4 Copyright (C) 2010 Belledonne Communications SARL
5 (simon.morlat@linphone.org)
7 This program is free software; you can redistribute it and/or
8 modify it under the terms of the GNU General Public License
9 as published by the Free Software Foundation; either version 2
10 of the License, or (at your option) any later version.
12 This program is distributed in the hope that it will be useful,
13 but WITHOUT ANY WARRANTY; without even the implied warranty of
14 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 GNU General Public License for more details.
17 You should have received a copy of the GNU General Public License
18 along with this program; if not, write to the Free Software
19 Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
24 #include "linphonecore.h"
28 #include <ortp/event.h>
32 #include "mediastreamer2/mediastream.h"
33 #include "mediastreamer2/msvolume.h"
34 #include "mediastreamer2/msequalizer.h"
35 #include "mediastreamer2/msfileplayer.h"
36 #include "mediastreamer2/msjpegwriter.h"
37 #include "mediastreamer2/mseventqueue.h"
38 #include "mediastreamer2/mssndcard.h"
41 static MSWebCam *get_nowebcam_device(){
42 return ms_web_cam_manager_get_cam(ms_web_cam_manager_get(),"StaticImage: Static picture");
46 static bool_t generate_b64_crypto_key(int key_length, char* key_out) {
48 uint8_t* tmp = (uint8_t*) malloc(key_length);
49 if (ortp_crypto_get_random(tmp, key_length)!=0) {
50 ms_error("Failed to generate random key");
55 b64_size = b64_encode((const char*)tmp, key_length, NULL, 0);
57 ms_error("Failed to b64 encode key");
61 key_out[b64_size] = '\0';
62 b64_encode((const char*)tmp, key_length, key_out, 40);
67 LinphoneCore *linphone_call_get_core(const LinphoneCall *call){
71 const char* linphone_call_get_authentication_token(LinphoneCall *call){
72 return call->auth_token;
75 bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
76 return call->auth_token_verified;
79 static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
80 // Check ZRTP encryption in audiostream
81 if (!call->audiostream_encrypted) {
86 // If video enabled, check ZRTP encryption in videostream
87 const LinphoneCallParams *params=linphone_call_get_current_params(call);
88 if (params->has_video && !call->videostream_encrypted) {
96 void propagate_encryption_changed(LinphoneCall *call){
97 LinphoneCore *lc=call->core;
98 if (!linphone_call_are_all_streams_encrypted(call)) {
99 ms_message("Some streams are not encrypted");
100 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
101 if (lc->vtable.call_encryption_changed)
102 lc->vtable.call_encryption_changed(call->core, call, FALSE, call->auth_token);
104 ms_message("All streams are encrypted");
105 call->current_params.media_encryption=LinphoneMediaEncryptionZRTP;
106 if (lc->vtable.call_encryption_changed)
107 lc->vtable.call_encryption_changed(call->core, call, TRUE, call->auth_token);
112 static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
113 ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
115 LinphoneCall *call = (LinphoneCall *)data;
116 call->videostream_encrypted=encrypted;
117 propagate_encryption_changed(call);
121 static void linphone_call_audiostream_encryption_changed(void *data, bool_t encrypted) {
122 char status[255]={0};
123 ms_message("Audio stream is %s ", encrypted ? "encrypted" : "not encrypted");
125 LinphoneCall *call = (LinphoneCall *)data;
126 call->audiostream_encrypted=encrypted;
128 if (encrypted && call->core->vtable.display_status != NULL) {
129 snprintf(status,sizeof(status)-1,_("Authentication token is %s"),call->auth_token);
130 call->core->vtable.display_status(call->core, status);
133 propagate_encryption_changed(call);
137 // Enable video encryption
138 const LinphoneCallParams *params=linphone_call_get_current_params(call);
139 if (params->has_video) {
140 ms_message("Trying to enable encryption on video stream");
141 OrtpZrtpParams params;
142 params.zid_file=NULL; //unused
143 video_stream_enable_zrtp(call->videostream,call->audiostream,¶ms);
149 static void linphone_call_audiostream_auth_token_ready(void *data, const char* auth_token, bool_t verified) {
150 LinphoneCall *call=(LinphoneCall *)data;
151 if (call->auth_token != NULL)
152 ms_free(call->auth_token);
154 call->auth_token=ms_strdup(auth_token);
155 call->auth_token_verified=verified;
157 ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
160 void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t verified){
