X-Git-Url: http://sjero.net/git/?a=blobdiff_plain;f=coreapi%2Flinphonecall.c;h=db3190bbb3e851709690144472689ea3cc3e6ad5;hb=0fa5d7c73b672772a8841c65ff0c7fcdc4e1fce7;hp=d43c457b67de9193b6aa9d300caa32cafd87ecd5;hpb=2043ec6a90a4f2e7f077391909c968b4f7c98e9e;p=linphone

diff --git a/coreapi/linphonecall.c b/coreapi/linphonecall.c
index d43c457b..db3190bb 100644
--- a/coreapi/linphonecall.c
+++ b/coreapi/linphonecall.c
@@ -27,7 +27,7 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA  02111-1307, USA.
 #include "private.h"
 #include <ortp/event.h>
 #include <ortp/b64.h>
-
+#include <math.h>
 
 #include "mediastreamer2/mediastream.h"
 #include "mediastreamer2/msvolume.h"
@@ -161,21 +161,22 @@ 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->ortpZrtpContext==NULL){
+	if (call->audiostream->ms.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->ortpZrtpContext);
+		ortp_zrtp_sas_verified(call->audiostream->ms.zrtp_context);
 	}else if (call->auth_token_verified && !verified){
-		ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
+		ortp_zrtp_sas_reset_verified(call->audiostream->ms.zrtp_context);
 	}
 	call->auth_token_verified=verified;
 	propagate_encryption_changed(call);
 }
 
-static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
+static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate, int nb_codecs_limit){
 	MSList *l=NULL;
 	const MSList *it;
+	int nb = 0;
 	if (max_sample_rate) *max_sample_rate=0;
 	for(it=codecs;it!=NULL;it=it->next){
 		PayloadType *pt=(PayloadType*)it->data;
@@ -186,45 +187,49 @@ static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandw
 			}
 			if (linphone_core_check_payload_type_usability(lc,pt)){
 				l=ms_list_append(l,payload_type_clone(pt));
+				nb++;
 				if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
 			}
 		}
+		if ((nb_codecs_limit > 0) && (nb >= nb_codecs_limit)) break;
 	}
 	return l;
 }
 
 static void update_media_description_from_stun(SalMediaDescription *md, const StunCandidate *ac, const StunCandidate *vc){
-	if (ac->port!=0){
-		strcpy(md->streams[0].rtp_addr,ac->addr);
-		md->streams[0].rtp_port=ac->port;
-		if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->nstreams==1){
-			strcpy(md->addr,ac->addr);
+	int i;
+	for (i = 0; i < md->n_active_streams; i++) {
+		if ((md->streams[i].type == SalAudio) && (ac->port != 0)) {
+			strcpy(md->streams[0].rtp_addr,ac->addr);
+			md->streams[0].rtp_port=ac->port;
+			if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->n_active_streams==1){
+				strcpy(md->addr,ac->addr);
+			}
+		}
+		if ((md->streams[i].type == SalVideo) && (vc->port != 0)) {
+			strcpy(md->streams[1].rtp_addr,vc->addr);
+			md->streams[1].rtp_port=vc->port;
 		}
 	}
-	if (vc->port!=0){
-		strcpy(md->streams[1].rtp_addr,vc->addr);
-		md->streams[1].rtp_port=vc->port;
-	}
-	
 }
 
-
-static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
+void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *call){
 	MSList *l;
 	PayloadType *pt;
+	SalMediaDescription *old_md=call->localdesc;
 	int i;
 	const char *me=linphone_core_get_identity(lc);
 	LinphoneAddress *addr=linphone_address_new(me);
 	const char *username=linphone_address_get_username (addr);
 	SalMediaDescription *md=sal_media_description_new();
-	
-	if (call->ping_time>0) {
-		linphone_core_adapt_to_network(lc,call->ping_time,&call->params);
-	}
+	bool_t keep_srtp_keys=lp_config_get_int(lc->config,"sip","keep_srtp_keys",0);
 
-	md->session_id=session_id;
-	md->session_ver=session_ver;
-	md->nstreams=1;
+	linphone_core_adapt_to_network(lc,call->ping_time,&call->params);
+
+	md->session_id=(old_md ? old_md->session_id : (rand() & 0xfff));
+	md->session_ver=(old_md ? (old_md->session_ver+1) : (rand() & 0xfff));
+	md->n_total_streams=(old_md ? old_md->n_total_streams : 1);
+	md->n_active_streams=1;
 	strncpy(md->addr,call->localip,sizeof(md->addr));
 	strncpy(md->username,username,sizeof(md->username));
 	
@@ -244,70 +249,99 @@ static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, Li
 		md->streams[0].ptime=call->params.down_ptime;
 	else
 		md->streams[0].ptime=linphone_core_get_download_ptime(lc);
-	l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
-	pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
+	l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate,-1);
+	pt=payload_type_clone(rtp_profile_get_payload_from_mime(lc->default_profile,"telephone-event"));
 	l=ms_list_append(l,pt);
 	md->streams[0].payloads=l;
-	
-
 
