-1) return;
-		call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
-	} else {
-		/* Select random RTP audio port in the specified range. */
-		call->audio_port = select_random_port(call->core, SalAudio);
-	}
+	port_config_set(call,0,min_port,max_port);
+	
 	linphone_core_get_video_port_range(call->core, &min_port, &max_port);
-	if (min_port == max_port) {
-		/* Used fixed RTP video port. */
-		port_offset=find_port_offset (call->core, SalVideo);
-		if (port_offset==-1) return;
-		call->video_port=linphone_core_get_video_port(call->core)+port_offset;
-	} else {
-		/* Select random RTP video port in the specified range. */
-		call->video_port = select_random_port(call->core, SalVideo);
-	}
+	port_config_set(call,1,min_port,max_port);
+	
 	linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
 	linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
 }
@@ -663,20 +639,22 @@ LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *fro
 		// In this case WE chose the media parameters according to policy.
 		call->params.has_video &= linphone_core_media_description_contains_video_stream(md);
 	}
-
+	/*create the ice session now if ICE is required*/
+	if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseIce){
+		call->ice_session = ice_session_new();
+		ice_session_set_role(call->ice_session, IR_Controlled);
+	}
+	/*reserve the sockets immediately*/
+	linphone_call_init_media_streams(call);
 	switch (linphone_core_get_firewall_policy(call->core)) {
 		case LinphonePolicyUseIce:
-			call->ice_session = ice_session_new();
-			ice_session_set_role(call->ice_session, IR_Controlled);
+			/*start ICE gathering*/
 			linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
-			if (call->ice_session != NULL) {
-				linphone_call_init_media_streams(call);
-				linphone_call_start_media_streams_for_ice_gathering(call);
-				if (linphone_core_gather_ice_candidates(call->core,call)<0) {
-					/* Ice candidates gathering failed, proceed with the call anyway. */
-					linphone_call_delete_ice_session(call);
-					linphone_call_stop_media_streams_for_ice_gathering(call);
-				}
+			linphone_call_start_media_streams_for_ice_gathering(call);
+			if (linphone_core_gather_ice_candidates(call->core,call)<0) {
+				/* Ice candidates gathering failed, proceed with the call anyway. */
+				linphone_call_delete_ice_session(call);
+				linphone_call_stop_media_streams_for_ice_gathering(call);
 			}
 			break;
 		case LinphonePolicyUseStun:
@@ -688,7 +666,6 @@ LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *fro
 			if(!lc->rtp_conf.disable_upnp) {
 				call->upnp_session = linphone_upnp_session_new(call);
 				if (call->upnp_session != NULL) {
-					linphone_call_init_media_streams(call);
 					if (linphone_core_update_upnp_from_remote_media_description(call, sal_call_get_remote_media_description(op))<0) {
 						/* uPnP port mappings failed, proceed with the call anyway. */
 						linphone_call_delete_upnp_session(call);
@@ -714,6 +691,11 @@ LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *fro
 static void linphone_call_set_terminated(LinphoneCall *call){
 	LinphoneCore *lc=call->core;
 
+	linphone_call_stop_media_streams(call);
+	ms_media_stream_sessions_uninit(&call->sessions[0]);
+	ms_media_stream_sessions_uninit(&call->sessions[1]);
+	linphone_call_delete_upnp_session(call);
+	linphone_call_delete_ice_session(call);
 	linphone_core_update_allocated_audio_bandwidth(lc);
 
 	call->owns_call_log=FALSE;
@@ -856,11 +838,6 @@ void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const
 static void linphone_call_destroy(LinphoneCall *obj)
 {
 	ms_message("Call [%p] freed.",obj);
-	linphone_call_stop_media_streams(obj);
-#ifdef BUILD_UPNP
-	linphone_call_delete_upnp_session(obj);
-#endif //BUILD_UPNP
-	linphone_call_delete_ice_session(obj);
 	if (obj->op!=NULL) {
 		sal_op_release(obj->op);
 		obj->op=NULL;
@@ -1444,17 +1421,31 @@ void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, Lin
 	call->nextVideoFrameDecoded._func = cb;
 	call->nextVideoFrameDecoded._user_data = user_data;
 #ifdef VIDEO_ENABLED
-	ms_filter_call_method_noarg(call->videostream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
+	if (call->videostream && call->videostream->ms.decoder)
+		ms_filter_call_method_noarg(call->videostream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
 #endif
 }
 