161 if (call->audiostream==NULL){
162 ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
164 if (call->audiostream->ortpZrtpContext==NULL){
165 ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
167 if (!call->auth_token_verified && verified){
168 ortp_zrtp_sas_verified(call->audiostream->ortpZrtpContext);
169 }else if (call->auth_token_verified && !verified){
170 ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
172 call->auth_token_verified=verified;
173 propagate_encryption_changed(call);
176 static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
179 if (max_sample_rate) *max_sample_rate=0;
180 for(it=codecs;it!=NULL;it=it->next){
181 PayloadType *pt=(PayloadType*)it->data;
182 if (pt->flags & PAYLOAD_TYPE_ENABLED){
183 if (bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,bandwidth_limit)){
184 ms_message("Codec %s/%i eliminated because of audio bandwidth constraint.",pt->mime_type,pt->clock_rate);
187 if (linphone_core_check_payload_type_usability(lc,pt)){
188 l=ms_list_append(l,payload_type_clone(pt));
189 if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
196 static void update_media_description_from_stun(SalMediaDescription *md, const StunCandidate *ac, const StunCandidate *vc){
198 strcpy(md->streams[0].rtp_addr,ac->addr);
199 md->streams[0].rtp_port=ac->port;
200 if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->nstreams==1){
201 strcpy(md->addr,ac->addr);
205 strcpy(md->streams[1].rtp_addr,vc->addr);
206 md->streams[1].rtp_port=vc->port;
212 static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
216 const char *me=linphone_core_get_identity(lc);
217 LinphoneAddress *addr=linphone_address_new(me);
218 const char *username=linphone_address_get_username (addr);
219 SalMediaDescription *md=sal_media_description_new();
221 if (call->ping_time>0) {
222 linphone_core_adapt_to_network(lc,call->ping_time,&call->params);
225 md->session_id=session_id;
226 md->session_ver=session_ver;
228 strncpy(md->addr,call->localip,sizeof(md->addr));
229 strncpy(md->username,username,sizeof(md->username));
231 if (call->params.down_bw)
232 md->bandwidth=call->params.down_bw;
233 else md->bandwidth=linphone_core_get_download_bandwidth(lc);
235 /*set audio capabilities */
236 strncpy(md->streams[0].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
237 strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
238 md->streams[0].rtp_port=call->audio_port;
239 md->streams[0].rtcp_port=call->audio_port+1;
240 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
241 SalProtoRtpSavp : SalProtoRtpAvp;
242 md->streams[0].type=SalAudio;
243 if (call->params.down_ptime)
244 md->streams[0].ptime=call->params.down_ptime;
246 md->streams[0].ptime=linphone_core_get_download_ptime(lc);
247 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
248 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
249 l=ms_list_append(l,pt);
250 md->streams[0].payloads=l;
254 if (call->params.has_video){
256 md->streams[1].rtp_port=call->video_port;
257 md->streams[1].rtcp_port=call->video_port+1;
258 md->streams[1].proto=md->streams[0].proto;
259 md->streams[1].type=SalVideo;
260 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
261 md->streams[1].payloads=l;
264 for(i=0; i<md->nstreams; i++) {
265 if (md->streams[i].proto == SalProtoRtpSavp) {
266 md->streams[i].crypto[0].tag = 1;
267 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
268 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
269 md->streams[i].crypto[0].algo = 0;
270 md->streams[i].crypto[1].tag = 2;
271 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
272 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
273 md->streams[i].crypto[1].algo = 0;
274 md->streams[i].crypto[2].algo = 0;
277 update_media_description_from_stun(md,&call->ac,&call->vc);
278 if (call->ice_session != NULL) {
279 linphone_core_update_local_media_description_from_ice(md, call->ice_session);
280 linphone_core_update_ice_state_in_call_stats(call);
282 linphone_address_destroy(addr);
286 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
287 SalMediaDescription *md=call->localdesc;
289 call->localdesc = create_local_media_description(lc,call);
291 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
292 sal_media_description_unref(md);
296 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
297 unsigned int id=rand() & 0xfff;
298 return _create_local_media_description(lc,call,id,id);
301 static int find_port_offset(LinphoneCore *lc){