 	if (call->params.has_video){
-		md->nstreams++;
+		md->n_active_streams++;
 		md->streams[1].rtp_port=call->video_port;
 		md->streams[1].rtcp_port=call->video_port+1;
 		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);
+		l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL,-1);
 		md->streams[1].payloads=l;
 	}
-	
-	for(i=0; i<md->nstreams; i++) {
+	if (md->n_total_streams < md->n_active_streams)
+		md->n_total_streams = md->n_active_streams;
+
+	/* Deactivate inactive streams. */
+	for (i = md->n_active_streams; i < md->n_total_streams; i++) {
+		md->streams[i].rtp_port = 0;
+		md->streams[i].rtcp_port = 0;
+		md->streams[i].proto = SalProtoRtpAvp;
+		md->streams[i].type = old_md->streams[i].type;
+		md->streams[i].dir = SalStreamInactive;
+		l = make_codec_list(lc, lc->codecs_conf.video_codecs, 0, NULL, 1);
+		md->streams[i].payloads = l;
+	}
+
+	for(i=0; i<md->n_active_streams; i++) {
 		if (md->streams[i].proto == SalProtoRtpSavp) {
-			md->streams[i].crypto[0].tag = 1;
-			md->streams[i].crypto[0].algo = AES_128_SHA1_80;
-			if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
-				md->streams[i].crypto[0].algo = 0;
-			md->streams[i].crypto[1].tag = 2;
-			md->streams[i].crypto[1].algo = AES_128_SHA1_32;
-			if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
-				md->streams[i].crypto[1].algo = 0;
-			md->streams[i].crypto[2].algo = 0;
+			if (keep_srtp_keys && old_md && old_md->streams[i].proto==SalProtoRtpSavp){
+				int j;
+				for(j=0;j<SAL_CRYPTO_ALGO_MAX;++j){
+					memcpy(&md->streams[i].crypto[j],&old_md->streams[i].crypto[j],sizeof(SalSrtpCryptoAlgo));
+				}
+			}else{
+				md->streams[i].crypto[0].tag = 1;
+				md->streams[i].crypto[0].algo = AES_128_SHA1_80;
+				if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
+					md->streams[i].crypto[0].algo = 0;
+				md->streams[i].crypto[1].tag = 2;
+				md->streams[i].crypto[1].algo = AES_128_SHA1_32;
+				if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
+					md->streams[i].crypto[1].algo = 0;
+				md->streams[i].crypto[2].algo = 0;
+			}
 		}
 	}
 	update_media_description_from_stun(md,&call->ac,&call->vc);
 	if (call->ice_session != NULL) {
 		linphone_core_update_local_media_description_from_ice(md, call->ice_session);
+		linphone_core_update_ice_state_in_call_stats(call);
 	}
-	linphone_address_destroy(addr);
-	return md;
-}
-
-void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
-	SalMediaDescription *md=call->localdesc;
-	if (md== NULL) {
-		call->localdesc = create_local_media_description(lc,call);
-	} else {
-		call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
-		sal_media_description_unref(md);
+#ifdef BUILD_UPNP
+	if(call->upnp_session != NULL) {
+		linphone_core_update_local_media_description_from_upnp(md, call->upnp_session);
+		linphone_core_update_upnp_state_in_call_stats(call);
 	}
+#endif  //BUILD_UPNP
+	linphone_address_destroy(addr);
+	call->localdesc=md;
+	if (old_md) sal_media_description_unref(old_md);
 }
 
-SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
-	unsigned int id=rand() & 0xfff;
-	return _create_local_media_description(lc,call,id,id);
-}
-
-static int find_port_offset(LinphoneCore *lc){
+static int find_port_offset(LinphoneCore *lc, SalStreamType type){
 	int offset;
 	MSList *elem;
-	int audio_port;
+	int tried_port;
+	int existing_port;
 	bool_t already_used=FALSE;
 	for(offset=0;offset<100;offset+=2){
-		audio_port=linphone_core_get_audio_port (lc)+offset;
+		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;
+		}
 		already_used=FALSE;
 		for(elem=lc->calls;elem!=NULL;elem=elem->next){
 			LinphoneCall *call=(LinphoneCall*)elem->data;
-			if (call->audio_port==audio_port) {
+			switch (type) {
+				default:
+				case SalAudio:
+					existing_port = call->audio_port;
+					break;
+				case SalVideo:
+					existing_port = call->video_port;
+					break;
+			}
+			if (existing_port==tried_port) {
 				already_used=TRUE;
 				break;
 			}
@@ -321,8 +355,54 @@ static int find_port_offset(LinphoneCore *lc){
 	return offset;
 }
 
+static int select_random_port(LinphoneCore *lc, SalStreamType type) {
+	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;
+			}
+			if (existing_port == tried_port) {
+				already_used = TRUE;
+				break;
+			}
+		}
+		if (!already_used) break;
+	}
+	if (nb_tries == 100) {
+		ms_error("Could not find any free port!");
+		return -1;
+	}
+	return tried_port;
+}
+
 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
 	int port_offset;
+	int min_port, max_port;
 	call->magic=linphone_call_magic;
 	call->refcnt=1;
 	call->state=LinphoneCallIdle;
@@ -332,10 +412,26 @@ static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from,
 	call->log=linphone_call_log_new(call, from, to);
 	call->owns_call_log=TRUE;
 	linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
-	port_offset=find_port_offset (call->core);
-	if (port_offset==-1) return;
-	call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
-	call->video_port=linphone_core_get_video_port(call->core)+port_offset;
+	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);
+	}
+	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);
+	}
 	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);
 }
@@ -345,6 +441,11 @@ void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
 	stats->received_rtcp = NULL;
 	stats->sent_rtcp = NULL;
 	stats->ice_state = LinphoneIceStateNotActivated;
+#ifdef BUILD_UPNP
+	stats->upnp_state = LinphoneUpnpStateIdle;
+#else
+	stats->upnp_state = LinphoneUpnpStateNotAvailable;
+#endif //BUILD_UPNP
 }
 
 
@@ -368,9 +469,12 @@ LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddr
 	call->op=sal_op_new(lc->sal);
 	sal_op_set_user_pointer(call->op,call);
 	call->core=lc;
-	linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
+	linphone_core_get_local_ip(lc,NULL,call->localip);
 	linphone_call_init_common(call,from,to);
-	call->params=*params;
+	_linphone_call_params_copy(&call->params,params);
+	sal_op_set_custom_header(call->op,call->params.custom_headers);
+	call->params.custom_headers=NULL;
+	
 	if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
 		call->ice_session = ice_session_new();
 		ice_session_set_role(call->ice_session, IR_Controlling);
@@ -378,6 +482,13 @@ LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddr
 	if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
 		call->ping_time=linphone_core_run_stun_tests(call->core,call);
 	}
+#ifdef BUILD_UPNP
+	if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseUpnp) {
+		if(!lc->rtp_conf.disable_upnp) {
+			call->upnp_session = linphone_upnp_session_new(call);
+		}
+	}
+#endif //BUILD_UPNP
 	call->camera_active=params->has_video;
 	