+static void port_config_set_random_choosed(LinphoneCall *call, int stream_index, RtpSession *session){
+	call->media_ports[stream_index].rtp_port=rtp_session_get_local_port(session);
+	call->media_ports[stream_index].rtcp_port=rtp_session_get_local_rtcp_port(session);
+}
+
 void linphone_call_init_audio_stream(LinphoneCall *call){
 	LinphoneCore *lc=call->core;
 	AudioStream *audiostream;
 	int dscp;
 
 	if (call->audiostream != NULL) return;
-	call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,call->af==AF_INET6);
+	if (call->sessions[0].rtp_session==NULL){
+		call->audiostream=audiostream=audio_stream_new(call->media_ports[0].rtp_port,call->media_ports[0].rtcp_port,call->af==AF_INET6);
+	}else{
+		call->audiostream=audio_stream_new_with_sessions(&call->sessions[0]);
+	}
+	audiostream=call->audiostream;
+	if (call->media_ports[0].rtp_port==-1){
+		port_config_set_random_choosed(call,0,audiostream->ms.sessions.rtp_session);
+	}
 	dscp=linphone_core_get_audio_dscp(lc);
 	if (dscp!=-1)
 		audio_stream_set_dscp(audiostream,dscp);
@@ -1486,69 +1477,73 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
 	audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
 
 	if (lc->rtptf){
-		RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
-		RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
-		rtp_session_set_transports(audiostream->ms.session,artp,artcp);
+		RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->media_ports[0].rtp_port);
+		RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->media_ports[0].rtcp_port);
+		rtp_session_set_transports(audiostream->ms.sessions.rtp_session,artp,artcp);
 	}
 	if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
-		rtp_session_set_pktinfo(audiostream->ms.session, TRUE);
-		rtp_session_set_symmetric_rtp(audiostream->ms.session, FALSE);
+		rtp_session_set_pktinfo(audiostream->ms.sessions.rtp_session, TRUE);
+		rtp_session_set_symmetric_rtp(audiostream->ms.sessions.rtp_session, FALSE);
 		if (ice_session_check_list(call->ice_session, 0) == NULL) {
 			ice_session_add_check_list(call->ice_session, ice_check_list_new());
 		}
 		audiostream->ms.ice_check_list = ice_session_check_list(call->ice_session, 0);
-		ice_check_list_set_rtp_session(audiostream->ms.ice_check_list, audiostream->ms.session);
+		ice_check_list_set_rtp_session(audiostream->ms.ice_check_list, audiostream->ms.sessions.rtp_session);
 	}
 
 	call->audiostream_app_evq = ortp_ev_queue_new();
-	rtp_session_register_event_queue(audiostream->ms.session,call->audiostream_app_evq);
+	rtp_session_register_event_queue(audiostream->ms.sessions.rtp_session,call->audiostream_app_evq);
 }
 
 void linphone_call_init_video_stream(LinphoneCall *call){
 #ifdef VIDEO_ENABLED
 	LinphoneCore *lc=call->core;
 
-	if (!call->params.has_video) {
-		linphone_call_stop_video_stream(call);
-		return;
-	}
-	if (call->videostream != NULL) return;
-	if ((lc->video_conf.display || lc->video_conf.capture) && call->params.has_video){
+	if (call->videostream == NULL){
 		int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
 		int dscp=linphone_core_get_video_dscp(lc);
 		const char *display_filter=linphone_core_get_video_display_filter(lc);
-
-		call->videostream=video_stream_new(call->video_port,call->video_port+1,call->af==AF_INET6);
+		
+		if (call->sessions[1].rtp_session==NULL){
+			call->videostream=video_stream_new(call->media_ports[1].rtp_port,call->media_ports[1].rtcp_port, call->af==AF_INET6);
+		}else{
+			call->videostream=video_stream_new_with_sessions(&call->sessions[1]);
+		}
+		if (call->media_ports[1].rtp_port==-1){
+			port_config_set_random_choosed(call,1,call->videostream->ms.sessions.rtp_session);
+		}
 		if (dscp!=-1)
 			video_stream_set_dscp(call->videostream,dscp);
 		video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
-		if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->ms.session,video_recv_buf_size);
+		if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->ms.sessions.rtp_session,video_recv_buf_size);
 