305 bool_t already_used=FALSE;
306 for(offset=0;offset<100;offset+=2){
307 audio_port=linphone_core_get_audio_port (lc)+offset;
309 for(elem=lc->calls;elem!=NULL;elem=elem->next){
310 LinphoneCall *call=(LinphoneCall*)elem->data;
311 if (call->audio_port==audio_port) {
316 if (!already_used) break;
319 ms_error("Could not find any free port !");
325 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
327 call->magic=linphone_call_magic;
329 call->state=LinphoneCallIdle;
330 call->transfer_state = LinphoneCallIdle;
331 call->start_time=time(NULL);
332 call->media_start_time=0;
333 call->log=linphone_call_log_new(call, from, to);
334 call->owns_call_log=TRUE;
335 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
336 port_offset=find_port_offset (call->core);
337 if (port_offset==-1) return;
338 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
339 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
340 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
341 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
344 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
346 stats->received_rtcp = NULL;
347 stats->sent_rtcp = NULL;
348 stats->ice_state = LinphoneIceStateNotActivated;
352 static void discover_mtu(LinphoneCore *lc, const char *remote){
354 if (lc->net_conf.mtu==0 ){
355 /*attempt to discover mtu*/
356 mtu=ms_discover_mtu(remote);
359 ms_message("Discovered mtu is %i, RTP payload max size is %i",
360 mtu, ms_get_payload_max_size());
365 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
367 LinphoneCall *call=ms_new0(LinphoneCall,1);
368 call->dir=LinphoneCallOutgoing;
369 call->op=sal_op_new(lc->sal);
370 sal_op_set_user_pointer(call->op,call);
372 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
373 linphone_call_init_common(call,from,to);
374 call->params=*params;
375 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
376 call->ice_session = ice_session_new();
377 ice_session_set_role(call->ice_session, IR_Controlling);
379 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
380 call->ping_time=linphone_core_run_stun_tests(call->core,call);
382 call->camera_active=params->has_video;
384 discover_mtu(lc,linphone_address_get_domain (to));
385 if (params->referer){
386 sal_call_set_referer(call->op,params->referer->op);
387 call->referer=linphone_call_ref(params->referer);
392 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
393 LinphoneCall *call=ms_new0(LinphoneCall,1);
396 call->dir=LinphoneCallIncoming;
397 sal_op_set_user_pointer(op,call);
401 if (lc->sip_conf.ping_with_options){
402 /*the following sends an option request back to the caller so that
403 we get a chance to discover our nat'd address before answering.*/
404 call->ping_op=sal_op_new(lc->sal);
405 from_str=linphone_address_as_string_uri_only(from);
406 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
407 sal_op_set_user_pointer(call->ping_op,call);
408 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
412 linphone_address_clean(from);
413 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
414 linphone_call_init_common(call, from, to);
415 call->log->call_id=ms_strdup(sal_op_get_call_id(op)); /*must be known at that time*/
416 linphone_core_init_default_params(lc, &call->params);
417 call->params.has_video &= !!lc->video_policy.automatically_accept;
418 call->params.has_video &= linphone_core_media_description_contains_video_stream(sal_call_get_remote_media_description(op));
419 switch (linphone_core_get_firewall_policy(call->core)) {
420 case LinphonePolicyUseIce:
421 call->ice_session = ice_session_new();
422 ice_session_set_role(call->ice_session, IR_Controlled);
423 linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
424 if (call->ice_session != NULL) {
425 linphone_call_init_media_streams(call);
426 linphone_call_start_media_streams_for_ice_gathering(call);
427 if (linphone_core_gather_ice_candidates(call->core,call)<0) {
428 /* Ice candidates gathering failed, proceed with the call anyway. */
429 linphone_call_delete_ice_session(call);
430 linphone_call_stop_media_streams_for_ice_gathering(call);
434 case LinphonePolicyUseStun:
435 call->ping_time=linphone_core_run_stun_tests(call->core,call);
436 /* No break to also destroy ice session in this case. */
440 call->camera_active=call->params.has_video;
442 discover_mtu(lc,linphone_address_get_domain(from));
446 /* this function is called internally to get rid of a call.