 	discover_mtu(lc,linphone_address_get_domain (to));
@@ -391,6 +502,7 @@ LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddr
 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
 	LinphoneCall *call=ms_new0(LinphoneCall,1);
 	char *from_str;
+	const SalMediaDescription *md;
 
 	call->dir=LinphoneCallIncoming;
 	sal_op_set_user_pointer(op,call);
@@ -398,21 +510,35 @@ LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *fro
 	call->core=lc;
 
 	if (lc->sip_conf.ping_with_options){
-		/*the following sends an option request back to the caller so that
-		 we get a chance to discover our nat'd address before answering.*/
-		call->ping_op=sal_op_new(lc->sal);
-		from_str=linphone_address_as_string_uri_only(from);
-		sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
-		sal_op_set_user_pointer(call->ping_op,call);
-		sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
-		ms_free(from_str);
+#ifdef BUILD_UPNP
+		if (lc->upnp != NULL && linphone_core_get_firewall_policy(lc)==LinphonePolicyUseUpnp &&
+			linphone_upnp_context_get_state(lc->upnp) == LinphoneUpnpStateOk) {
+#else //BUILD_UPNP
+		{
+#endif //BUILD_UPNP
+			/*the following sends an option request back to the caller so that
+			 we get a chance to discover our nat'd address before answering.*/
+			call->ping_op=sal_op_new(lc->sal);
+			from_str=linphone_address_as_string_uri_only(from);
+			sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
+			sal_op_set_user_pointer(call->ping_op,call);
+			sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
+			ms_free(from_str);
+		}
 	}
 
 	linphone_address_clean(from);
-	linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
+	linphone_core_get_local_ip(lc,NULL,call->localip);
 	linphone_call_init_common(call, from, to);
+	call->log->call_id=ms_strdup(sal_op_get_call_id(op)); /*must be known at that time*/
 	linphone_core_init_default_params(lc, &call->params);
+	md=sal_call_get_remote_media_description(op);
 	call->params.has_video &= !!lc->video_policy.automatically_accept;
+	if (md) {
+		// It is licit to receive an INVITE without SDP
+		// In this case WE chose the media parameters according to policy.
+		call->params.has_video &= linphone_core_media_description_contains_video_stream(md);
+	}
 	switch (linphone_core_get_firewall_policy(call->core)) {
 		case LinphonePolicyUseIce:
 			call->ice_session = ice_session_new();
@@ -424,13 +550,28 @@ LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *fro
 				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(call);
+					linphone_call_stop_media_streams_for_ice_gathering(call);
 				}
 			}
 			break;
 		case LinphonePolicyUseStun:
 			call->ping_time=linphone_core_run_stun_tests(call->core,call);
 			/* No break to also destroy ice session in this case. */
+			break;
+		case LinphonePolicyUseUpnp:
+#ifdef BUILD_UPNP
+			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);
+					}
+				}
+			}
+#endif //BUILD_UPNP
+			break;
 		default:
 			break;
 	}
@@ -519,8 +660,8 @@ const char *linphone_call_state_to_string(LinphoneCallState cs){
 			return "LinphoneCallUpdatedByRemote";
 		case LinphoneCallIncomingEarlyMedia:
 			return "LinphoneCallIncomingEarlyMedia";
-		case LinphoneCallUpdated:
-			return "LinphoneCallUpdated";
+		case LinphoneCallUpdating:
+			return "LinphoneCallUpdating";
 		case LinphoneCallReleased:
 			return "LinphoneCallReleased";
 	}
@@ -579,6 +720,10 @@ void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const
 
 static void linphone_call_destroy(LinphoneCall *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;
@@ -602,10 +747,7 @@ static void linphone_call_destroy(LinphoneCall *obj)
 	if (obj->auth_token) {
 		ms_free(obj->auth_token);
 	}
-	if (obj->ice_session) {
-		ice_session_destroy(obj->ice_session);
-	}
-
+	linphone_call_params_uninit(&obj->params);
 	ms_free(obj);
 }
 
@@ -640,7 +782,9 @@ void linphone_call_unref(LinphoneCall *obj){
 /**
  * Returns current parameters associated to the call.
 **/
-const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
+const LinphoneCallParams * linphone_call_get_current_params(LinphoneCall *call){
+	if (call->params.record_file)
+		call->current_params.record_file=call->params.record_file;
 	return &call->current_params;
 }
 
@@ -673,6 +817,12 @@ const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
 			}else if (vsd){
 				cp->has_video=is_video_active(vsd);
 			}
+			if (!cp->has_video){
+				if (md->bandwidth>0 && md->bandwidth<=linphone_core_get_edge_bw(call->core)){
+					cp->low_bandwidth=TRUE;
+				}
+			}
+			cp->custom_headers=(SalCustomHeader*)sal_op_get_custom_header(call->op);
 			return cp;
 		}
 	}
@@ -765,6 +915,16 @@ const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
 	return NULL;
 }
 
+/**
+ * Returns the far end's sip contact as a string, if available.
+**/
+const char *linphone_call_get_remote_contact(LinphoneCall *call){
+	if (call->op){
+		return sal_op_get_remote_contact(call->op);
+	}
+	return NULL;
+}
+
 /**
  * Returns true if this calls has received a transfer that has not been
  * executed yet.
@@ -804,7 +964,7 @@ LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
 **/
 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
 #ifdef VIDEO_ENABLED
-	if (call->videostream!=NULL && call->videostream->ticker!=NULL){
+	if (call->videostream!=NULL && call->videostream->ms.ticker!=NULL){
 		LinphoneCore *lc=call->core;
 		MSWebCam *nowebcam=get_nowebcam_device();
 		if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
@@ -816,6 +976,18 @@ void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
 #endif
 }
 