 		if (display_filter != NULL)
 			video_stream_set_display_filter_name(call->videostream,display_filter);
 		video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
 		if (lc->rtptf){
-			RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
-			RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
-			rtp_session_set_transports(call->videostream->ms.session,vrtp,vrtcp);
+			RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->media_ports[1].rtp_port);
+			RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->media_ports[1].rtcp_port);
+			rtp_session_set_transports(call->videostream->ms.sessions.rtp_session,vrtp,vrtcp);
 		}
-		if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
-			rtp_session_set_pktinfo(call->videostream->ms.session, TRUE);
-			rtp_session_set_symmetric_rtp(call->videostream->ms.session, FALSE);
-			if (ice_session_check_list(call->ice_session, 1) == NULL) {
-				ice_session_add_check_list(call->ice_session, ice_check_list_new());
-			}
-			call->videostream->ms.ice_check_list = ice_session_check_list(call->ice_session, 1);
-			ice_check_list_set_rtp_session(call->videostream->ms.ice_check_list, call->videostream->ms.session);
-			ms_message ("creating new ice video check list [%p] for session [%p]",call->videostream->ms.ice_check_list,call->videostream->ms.session);
-		}
-
-		call->videostream_app_evq = ortp_ev_queue_new();
-		rtp_session_register_event_queue(call->videostream->ms.session,call->videostream_app_evq);
+			call->videostream_app_evq = ortp_ev_queue_new();
+		rtp_session_register_event_queue(call->videostream->ms.sessions.rtp_session,call->videostream_app_evq);
 #ifdef TEST_EXT_RENDERER
 		video_stream_set_render_callback(call->videostream,rendercb,NULL);
 #endif
 	}
+	/*eventually re-create the ICE check list that may have been destroyed if the stream wasn't used recently*/
+	if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
+		rtp_session_set_pktinfo(call->videostream->ms.sessions.rtp_session, TRUE);
+		rtp_session_set_symmetric_rtp(call->videostream->ms.sessions.rtp_session, FALSE);
+		if (ice_session_check_list(call->ice_session, 1) == NULL) {
+			ice_session_add_check_list(call->ice_session, ice_check_list_new());
+		}
+		call->videostream->ms.ice_check_list = ice_session_check_list(call->ice_session, 1);
+		ice_check_list_set_rtp_session(call->videostream->ms.ice_check_list, call->videostream->ms.sessions.rtp_session);
+		ms_message ("creating new ice video check list [%p] for session [%p]",call->videostream->ms.ice_check_list,call->videostream->ms.sessions.rtp_session);
+	}
+
+
 #else
 	call->videostream=NULL;
 #endif
@@ -1779,9 +1774,9 @@ static void configure_rtp_session_for_rtcp_xr(LinphoneCore *lc, LinphoneCall *ca
 		memcpy(&currentconfig, remoteconfig, sizeof(currentconfig));
 	}
 	if (type == SalAudio) {
-		session = call->audiostream->ms.session;
+		session = call->audiostream->ms.sessions.rtp_session;
 	} else {
-		session = call->videostream->ms.session;
+		session = call->videostream->ms.sessions.rtp_session;
 	}
 	rtp_session_configure_rtcp_xr(session, &currentconfig);
 	if (currentconfig.rcvr_rtt_mode != OrtpRtcpXrRcvrRttNone) {
@@ -2166,14 +2161,16 @@ void linphone_call_delete_ice_session(LinphoneCall *call){
 	}
 }
 
-#ifdef BUILD_UPNP
+
 void linphone_call_delete_upnp_session(LinphoneCall *call){
+#ifdef BUILD_UPNP
 	if(call->upnp_session!=NULL) {
 		linphone_upnp_session_destroy(call->upnp_session);
 		call->upnp_session=NULL;
 	}
-}
 #endif //BUILD_UPNP
+}
+
 
 static void linphone_call_log_fill_stats(LinphoneCallLog *log, MediaStream *st){
 	float quality=media_stream_get_average_quality_rating(st);
@@ -2186,7 +2183,8 @@ static void linphone_call_log_fill_stats(LinphoneCallLog *log, MediaStream *st){
 