447 It performs the following tasks:
448 - remove the call from the internal list of calls
449 - update the call logs accordingly
452 static void linphone_call_set_terminated(LinphoneCall *call){
453 LinphoneCore *lc=call->core;
455 linphone_core_update_allocated_audio_bandwidth(lc);
457 call->owns_call_log=FALSE;
458 linphone_call_log_completed(call);
461 if (call == lc->current_call){
462 ms_message("Resetting the current call");
463 lc->current_call=NULL;
466 if (linphone_core_del_call(lc,call) != 0){
467 ms_error("Could not remove the call from the list !!!");
470 if (ms_list_size(lc->calls)==0)
471 linphone_core_notify_all_friends(lc,lc->presence_mode);
473 linphone_core_conference_check_uninit(lc);
474 if (call->ringing_beep){
475 linphone_core_stop_dtmf(lc);
476 call->ringing_beep=FALSE;
479 linphone_call_unref(call->referer);
484 void linphone_call_fix_call_parameters(LinphoneCall *call){
485 call->params.has_video=call->current_params.has_video;
486 call->params.media_encryption=call->current_params.media_encryption;
489 const char *linphone_call_state_to_string(LinphoneCallState cs){
491 case LinphoneCallIdle:
492 return "LinphoneCallIdle";
493 case LinphoneCallIncomingReceived:
494 return "LinphoneCallIncomingReceived";
495 case LinphoneCallOutgoingInit:
496 return "LinphoneCallOutgoingInit";
497 case LinphoneCallOutgoingProgress:
498 return "LinphoneCallOutgoingProgress";
499 case LinphoneCallOutgoingRinging:
500 return "LinphoneCallOutgoingRinging";
501 case LinphoneCallOutgoingEarlyMedia:
502 return "LinphoneCallOutgoingEarlyMedia";
503 case LinphoneCallConnected:
504 return "LinphoneCallConnected";
505 case LinphoneCallStreamsRunning:
506 return "LinphoneCallStreamsRunning";
507 case LinphoneCallPausing:
508 return "LinphoneCallPausing";
509 case LinphoneCallPaused:
510 return "LinphoneCallPaused";
511 case LinphoneCallResuming:
512 return "LinphoneCallResuming";
513 case LinphoneCallRefered:
514 return "LinphoneCallRefered";
515 case LinphoneCallError:
516 return "LinphoneCallError";
517 case LinphoneCallEnd:
518 return "LinphoneCallEnd";
519 case LinphoneCallPausedByRemote:
520 return "LinphoneCallPausedByRemote";
521 case LinphoneCallUpdatedByRemote:
522 return "LinphoneCallUpdatedByRemote";
523 case LinphoneCallIncomingEarlyMedia:
524 return "LinphoneCallIncomingEarlyMedia";
525 case LinphoneCallUpdating:
526 return "LinphoneCallUpdating";
527 case LinphoneCallReleased:
528 return "LinphoneCallReleased";
530 return "undefined state";
533 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
534 LinphoneCore *lc=call->core;
536 if (call->state!=cstate){
537 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
538 if (cstate!=LinphoneCallReleased){
539 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
540 linphone_call_state_to_string(cstate));
544 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
545 linphone_call_state_to_string(cstate));
546 if (cstate!=LinphoneCallRefered){
547 /*LinphoneCallRefered is rather an event, not a state.