+#ifdef VIDEO_ENABLED
+/**
+ * Request remote side to send us a Video Fast Update.
+**/
+void linphone_call_send_vfu_request(LinphoneCall *call)
+{
+	if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
+		sal_call_send_vfu_request(call->op);
+}
+#endif
+
+
 /**
  * Take a photo of currently received video and write it into a jpeg file.
 **/
@@ -844,14 +1016,46 @@ void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
 	cp->has_video=enabled;
 }
 
+/**
+ * Returns the audio codec used in the call, described as a PayloadType structure.
+**/
 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
 	return cp->audio_codec;
 }
 
+
+/**
+ * Returns the video codec used in the call, described as a PayloadType structure.
+**/
 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
 	return cp->video_codec;
 }
 
+/**
+ * @ingroup call_control
+ * Use to know if this call has been configured in low bandwidth mode.
+ * This mode can be automatically discovered thanks to a stun server when activate_edge_workarounds=1 in section [net] of configuration file.
+ * An application that would have reliable way to know network capacity may not use activate_edge_workarounds=1 but instead manually configure
+ * low bandwidth mode with linphone_call_params_enable_low_bandwidth().
+ * <br> When enabled, this param may transform a call request with video in audio only mode.
+ * @return TRUE if low bandwidth has been configured/detected
+ */
+bool_t linphone_call_params_low_bandwidth_enabled(const LinphoneCallParams *cp) {
+	return cp->low_bandwidth;
+}
+
+/**
+ * @ingroup call_control
+ * Indicate low bandwith mode. 
+ * Configuring a call to low bandwidth mode will result in the core to activate several settings for the call in order to ensure that bitrate usage
+ * is lowered to the minimum possible. Typically, ptime (packetization time) will be increased, audio codec's output bitrate will be targetted to 20kbit/s provided
+ * that it is achievable by the codec selected after SDP handshake. Video is automatically disabled.
+ * 
+**/
+void linphone_call_params_enable_low_bandwidth(LinphoneCallParams *cp, bool_t enabled){
+	cp->low_bandwidth=enabled;
+}
+
 /**
  * Returns whether video is enabled.
 **/
@@ -859,10 +1063,16 @@ bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
 	return cp->has_video;
 }
 
+/**
+ * Returns kind of media encryption selected for the call.
+**/
 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
 	return cp->media_encryption;
 }
 
+/**
+ * Set requested media encryption for a call.
+**/
 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
 	cp->media_encryption = e;
 }
@@ -875,6 +1085,9 @@ void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, boo
 	cp->real_early_media=enabled;
 }
 
+/**
+ * Indicates whether sending of early media was enabled.
+**/
 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
 	return cp->real_early_media;
 }
@@ -894,33 +1107,46 @@ void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int
 	cp->audio_bw=bandwidth;
 }
 
-#ifdef VIDEO_ENABLED
-/**
- * Request remote side to send us a Video Fast Update.
-**/
-void linphone_call_send_vfu_request(LinphoneCall *call)
-{
-	if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
-		sal_call_send_vfu_request(call->op);
+void linphone_call_params_add_custom_header(LinphoneCallParams *params, const char *header_name, const char *header_value){
+	params->custom_headers=sal_custom_header_append(params->custom_headers,header_name,header_value);
+}
+
+const char *linphone_call_params_get_custom_header(const LinphoneCallParams *params, const char *header_name){
+	return sal_custom_header_find(params->custom_headers,header_name);
+}
+
+void _linphone_call_params_copy(LinphoneCallParams *ncp, const LinphoneCallParams *cp){
+	memcpy(ncp,cp,sizeof(LinphoneCallParams));
+	if (cp->record_file) ncp->record_file=ms_strdup(cp->record_file);
+	/*
+	 * The management of the custom headers is not optimal. We copy everything while ref counting would be more efficient.
+	 */
+	if (cp->custom_headers) ncp->custom_headers=sal_custom_header_clone(cp->custom_headers);
 }
-#endif
 
 /**
- *
+ * Copy existing LinphoneCallParams to a new LinphoneCallParams object.
 **/
 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
 	LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
-	memcpy(ncp,cp,sizeof(LinphoneCallParams));
+	_linphone_call_params_copy(ncp,cp);
 	return ncp;
 }
 
+void linphone_call_params_uninit(LinphoneCallParams *p){
+	if (p->record_file) ms_free(p->record_file);
+	if (p->custom_headers) sal_custom_header_free(p->custom_headers);
+}
+
 /**
- *
+ * Destroy LinphoneCallParams.
 **/
 void linphone_call_params_destroy(LinphoneCallParams *p){
+	linphone_call_params_uninit(p);
 	ms_free(p);
 }
 
+
 /**
  * @}
 **/
@@ -942,11 +1168,11 @@ static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const u
 			ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
 			linphone_call_send_vfu_request(call);
 			break;
-        case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
-            ms_message("First video frame decoded successfully");
-            if (call->nextVideoFrameDecoded._func != NULL)
-                call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
-            break;
+		case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
+			ms_message("First video frame decoded successfully");
+			if (call->nextVideoFrameDecoded._func != NULL)
+			call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
+			break;
 		default:
 			ms_warning("Unhandled event %i", event_id);
 			break;
@@ -955,19 +1181,21 @@ static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const u
 #endif
 
 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
-    call->nextVideoFrameDecoded._func = cb;
-    call->nextVideoFrameDecoded._user_data = user_data;
+	call->nextVideoFrameDecoded._func = cb;
+	call->nextVideoFrameDecoded._user_data = user_data;
 #ifdef VIDEO_ENABLED
-    ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
+	ms_filter_call_method_noarg(call->videostream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
 #endif
 }
 
 void linphone_call_init_audio_stream(LinphoneCall *call){
 	LinphoneCore *lc=call->core;
 	AudioStream *audiostream;
-	int dscp=linphone_core_get_audio_dscp(lc);
+	int dscp;
 