 void linphone_call_stop_audio_stream(LinphoneCall *call) {
 	if (call->audiostream!=NULL) {
-		rtp_session_unregister_event_queue(call->audiostream->ms.session,call->audiostream_app_evq);
+		media_stream_reclaim_sessions(&call->audiostream->ms,&call->sessions[0]);
+		rtp_session_unregister_event_queue(call->audiostream->ms.sessions.rtp_session,call->audiostream_app_evq);
 		ortp_ev_queue_flush(call->audiostream_app_evq);
 		ortp_ev_queue_destroy(call->audiostream_app_evq);
 		call->audiostream_app_evq=NULL;
@@ -2213,7 +2211,8 @@ void linphone_call_stop_audio_stream(LinphoneCall *call) {
 void linphone_call_stop_video_stream(LinphoneCall *call) {
 #ifdef VIDEO_ENABLED
 	if (call->videostream!=NULL){
-		rtp_session_unregister_event_queue(call->videostream->ms.session,call->videostream_app_evq);
+		media_stream_reclaim_sessions(&call->videostream->ms,&call->sessions[1]);
+		rtp_session_unregister_event_queue(call->videostream->ms.sessions.rtp_session,call->videostream_app_evq);
 		ortp_ev_queue_flush(call->videostream_app_evq);
 		ortp_ev_queue_destroy(call->videostream_app_evq);
 		call->videostream_app_evq=NULL;
@@ -2684,12 +2683,12 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 	if ((call->state==LinphoneCallStreamsRunning || call->state==LinphoneCallOutgoingEarlyMedia || call->state==LinphoneCallIncomingEarlyMedia) && one_second_elapsed){
 		float audio_load=0, video_load=0;
 		if (call->audiostream!=NULL){
-			if (call->audiostream->ms.ticker)
-				audio_load=ms_ticker_get_average_load(call->audiostream->ms.ticker);
+			if (call->audiostream->ms.sessions.ticker)
+				audio_load=ms_ticker_get_average_load(call->audiostream->ms.sessions.ticker);
 		}
 		if (call->videostream!=NULL){
-			if (call->videostream->ms.ticker)
-				video_load=ms_ticker_get_average_load(call->videostream->ms.ticker);
+			if (call->videostream->ms.sessions.ticker)
+				video_load=ms_ticker_get_average_load(call->videostream->ms.sessions.ticker);
 		}
 		report_bandwidth(call,(MediaStream*)call->audiostream,(MediaStream*)call->videostream);
 		ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
@@ -2716,7 +2715,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 			if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
 				linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
 			} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
-				call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->ms.session);
+				call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->ms.sessions.rtp_session);
 				if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
 					freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
 				call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
@@ -2726,7 +2725,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 				if (lc->vtable.call_stats_updated)
 					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
 			} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
-				memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->ms.session), sizeof(jitter_stats_t));
+				memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->ms.sessions.rtp_session), sizeof(jitter_stats_t));
 				if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
 					freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
 				call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
@@ -2761,7 +2760,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 			} else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
 				linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
 			} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
-				call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->ms.session);
+				call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->ms.sessions.rtp_session);
 				if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
 					freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
 				call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
@@ -2771,7 +2770,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 				if (lc->vtable.call_stats_updated)
 					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
 			} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
-				memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->ms.session), sizeof(jitter_stats_t));
+				memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->ms.sessions.rtp_session), sizeof(jitter_stats_t));
 				if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
 					freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
 				call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
diff --git a/coreapi/linphonecore.c b/coreapi/linphonecore.c
index 8fccd7561..6c609d037 100644
--- a/coreapi/linphonecore.c
+++ b/coreapi/linphonecore.c
@@ -1755,6 +1755,8 @@ void linphone_core_set_rtp_no_xmit_on_audio_mute(LinphoneCore *lc,bool_t rtp_no_
 
 /**
  * Sets the UDP port used for audio streaming.
+ * A value if -1 will request the system to allocate the local port randomly.
+ * This is recommended in order to avoid firewall warnings.
  *
  * @ingroup network_parameters
 **/
@@ -1775,6 +1777,8 @@ void linphone_core_set_audio_port_range(LinphoneCore *lc, int min_port, int max_
 
 /**
  * Sets the UDP port used for video streaming.
+ * A value if -1 will request the system to allocate the local port randomly.
+ * This is recommended in order to avoid firewall warnings.
  *
  * @ingroup network_parameters
 **/
@@ -2604,6 +2608,10 @@ int linphone_core_proceed_with_invite_if_ready(LinphoneCore *lc, LinphoneCall *c
 
 int linphone_core_restart_invite(LinphoneCore *lc, LinphoneCall *call){
 	linphone_call_create_op(call);
+	linphone_call_stop_media_streams(call);
+	ms_media_stream_sessions_uninit(&call->sessions[0]);
+	ms_media_stream_sessions_uninit(&call->sessions[1]);
+	linphone_call_init_media_streams(call);
 	return linphone_core_start_invite(lc,call, NULL);
 }
 