548 Indeed it does not change the state of the call (still paused or running)*/
551 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
552 switch(call->reason){
553 case LinphoneReasonDeclined:
554 call->log->status=LinphoneCallDeclined;
556 case LinphoneReasonNotAnswered:
557 call->log->status=LinphoneCallMissed;
562 linphone_call_set_terminated (call);
564 if (cstate == LinphoneCallConnected) {
565 call->log->status=LinphoneCallSuccess;
566 call->media_start_time=time(NULL);
569 if (lc->vtable.call_state_changed)
570 lc->vtable.call_state_changed(lc,call,cstate,message);
571 if (cstate==LinphoneCallReleased){
572 if (call->op!=NULL) {
573 /* so that we cannot have anymore upcalls for SAL
574 concerning this call*/
575 sal_op_release(call->op);
578 linphone_call_unref(call);
583 static void linphone_call_destroy(LinphoneCall *obj)
586 sal_op_release(obj->op);
589 if (obj->resultdesc!=NULL) {
590 sal_media_description_unref(obj->resultdesc);
591 obj->resultdesc=NULL;
593 if (obj->localdesc!=NULL) {
594 sal_media_description_unref(obj->localdesc);
598 sal_op_release(obj->ping_op);
601 ms_free(obj->refer_to);
603 if (obj->owns_call_log)
604 linphone_call_log_destroy(obj->log);
605 if (obj->auth_token) {
606 ms_free(obj->auth_token);
608 if (obj->ice_session) {
609 ice_session_destroy(obj->ice_session);
616 * @addtogroup call_control
621 * Increments the call 's reference count.
622 * An application that wishes to retain a pointer to call object
623 * must use this function to unsure the pointer remains
624 * valid. Once the application no more needs this pointer,
625 * it must call linphone_call_unref().
627 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
633 * Decrements the call object reference count.
634 * See linphone_call_ref().
636 void linphone_call_unref(LinphoneCall *obj){
639 linphone_call_destroy(obj);
644 * Returns current parameters associated to the call.
646 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
647 return &call->current_params;
650 static bool_t is_video_active(const SalStreamDescription *sd){
651 return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
655 * Returns call parameters proposed by remote.
657 * This is useful when receiving an incoming call, to know whether the remote party
658 * supports video, encryption or whatever.
660 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
661 LinphoneCallParams *cp=&call->remote_params;
662 memset(cp,0,sizeof(*cp));
664 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
666 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
668 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
669 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
670 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
671 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
673 cp->has_video=is_video_active(secure_vsd);
674 if (secure_asd || asd==NULL)
675 cp->media_encryption=LinphoneMediaEncryptionSRTP;
677 cp->has_video=is_video_active(vsd);
686 * Returns the remote address associated to this call
689 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
690 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
694 * Returns the remote address associated to this call as a string.
696 * The result string must be freed by user using ms_free().
698 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
699 return linphone_address_as_string(linphone_call_get_remote_address(call));
703 * Retrieves the call's current state.
705 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
710 * Returns the reason for a call termination (either error or normal termination)
712 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
717 * Get the user_pointer in the LinphoneCall
719 * @ingroup call_control
721 * return user_pointer an opaque user pointer that can be retrieved at any time
723 void *linphone_call_get_user_pointer(LinphoneCall *call)
725 return call->user_pointer;
729 * Set the user_pointer in the LinphoneCall
731 * @ingroup call_control
733 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
735 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
737 call->user_pointer = user_pointer;
741 * Returns the call log associated to this call.
743 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
748 * Returns the refer-to uri (if the call was transfered).