+	if (call->audiostream != NULL) return;
 	call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,linphone_core_ipv6_enabled(lc));
+	dscp=linphone_core_get_audio_dscp(lc);
 	if (dscp!=-1)
 		audio_stream_set_dscp(audiostream,dscp);
 	if (linphone_core_echo_limiter_enabled(lc)){
@@ -1000,26 +1228,31 @@ void linphone_call_init_audio_stream(LinphoneCall *call){
 	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->session,artp,artcp);
+		rtp_session_set_transports(audiostream->ms.session,artp,artcp);
 	}
 	if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
-		rtp_session_set_pktinfo(audiostream->session, TRUE);
-		rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
+		rtp_session_set_pktinfo(audiostream->ms.session, TRUE);
+		rtp_session_set_symmetric_rtp(audiostream->ms.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->ice_check_list = ice_session_check_list(call->ice_session, 0);
-		ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
+		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);
 	}
 
 	call->audiostream_app_evq = ortp_ev_queue_new();
-	rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
+	rtp_session_register_event_queue(audiostream->ms.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){
 		int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
 		int dscp=linphone_core_get_video_dscp(lc);
@@ -1028,7 +1261,7 @@ void linphone_call_init_video_stream(LinphoneCall *call){
 		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->session,video_recv_buf_size);
+		if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->ms.session,video_recv_buf_size);
 
 		if( lc->video_conf.displaytype != NULL)
 			video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
@@ -1036,19 +1269,19 @@ void linphone_call_init_video_stream(LinphoneCall *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->session,vrtp,vrtcp);
+			rtp_session_set_transports(call->videostream->ms.session,vrtp,vrtcp);
 		}
 		if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
-			rtp_session_set_pktinfo(call->videostream->session, TRUE);
-			rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
+			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->ice_check_list = ice_session_check_list(call->ice_session, 1);
-			ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
+			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);
 		}
 		call->videostream_app_evq = ortp_ev_queue_new();
-		rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
+		rtp_session_register_event_queue(call->videostream->ms.session,call->videostream_app_evq);
 #ifdef TEST_EXT_RENDERER
 		video_stream_set_render_callback(call->videostream,rendercb,NULL);
 #endif
@@ -1066,8 +1299,7 @@ void linphone_call_init_media_streams(LinphoneCall *call){
 
 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
 
-static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
-	LinphoneCore* lc = (LinphoneCore*)user_data;
+static void linphone_core_dtmf_received(LinphoneCore *lc, int dtmf){
 	if (dtmf<0 || dtmf>15){
 		ms_warning("Bad dtmf value %i",dtmf);
 		return;
@@ -1099,7 +1331,7 @@ static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
 }
 
 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
-	float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
+	float mic_gain=lc->sound_conf.soft_mic_lev;
 	float thres = 0;
 	float recv_gain;
 	float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
@@ -1107,7 +1339,7 @@ void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t mute
 	int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
 
 	if (!muted)
-		audio_stream_set_mic_gain(st,mic_gain);
+		linphone_core_set_mic_gain_db (lc, mic_gain);
 	else
 		audio_stream_set_mic_gain(st,0);
 
@@ -1141,7 +1373,7 @@ void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t mute
 	}
 	if (st->volrecv){
 		/* parameters for a limited noise-gate effect, using echo limiter threshold */
-		float floorgain = 1/mic_gain;
+		float floorgain = 1/pow(10,(mic_gain)/10);
 		int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
 		ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
 		ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
@@ -1155,10 +1387,10 @@ static void post_configure_audio_streams(LinphoneCall*call){
 	LinphoneCore *lc=call->core;
 	_post_configure_audio_stream(st,lc,call->audio_muted);
 	if (lc->vtable.dtmf_received!=NULL){
-		/* replace by our default action*/
 		audio_stream_play_received_dtmfs(call->audiostream,FALSE);
-		rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
 	}
+	if (call->record_active) 
+		linphone_call_start_recording(call);
 }
 
 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
@@ -1310,6 +1542,8 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cna
 			if (captcard &&  stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
 			audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
 			audio_stream_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
+			if (!call->params.in_conference && call->params.record_file)
+				audio_stream_mixed_record_open(call->audiostream,call->params.record_file);
 			audio_stream_start_full(
 				call->audiostream,
 				call->audio_profile,
@@ -1338,23 +1572,23 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cna
 			}
 			audio_stream_set_rtcp_information(call->audiostream, cname, rtcp_tool);
 			
-            /* valid local tags are > 0 */
+			/* valid local tags are > 0 */
 			if (stream->proto == SalProtoRtpSavp) {
-                const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
-                                                                                            SalProtoRtpSavp,SalAudio);
-                int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
-                
-                if (crypto_idx >= 0) {
-                    audio_stream_enable_strp(
-                                             call->audiostream, 
-                                             stream->crypto[0].algo,
-                                             local_st_desc->crypto[crypto_idx].master_key,
-                                             stream->crypto[0].master_key);
-                    call->audiostream_encrypted=TRUE;
-                } else {
-                    ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
-                    call->audiostream_encrypted=FALSE;
-                }
+			const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
+													SalProtoRtpSavp,SalAudio);
+			int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
+
+			if (crypto_idx >= 0) {
+				audio_stream_enable_srtp(
+							call->audiostream, 
+							stream->crypto[0].algo,
+							local_st_desc->crypto[crypto_idx].master_key,
+							stream->crypto[0].master_key);
+				call->audiostream_encrypted=TRUE;
+			} else {
+				ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
+				call->audiostream_encrypted=FALSE;
+			}
 			}else call->audiostream_encrypted=FALSE;
 			if (call->params.in_conference){
 				/*transform the graph to connect it to the conference filter */
@@ -1362,6 +1596,7 @@ static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cna
 				linphone_call_add_to_conf(call, mute);
 			}
 			call->current_params.in_conference=call->params.in_conference;
+			call->current_params.low_bandwidth=call->params.low_bandwidth;
 		}else ms_warning("No audio stream accepted ?");
 	}
 }
@@ -1431,7 +1666,7 @@ static void linphone_call_start_video_stream(LinphoneCall *call, const char *cna
 				cam=get_nowebcam_device();
 			}
 			if (!is_inactive){
-                call->log->video_enabled = TRUE;
+				call->log->video_enabled = TRUE;
 				video_stream_set_direction (call->videostream, dir);
 				ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
 				video_stream_set_device_rotation(call->videostream, lc->device_rotation);
@@ -1535,26 +1770,105 @@ void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
 #endif
 }
 