@@ -2615,7 +2623,6 @@ int linphone_core_start_invite(LinphoneCore *lc, LinphoneCall *call, const Linph
 	linphone_call_set_contact_op(call);
 
 	linphone_core_stop_dtmf_stream(lc);
-	linphone_call_init_media_streams(call);
 	linphone_call_make_local_media_description(lc,call);
 
 	if (lc->ringstream==NULL) {
@@ -2826,9 +2833,9 @@ LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const
 	/* this call becomes now the current one*/
 	lc->current_call=call;
 	linphone_call_set_state (call,LinphoneCallOutgoingInit,"Starting outgoing call");
+	linphone_call_init_media_streams(call);
 	if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
 		/* Defer the start of the call after the ICE gathering process. */
-		linphone_call_init_media_streams(call);
 		linphone_call_start_media_streams_for_ice_gathering(call);
 		call->log->start_date_time=time(NULL);
 		if (linphone_core_gather_ice_candidates(lc,call)<0) {
@@ -2841,7 +2848,6 @@ LinphoneCall * linphone_core_invite_address_with_params(LinphoneCore *lc, const
 	}
 	else if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseUpnp) {
 #ifdef BUILD_UPNP
-		linphone_call_init_media_streams(call);
 		call->log->start_date_time=time(NULL);
 		if (linphone_core_update_upnp(lc,call)<0) {
 			/* uPnP port mappings failed, proceed with the call anyway. */
@@ -3280,6 +3286,7 @@ int linphone_core_accept_call_update(LinphoneCore *lc, LinphoneCall *call, const
 	}
 	return _linphone_core_accept_call_update(lc, call, params);
 }
+
 int _linphone_core_accept_call_update(LinphoneCore *lc, LinphoneCall *call, const LinphoneCallParams *params){
 	SalMediaDescription *remote_desc;
 	bool_t keep_sdp_version;
@@ -3310,13 +3317,13 @@ int _linphone_core_accept_call_update(LinphoneCore *lc, LinphoneCall *call, cons
 		call->params.has_video = FALSE;
 	}
 	call->params.has_video &= linphone_core_media_description_contains_video_stream(remote_desc);
+	linphone_call_init_media_streams(call); /*so that video stream is initialized if necessary*/
 	linphone_call_make_local_media_description(lc,call);
 	if (call->ice_session != NULL) {
 		linphone_core_update_ice_from_remote_media_description(call, remote_desc);
 #ifdef VIDEO_ENABLED
-		if ((call->ice_session != NULL) &&!ice_session_candidates_gathered(call->ice_session)) {
+		if ((call->ice_session != NULL) && !ice_session_candidates_gathered(call->ice_session)) {
 			if ((call->params.has_video) && (call->params.has_video != old_has_video)) {
-				linphone_call_init_video_stream(call);
 				video_stream_prepare_video(call->videostream);
 				if (linphone_core_gather_ice_candidates(lc,call)<0) {
 					/* Ice candidates gathering failed, proceed with the call anyway. */
@@ -3332,7 +3339,6 @@ int _linphone_core_accept_call_update(LinphoneCore *lc, LinphoneCall *call, cons
 		linphone_core_update_upnp_from_remote_media_description(call, sal_call_get_remote_media_description(call->op));
 #ifdef VIDEO_ENABLED
 		if ((call->params.has_video) && (call->params.has_video != old_has_video)) {
-			linphone_call_init_video_stream(call);
 			video_stream_prepare_video(call->videostream);
 			if (linphone_core_update_upnp(lc, call)<0) {
 				/* uPnP update failed, proceed with the call anyway. */
@@ -3435,9 +3441,6 @@ int linphone_core_accept_call_with_params(LinphoneCore *lc, LinphoneCall *call,
 		sal_op_set_sent_custom_header(call->op,params->custom_headers);
 	}
 	
-	if (call->audiostream==NULL)
-		linphone_call_init_media_streams(call);
-	
 	/*give a chance a set card prefered sampling frequency*/
 	if (call->localdesc->streams[0].max_rate>0) {
 		ms_message ("configuring prefered card sampling rate to [%i]",call->localdesc->streams[0].max_rate);
@@ -3447,7 +3450,7 @@ int linphone_core_accept_call_with_params(LinphoneCore *lc, LinphoneCall *call,
 			ms_snd_card_set_preferred_sample_rate(lc->sound_conf.capt_sndcard, call->localdesc->streams[0].max_rate);
 	}
 	
-	if (!was_ringing && call->audiostream->ms.ticker==NULL){
+	if (!was_ringing && call->audiostream->ms.state==MSStreamInitialized){
 		audio_stream_prepare_sound(call->audiostream,lc->sound_conf.play_sndcard,lc->sound_conf.capt_sndcard);
 	}
 
@@ -5366,7 +5369,7 @@ void linphone_core_set_play_file(LinphoneCore *lc, const char *file){
 	}
 	if (file!=NULL) {
 		lc->play_file=ms_strdup(file);
-		if (call && call->audiostream && call->audiostream->ms.ticker)
+		if (call && call->audiostream && call->audiostream->ms.state==MSStreamStarted)
 			audio_stream_play(call->audiostream,file);
 	}
 }
diff --git a/coreapi/misc.c b/coreapi/misc.c
index 4c9834053..37198490e 100644
--- a/coreapi/misc.c
+++ b/coreapi/misc.c
@@ -375,6 +375,10 @@ int linphone_core_run_stun_tests(LinphoneCore *lc, LinphoneCall *call){
 		ms_warning("stun support is not implemented for ipv6");
 		return -1;
 	}
+	if (call->media_ports[0].rtp_port==-1){
+		ms_warning("Stun-only support not available for system random port");
+		return -1;
+	}
 	if (server!=NULL){
 		const struct addrinfo *ai=linphone_core_get_stun_server_addrinfo(lc);
 		ortp_socket_t sock1=-1, sock2=-1;
@@ -394,10 +398,10 @@ int linphone_core_run_stun_tests(LinphoneCore *lc, LinphoneCall *call){
 			lc->vtable.display_status(lc,_("Stun lookup in progress..."));
 
 		/*create the two audio and video RTP sockets, and send STUN message to our stun server */
-		sock1=create_socket(call->audio_port);
+		sock1=create_socket(call->media_ports[0].rtp_port);
 		if (sock1==-1) return -1;
 		if (video_enabled){
-			sock2=create_socket(call->video_port);
+			sock2=create_socket(call->media_ports[1].rtp_port);
 			if (sock2==-1) return -1;
 		}
 		got_audio=FALSE;
@@ -581,14 +585,14 @@ int linphone_core_gather_ice_candidates(LinphoneCore *lc, LinphoneCall *call)
 		return -1;
 	}
 	if ((ice_check_list_state(audio_check_list) != ICL_Completed) && (ice_check_list_candidates_gathered(audio_check_list) == FALSE)) {
-		ice_add_local_candidate(audio_check_list, "host", local_addr, call->audio_port, 1, NULL);
-		ice_add_local_candidate(audio_check_list, "host", local_addr, call->audio_port + 1, 2, NULL);
+		ice_add_local_candidate(audio_check_list, "host", local_addr, call->media_ports[0].rtp_port, 1, NULL);
+		ice_add_local_candidate(audio_check_list, "host", local_addr, call->media_ports[0].rtcp_port, 2, NULL);
 		call->stats[LINPHONE_CALL_STATS_AUDIO].ice_state = LinphoneIceStateInProgress;
 	}
-	if (call->params.has_video && (video_check_list != NULL)
+	if (linphone_core_video_enabled(lc) && (video_check_list != NULL)
 		&& (ice_check_list_state(video_check_list) != ICL_Completed) && (ice_check_list_candidates_gathered(video_check_list) == FALSE)) {
-		ice_add_local_candidate(video_check_list, "host", local_addr, call->video_port, 1, NULL);
-		ice_add_local_candidate(video_check_list, "host", local_addr, call->video_port + 1, 2, NULL);
+		ice_add_local_candidate(video_check_list, "host", local_addr, call->media_ports[1].rtp_port, 1, NULL);
+		ice_add_local_candidate(video_check_list, "host", local_addr, call->media_ports[1].rtcp_port, 2, NULL);
 		call->stats[LINPHONE_CALL_STATS_VIDEO].ice_state = LinphoneIceStateInProgress;
 	}
 
@@ -883,7 +887,12 @@ void linphone_core_update_ice_from_remote_media_description(LinphoneCall *call,
 			}
 		}
 		for (i = ice_session_nb_check_lists(call->ice_session); i > md->n_active_streams; i--) {
-			ice_session_remove_check_list(call->ice_session, ice_session_check_list(call->ice_session, i - 1));
+			IceCheckList *removed=ice_session_check_list(call->ice_session, i - 1);
+			ice_session_remove_check_list(call->ice_session, removed);
+			if (call->audiostream && call->audiostream->ms.ice_check_list==removed)
+				call->audiostream->ms.ice_check_list=NULL;
+			if (call->videostream && call->videostream->ms.ice_check_list==removed)
+				call->videostream->ms.ice_check_list=NULL;
 		}
 		ice_session_check_mismatch(call->ice_session);
 	} else {
diff --git a/coreapi/private.h b/coreapi/private.h
index 6f421e95e..9fa89dd3a 100644
--- a/coreapi/private.h
+++ b/coreapi/private.h
@@ -155,6 +155,11 @@ typedef struct StunCandidate{
 }StunCandidate;
 