+void linphone_call_stop_media_streams_for_ice_gathering(LinphoneCall *call){
+	audio_stream_unprepare_sound(call->audiostream);
+#ifdef VIDEO_ENABLED
+	if (call->videostream) {
+		video_stream_unprepare_video(call->videostream);
+	}
+#endif
+}
+
+void linphone_call_update_crypto_parameters(LinphoneCall *call, SalMediaDescription *old_md, SalMediaDescription *new_md) {
+	SalStreamDescription *old_stream;
+	SalStreamDescription *new_stream;
+	int i;
+
+	old_stream = sal_media_description_find_stream(old_md, SalProtoRtpSavp, SalAudio);
+	new_stream = sal_media_description_find_stream(new_md, SalProtoRtpSavp, SalAudio);
+	if (old_stream && new_stream) {
+		const SalStreamDescription *local_st_desc = sal_media_description_find_stream(call->localdesc, SalProtoRtpSavp, SalAudio);
+		if (local_st_desc) {
+			int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, new_stream->crypto_local_tag);
+			if (crypto_idx >= 0) {
+				audio_stream_enable_srtp(call->audiostream, new_stream->crypto[0].algo, local_st_desc->crypto[crypto_idx].master_key, new_stream->crypto[0].master_key);
+				call->audiostream_encrypted = TRUE;
+			} else {
+				ms_warning("Failed to find local crypto algo with tag: %d", new_stream->crypto_local_tag);
+				call->audiostream_encrypted = FALSE;
+			}
+			for (i = 0; i < SAL_CRYPTO_ALGO_MAX; i++) {
+				old_stream->crypto[i].tag = new_stream->crypto[i].tag;
+				old_stream->crypto[i].algo = new_stream->crypto[i].algo;
+				strncpy(old_stream->crypto[i].master_key, new_stream->crypto[i].master_key, sizeof(old_stream->crypto[i].master_key) - 1);
+			}
+		}
+	}
+
+#ifdef VIDEO_ENABLED
+	old_stream = sal_media_description_find_stream(old_md, SalProtoRtpSavp, SalVideo);
+	new_stream = sal_media_description_find_stream(new_md, SalProtoRtpSavp, SalVideo);
+	if (old_stream && new_stream) {
+		const SalStreamDescription *local_st_desc = sal_media_description_find_stream(call->localdesc, SalProtoRtpSavp, SalVideo);
+		if (local_st_desc) {
+			int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, new_stream->crypto_local_tag);
+			if (crypto_idx >= 0) {
+				video_stream_enable_strp(call->videostream, new_stream->crypto[0].algo, local_st_desc->crypto[crypto_idx].master_key, new_stream->crypto[0].master_key);
+				call->videostream_encrypted = TRUE;
+			} else {
+				ms_warning("Failed to find local crypto algo with tag: %d", new_stream->crypto_local_tag);
+				call->videostream_encrypted = FALSE;
+			}
+			for (i = 0; i < SAL_CRYPTO_ALGO_MAX; i++) {
+				old_stream->crypto[i].tag = new_stream->crypto[i].tag;
+				old_stream->crypto[i].algo = new_stream->crypto[i].algo;
+				strncpy(old_stream->crypto[i].master_key, new_stream->crypto[i].master_key, sizeof(old_stream->crypto[i].master_key) - 1);
+			}
+		}
+	}
+#endif
+}
+
+void linphone_call_update_remote_session_id_and_ver(LinphoneCall *call) {
+	SalMediaDescription *remote_desc = sal_call_get_remote_media_description(call->op);
+	if (remote_desc) {
+		call->remote_session_id = remote_desc->session_id;
+		call->remote_session_ver = remote_desc->session_ver;
+	}
+}
+
 void linphone_call_delete_ice_session(LinphoneCall *call){
 	if (call->ice_session != NULL) {
 		ice_session_destroy(call->ice_session);
 		call->ice_session = NULL;
-		if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
-		if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
+		if (call->audiostream != NULL) call->audiostream->ms.ice_check_list = NULL;
+		if (call->videostream != NULL) call->videostream->ms.ice_check_list = NULL;
 		call->stats[LINPHONE_CALL_STATS_AUDIO].ice_state = LinphoneIceStateNotActivated;
 		call->stats[LINPHONE_CALL_STATS_VIDEO].ice_state = LinphoneIceStateNotActivated;
 	}
 }
 
-static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
-	audio_stream_get_local_rtp_stats (st,&log->local_stats);
-	log->quality=audio_stream_get_average_quality_rating(st);
+#ifdef BUILD_UPNP
+void linphone_call_delete_upnp_session(LinphoneCall *call){
+	if(call->upnp_session!=NULL) {
+		linphone_upnp_session_destroy(call->upnp_session);
+		call->upnp_session=NULL;
+	}
 }
+#endif //BUILD_UPNP
 
-void linphone_call_stop_media_streams(LinphoneCall *call){
+static void linphone_call_log_fill_stats(LinphoneCallLog *log, MediaStream *st){
+	float quality=media_stream_get_average_quality_rating(st);
+	if (quality>=0){
+		if (log->quality!=-1){
+			log->quality*=quality/5.0;
+		}else log->quality=quality;
+	}
+}
+
+void linphone_call_stop_audio_stream(LinphoneCall *call) {
 	if (call->audiostream!=NULL) {
-		call->audiostream->ice_check_list = NULL;
-		rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
+		rtp_session_unregister_event_queue(call->audiostream->ms.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;
@@ -1567,26 +1881,33 @@ void linphone_call_stop_media_streams(LinphoneCall *call){
 				lp_config_set_string(call->core->config,"sound","ec_state",state_str);
 			}
 		}
-		linphone_call_log_fill_stats (call->log,call->audiostream);
+		audio_stream_get_local_rtp_stats(call->audiostream,&call->log->local_stats);
+		linphone_call_log_fill_stats (call->log,(MediaStream*)call->audiostream);
 		if (call->endpoint){
 			linphone_call_remove_from_conf(call);
 		}
 		audio_stream_stop(call->audiostream);
 		call->audiostream=NULL;
 	}
+}
 