 
+typedef struct _PortConfig{
+	int rtp_port;
+	int rtcp_port;
+}PortConfig;
+
 struct _LinphoneCall
 {
 	int magic; /*used to distinguish from proxy config*/
@@ -178,8 +183,8 @@ struct _LinphoneCall
 	LinphoneProxyConfig *dest_proxy;
 	int refcnt;
 	void * user_pointer;
-	int audio_port;
-	int video_port;
+	PortConfig media_ports[2];
+	MSMediaStreamSessions sessions[2]; /*the rtp, srtp, zrtp contexts for each stream*/
 	StunCandidate ac,vc; /*audio video ip/port discovered by STUN*/
 	struct _AudioStream *audiostream;  /**/
 	struct _VideoStream *videostream;
diff --git a/mediastreamer2 b/mediastreamer2
index 7111661bd..da5d1de60 160000
--- a/mediastreamer2
+++ b/mediastreamer2
@@ -1 +1 @@
-Subproject commit 7111661bd810352ed7a4ecc9c30c7754d3006a92
+Subproject commit da5d1de606699162f48751f006d9219321069545
diff --git a/oRTP b/oRTP
index 393857c0e..5008a70e0 160000
--- a/oRTP
+++ b/oRTP
@@ -1 +1 @@
-Subproject commit 393857c0e8e5cab10a8d647cc89c8390355745a7
+Subproject commit 5008a70e077b3706de4a48374f162f93071b4d08
diff --git a/tester/call_tester.c b/tester/call_tester.c
index 419def8cc..be60fd8af 100644
--- a/tester/call_tester.c
+++ b/tester/call_tester.c
@@ -519,7 +519,7 @@ static bool_t check_ice(LinphoneCoreManager* caller, LinphoneCoreManager* callee
 	return success;
 }
 
-static void call_with_ice(void) {
+static void _call_with_ice(bool_t random_ports) {
 	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
 	LinphoneCoreManager* pauline = linphone_core_manager_new( "pauline_rc");
 	
@@ -527,6 +527,13 @@ static void call_with_ice(void) {
 	linphone_core_set_stun_server(marie->lc,"stun.linphone.org");
 	linphone_core_set_firewall_policy(pauline->lc,LinphonePolicyUseIce);
 	linphone_core_set_stun_server(pauline->lc,"stun.linphone.org");
+	
+	if (random_ports){
+		linphone_core_set_audio_port(marie->lc,-1);
+		linphone_core_set_video_port(marie->lc,-1);
+		linphone_core_set_audio_port(pauline->lc,-1);
+		linphone_core_set_video_port(pauline->lc,-1);
+	}
 
 	CU_ASSERT_TRUE(call(pauline,marie));
 
@@ -536,8 +543,6 @@ static void call_with_ice(void) {
 	CU_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&marie->stat.number_of_LinphoneCallStreamsRunning,2));
 	
 	liblinphone_tester_check_rtcp(marie,pauline);
-
-
 	/*then close the call*/
 	linphone_core_terminate_all_calls(pauline->lc);
 	CU_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallEnd,1));
@@ -547,6 +552,14 @@ static void call_with_ice(void) {
 	linphone_core_manager_destroy(pauline);
 }
 
+static void call_with_ice(void){
+	_call_with_ice(FALSE);
+}
+
+static void call_with_ice_random_ports(void){
+	_call_with_ice(TRUE);
+}
+
 static void call_with_custom_headers(void) {
 	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
 	LinphoneCoreManager* pauline = linphone_core_manager_new( "pauline_rc");
@@ -718,6 +731,7 @@ static bool_t add_video(LinphoneCoreManager* caller,LinphoneCoreManager* callee)
 		return wait_for(caller->lc,callee->lc,&callee->stat.number_of_IframeDecoded,initial_callee_stat.number_of_IframeDecoded+1);
 	} else return 0;
 }
+
 static void call_with_video_added(void) {
 	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
 	LinphoneCoreManager* pauline = linphone_core_manager_new( "pauline_rc");
@@ -733,7 +747,26 @@ static void call_with_video_added(void) {
 	linphone_core_manager_destroy(pauline);
 }
 
+static void call_with_video_added_random_ports(void) {
+	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
+	LinphoneCoreManager* pauline = linphone_core_manager_new( "pauline_rc");
+	
+	linphone_core_set_audio_port(marie->lc,-1);
+	linphone_core_set_video_port(marie->lc,-1);
+	linphone_core_set_audio_port(pauline->lc,-1);
+	linphone_core_set_video_port(pauline->lc,-1);
+	
+	CU_ASSERT_TRUE(call(pauline,marie));
 