-
+void linphone_call_stop_video_stream(LinphoneCall *call) {
 #ifdef VIDEO_ENABLED
 	if (call->videostream!=NULL){
-		call->videostream->ice_check_list = NULL;
-		rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
+		rtp_session_unregister_event_queue(call->videostream->ms.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;
+		linphone_call_log_fill_stats(call->log,(MediaStream*)call->videostream);
 		video_stream_stop(call->videostream);
 		call->videostream=NULL;
 	}
 #endif
+}
+
+void linphone_call_stop_media_streams(LinphoneCall *call){
+	linphone_call_stop_audio_stream(call);
+	linphone_call_stop_video_stream(call);
 	ms_event_queue_skip(call->core->msevq);
 	
 	if (call->audio_profile){
@@ -1694,10 +2015,20 @@ float linphone_call_get_record_volume(LinphoneCall *call){
  * active audio stream exist. Otherwise it returns the quality rating.
 **/
 float linphone_call_get_current_quality(LinphoneCall *call){
+	float audio_rating=-1;
+	float video_rating=-1;
+	float result;
 	if (call->audiostream){
-		return audio_stream_get_quality_rating(call->audiostream);
+		audio_rating=media_stream_get_quality_rating((MediaStream*)call->audiostream)/5.0;
 	}
-	return -1;
+	if (call->videostream){
+		video_rating=media_stream_get_quality_rating((MediaStream*)call->videostream)/5.0;
+	}
+	if (audio_rating<0 && video_rating<0) result=-1;
+	else if (audio_rating<0) result=video_rating*5.0;
+	else if (video_rating<0) result=audio_rating*5.0;
+	else result=audio_rating*video_rating*5.0;
+	return result;
 }
 
 /**
@@ -1712,20 +2043,83 @@ float linphone_call_get_average_quality(LinphoneCall *call){
 	return -1;
 }
 
+static void update_local_stats(LinphoneCallStats *stats, MediaStream *stream){
+	const MSQualityIndicator *qi=media_stream_get_quality_indicator(stream);
+	if (qi) {
+		stats->local_late_rate=ms_quality_indicator_get_local_late_rate(qi);
+		stats->local_loss_rate=ms_quality_indicator_get_local_loss_rate(qi);
+	}
+}
+
 /**
  * Access last known statistics for audio stream, for a given call.
 **/
-const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
-	return &call->stats[LINPHONE_CALL_STATS_AUDIO];
+const LinphoneCallStats *linphone_call_get_audio_stats(LinphoneCall *call) {
+	LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_AUDIO];
+	if (call->audiostream){
+		update_local_stats(stats,(MediaStream*)call->audiostream);
+	}
+	return stats;
 }
 
 /**
  * Access last known statistics for video stream, for a given call.
 **/
-const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
-	return &call->stats[LINPHONE_CALL_STATS_VIDEO];
+const LinphoneCallStats *linphone_call_get_video_stats(LinphoneCall *call) {
+	LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_VIDEO];
+	if (call->videostream){
+		update_local_stats(stats,(MediaStream*)call->videostream);
+	}
+	return stats;
 }
 
+/**
+ * Enable recording of the call (voice-only).
+ * This function must be used before the call parameters are assigned to the call.
+ * The call recording can be started and paused after the call is established with
+ * linphone_call_start_recording() and linphone_call_pause_recording().
+ * @param cp the call parameters
+ * @param path path and filename of the file where audio is written.
+**/
+void linphone_call_params_set_record_file(LinphoneCallParams *cp, const char *path){
+	if (cp->record_file){
+		ms_free(cp->record_file);
+		cp->record_file=NULL;
+	}
+	if (path) cp->record_file=ms_strdup(path);
+}
+
+/**
+ * Retrieves the path for the audio recoding of the call.
+**/
+const char *linphone_call_params_get_record_file(const LinphoneCallParams *cp){
+	return cp->record_file;
+}
+
+/**
+ * Start call recording.
+ * The output file where audio is recorded must be previously specified with linphone_call_params_set_record_file().
+**/
+void linphone_call_start_recording(LinphoneCall *call){
+	if (!call->params.record_file){
+		ms_error("linphone_call_start_recording(): no output file specified. Use linphone_call_params_set_record_file().");
+		return;
+	}
+	if (call->audiostream && !call->params.in_conference){
+		audio_stream_mixed_record_start(call->audiostream);
+	}
+	call->record_active=TRUE;
+}
+
+/**
+ * Stop call recording.
+**/
+void linphone_call_stop_recording(LinphoneCall *call){
+	if (call->audiostream && !call->params.in_conference){
+		audio_stream_mixed_record_stop(call->audiostream);
+	}
+	call->record_active=FALSE;
+}
 
 /**
  * @}
@@ -1752,7 +2146,7 @@ static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
 	if (from)
 	{
 		snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
-		free(from);
+		ms_free(from);
 	}
 	else
 	{
@@ -1761,6 +2155,7 @@ static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
 	if (lc->vtable.display_warning!=NULL)
 		lc->vtable.display_warning(lc,temp);
 	linphone_core_terminate_call(lc,call);
+	linphone_core_play_named_tone(lc,LinphoneToneCallFailed);
 }
 
 static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
@@ -1795,9 +2190,8 @@ static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
 			ice_session_compute_candidates_foundations(call->ice_session);
 			ice_session_eliminate_redundant_candidates(call->ice_session);
 			ice_session_choose_default_candidates(call->ice_session);
-			ping_time = ice_session_gathering_duration(call->ice_session);
+			ping_time = ice_session_average_gathering_round_trip_time(call->ice_session);
 			if (ping_time >=0) {
-				ping_time /= ice_session_nb_check_lists(call->ice_session);
 				call->ping_time=ping_time;
 			}
 		} else {
@@ -1805,23 +2199,26 @@ static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
 			linphone_call_delete_ice_session(call);
 		}
 		switch (call->state) {
-			case LinphoneCallStreamsRunning:
+			case LinphoneCallUpdating:
 				linphone_core_start_update_call(call->core, call);
 				break;
 			case LinphoneCallUpdatedByRemote:
 				linphone_core_start_accept_call_update(call->core, call);
 				break;
 			case LinphoneCallOutgoingInit:
-				linphone_call_stop_media_streams(call);
+				linphone_call_stop_media_streams_for_ice_gathering(call);
 				linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
 				break;
-			default:
-				linphone_call_stop_media_streams(call);
+			case LinphoneCallIdle:
+				linphone_call_stop_media_streams_for_ice_gathering(call);
 				linphone_core_notify_incoming_call(call->core, call);
 				break;
+			default:
+				break;
 		}
 	} else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
 		linphone_core_start_accept_call_update(call->core, call);
+		linphone_core_update_ice_state_in_call_stats(call);
 	} else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
 		ice_session_restart(call->ice_session);
 		ice_session_set_role(call->ice_session, IR_Controlling);
@@ -1838,24 +2235,29 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 		RtpSession *as=NULL,*vs=NULL;
 		float audio_load=0, video_load=0;
 		if (call->audiostream!=NULL){
-			as=call->audiostream->session;
-			if (call->audiostream->ticker)
-				audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
+			as=call->audiostream->ms.session;
+			if (call->audiostream->ms.ticker)
+				audio_load=ms_ticker_get_average_load(call->audiostream->ms.ticker);
 		}
 		if (call->videostream!=NULL){
-			if (call->videostream->ticker)
-				video_load=ms_ticker_get_average_load(call->videostream->ticker);
-			vs=call->videostream->session;
+			if (call->videostream->ms.ticker)
+				video_load=ms_ticker_get_average_load(call->videostream->ms.ticker);
+			vs=call->videostream->ms.session;
 		}
 		report_bandwidth(call,as,vs);
 		ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
 	}
+
+#ifdef BUILD_UPNP
+	linphone_upnp_call_process(call);
+#endif //BUILD_UPNP
+
 #ifdef VIDEO_ENABLED
 	if (call->videostream!=NULL) {
 		OrtpEvent *ev;
 
 		/* Ensure there is no dangling ICE check list. */
-		if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
+		if (call->ice_session == NULL) call->videostream->ms.ice_check_list = NULL;
 
 		// Beware that the application queue should not depend on treatments fron the
 		// mediastreamer queue.
@@ -1867,19 +2269,21 @@ 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->session);
+				call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->ms.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;
 				evd->packet = NULL;
+				update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
 				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->session), sizeof(jitter_stats_t));
+				memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->ms.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;
 				evd->packet = NULL;
+				update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
 				if (lc->vtable.call_stats_updated)
 					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
 			} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
@@ -1894,7 +2298,7 @@ void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapse
 		OrtpEvent *ev;
 
 		/* Ensure there is no dangling ICE check list. */
-		if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
+		if (call->ice_session == NULL) call->audiostream->ms.ice_check_list = NULL;
 
 		// Beware that the application queue should not depend on treatments fron the
 		// mediastreamer queue.
@@ -1908,24 +2312,28 @@ 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->session);
+				call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->ms.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;
 				evd->packet = NULL;
+				update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
 				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->session), sizeof(jitter_stats_t));
+				memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->ms.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;
 				evd->packet = NULL;
+				update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
 				if (lc->vtable.call_stats_updated)
 					lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
 			} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
 				|| (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
 				handle_ice_events(call, ev);
+			} else if (evt==ORTP_EVENT_TELEPHONE_EVENT){
+				linphone_core_dtmf_received(lc,evd->info.telephone_event);
 			}
 			ortp_event_destroy(ev);
 		}
@@ -1981,6 +2389,46 @@ void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState stat
 	}
 }
 
+/**
+ * Returns true if the call is part of the conference.
+ * @ingroup conferencing
+**/
 bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
 	return call->params.in_conference;
 }
+
+
+/**
+ * Perform a zoom of the video displayed during a call.
+ * @param call the call.
+ * @param zoom_factor a floating point number describing the zoom factor. A value 1.0 corresponds to no zoom applied.
+ * @param cx a floating point number pointing the horizontal center of the zoom to be applied. This value should be between 0.0 and 1.0.
+ * @param cy a floating point number pointing the vertical center of the zoom to be applied. This value should be between 0.0 and 1.0.
+ * 
+ * cx and cy are updated in return in case their coordinates were to excentrated for the requested zoom factor. The zoom ensures that all the screen is fullfilled with the video.
+**/
+void linphone_call_zoom_video(LinphoneCall* call, float zoom_factor, float* cx, float* cy) {
+	VideoStream* vstream = call->videostream;
+	if (vstream && vstream->output) {
+		float zoom[3];
+		
+		if (zoom_factor < 1)
+			zoom_factor = 1;
+		float halfsize = 0.5 * 1.0 / zoom_factor;
+
+		if ((*cx - halfsize) < 0)
+			*cx = 0 + halfsize;
+		if ((*cx + halfsize) > 1)
+			*cx = 1 - halfsize;
+		if ((*cy - halfsize) < 0)
+			*cy = 0 + halfsize;
+		if ((*cy + halfsize) > 1)
+			*cy = 1 - halfsize;
+	
+		zoom[0] = zoom_factor;
+		zoom[1] = *cx;
+		zoom[2] = *cy;
+        	ms_filter_call_method(vstream->output, MS_VIDEO_DISPLAY_ZOOM, &zoom);
+	}else ms_warning("Could not apply zoom: video output wasn't activated.");
+}
+