+	CU_ASSERT_TRUE(add_video(pauline,marie));
+	/*just to sleep*/
+	linphone_core_terminate_all_calls(pauline->lc);
+	CU_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallEnd,1));
+	CU_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&marie->stat.number_of_LinphoneCallEnd,1));
+
+	linphone_core_manager_destroy(marie);
+	linphone_core_manager_destroy(pauline);
+}
 
 static void call_with_declined_video(void) {
 	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
@@ -808,19 +841,26 @@ static void video_call(void) {
 }
 #endif /*VIDEO_ENABLED*/
 
-static void call_with_media_relay(void) {
+static void _call_with_media_relay(bool_t random_ports) {
 	LinphoneCoreManager* marie = linphone_core_manager_new( "marie_rc");
 	LinphoneCoreManager* pauline = linphone_core_manager_new( "pauline_rc");
 	linphone_core_set_user_agent(marie->lc,"Natted Linphone",NULL);
 	linphone_core_set_user_agent(pauline->lc,"Natted Linphone",NULL);
+	
+	if (random_ports){
+		linphone_core_set_audio_port(marie->lc,-1);
+		linphone_core_set_video_port(marie->lc,-1);
+		linphone_core_set_audio_port(pauline->lc,-1);
+		linphone_core_set_video_port(pauline->lc,-1);
+	}
+	
 	CU_ASSERT_TRUE(call(pauline,marie));
 	liblinphone_tester_check_rtcp(pauline,marie);
-
+	
 #ifdef VIDEO_ENABLED
 	CU_ASSERT_TRUE(add_video(pauline,marie));
 	liblinphone_tester_check_rtcp(pauline,marie);
 #endif
-
 	/*just to sleep*/
 	linphone_core_terminate_all_calls(pauline->lc);
 	CU_ASSERT_TRUE(wait_for(pauline->lc,marie->lc,&pauline->stat.number_of_LinphoneCallEnd,1));
@@ -828,7 +868,14 @@ static void call_with_media_relay(void) {
 
 	linphone_core_manager_destroy(marie);
 	linphone_core_manager_destroy(pauline);
+}
 
+static void call_with_media_relay(void) {
+	_call_with_media_relay(FALSE);
+}
+
+static void call_with_media_relay_random_ports(void) {
+	_call_with_media_relay(TRUE);
 }
 
 static void call_with_privacy(void) {
@@ -1839,6 +1886,7 @@ test_t call_tests[] = {
 	{ "Call failed because of codecs", call_failed_because_of_codecs },
 	{ "Simple call", simple_call },
 	{ "Call with media relay", call_with_media_relay},
+	{ "Call with media relay (random ports)", call_with_media_relay_random_ports},
 	{ "Simple call compatibility mode", simple_call_compatibility_mode },
 	{ "Early-media call", early_media_call },
 	{ "Early-media call with ringing", early_media_call_with_ringing },
@@ -1855,6 +1903,7 @@ test_t call_tests[] = {
 	{ "Simple video call",video_call},
 	{ "SRTP ice video call", srtp_video_ice_call },
 	{ "Call with video added", call_with_video_added },
+	{ "Call with video added (random ports)", call_with_video_added_random_ports },
 	{ "Call with video declined",call_with_declined_video},
 #else
 	{ "SRTP ice call", srtp_ice_call },
@@ -1872,6 +1921,7 @@ test_t call_tests[] = {
 	{ "Unattended call transfer with error", unattended_call_transfer_with_error },
 	{ "Call transfer existing call outgoing call", call_transfer_existing_call_outgoing_call },
 	{ "Call with ICE", call_with_ice },
+	{ "Call with ICE (random ports)", call_with_ice_random_ports },
 	{ "Call with custom headers",call_with_custom_headers},
 	{ "Call established with rejected INFO",call_established_with_rejected_info},
 	{ "Call established with rejected RE-INVITE",call_established_with_rejected_reinvite},
diff --git a/tester/stun_tester.c b/tester/stun_tester.c
index e416fa02c..d19650f06 100644
--- a/tester/stun_tester.c
+++ b/tester/stun_tester.c
@@ -73,8 +73,8 @@ static void linphone_stun_test_grab_ip()
 	int tmp=0;
 
 	memset(&dummy_call, 0, sizeof(LinphoneCall));
-	dummy_call.audio_port = 7078;
-	dummy_call.audio_port = 9078;
+	dummy_call.media_ports[0].rtp_port = 7078;
+	dummy_call.media_ports[1].rtp_port = 9078;
 
 	linphone_core_set_stun_server(lc_stun->lc, stun_address);
 	CU_ASSERT_STRING_EQUAL(stun_address, linphone_core_get_stun_server(lc_stun->lc));