750 const char *linphone_call_get_refer_to(const LinphoneCall *call){
751 return call->refer_to;
755 * Returns direction of the call (incoming or outgoing).
757 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
758 return call->log->dir;
762 * Returns the far end's user agent description string, if available.
764 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
766 return sal_op_get_remote_ua (call->op);
772 * Returns true if this calls has received a transfer that has not been
774 * Pending transfers are executed when this call is being paused or closed,
775 * locally or by remote endpoint.
776 * If the call is already paused while receiving the transfer request, the
777 * transfer immediately occurs.
779 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
780 return call->refer_pending;
784 * Returns call's duration in seconds.
786 int linphone_call_get_duration(const LinphoneCall *call){
787 if (call->media_start_time==0) return 0;
788 return time(NULL)-call->media_start_time;
792 * Returns the call object this call is replacing, if any.
793 * Call replacement can occur during call transfers.
794 * By default, the core automatically terminates the replaced call and accept the new one.
795 * This function allows the application to know whether a new incoming call is a one that replaces another one.
797 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
798 SalOp *op=sal_call_get_replaces(call->op);
800 return (LinphoneCall*)sal_op_get_user_pointer(op);
806 * Indicate whether camera input should be sent to remote end.
808 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
810 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
811 LinphoneCore *lc=call->core;
812 MSWebCam *nowebcam=get_nowebcam_device();
813 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
814 video_stream_change_camera(call->videostream,
815 enable ? lc->video_conf.device : nowebcam);
818 call->camera_active=enable;
823 * Take a photo of currently received video and write it into a jpeg file.
825 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
827 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
828 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
830 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
837 * Returns TRUE if camera pictures are sent to the remote party.
839 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
840 return call->camera_active;
844 * Enable video stream.
846 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
847 cp->has_video=enabled;
850 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
851 return cp->audio_codec;
854 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
855 return cp->video_codec;
858 bool_t linphone_call_params_low_bandwidth_enabled(const LinphoneCallParams *cp) {
859 return cp->low_bandwidth;
862 * Returns whether video is enabled.
864 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
865 return cp->has_video;
868 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
869 return cp->media_encryption;
872 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
873 cp->media_encryption = e;
878 * Enable sending of real early media (during outgoing calls).
880 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
881 cp->real_early_media=enabled;
884 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
885 return cp->real_early_media;
889 * Returns true if the call is part of the locally managed conference.
891 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
892 return cp->in_conference;
896 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
897 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
899 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
900 cp->audio_bw=bandwidth;
905 * Request remote side to send us a Video Fast Update.
907 void linphone_call_send_vfu_request(LinphoneCall *call)
909 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
910 sal_call_send_vfu_request(call->op);
917 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
918 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
919 memcpy(ncp,cp,sizeof(LinphoneCallParams));
926 void linphone_call_params_destroy(LinphoneCallParams *p){
935 #ifdef TEST_EXT_RENDERER
936 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
937 ms_message("rendercb, local buffer=%p, remote buffer=%p",
938 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
943 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
944 LinphoneCall* call = (LinphoneCall*) user_pointer;
945 ms_warning("In linphonecall.c: video_stream_event_cb");
947 case MS_VIDEO_DECODER_DECODING_ERRORS:
948 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
949 linphone_call_send_vfu_request(call);
951 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
952 ms_message("First video frame decoded successfully");
953 if (call->nextVideoFrameDecoded._func != NULL)
954 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
957 ms_warning("Unhandled event %i", event_id);
963 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
964 call->nextVideoFrameDecoded._func = cb;
965 call->nextVideoFrameDecoded._user_data = user_data;
967 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
971 void linphone_call_init_audio_stream(LinphoneCall *call){
972 LinphoneCore *lc=call->core;
973 AudioStream *audiostream;
976 if (call->audiostream != NULL) return;
977 call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,linphone_core_ipv6_enabled(lc));
978 dscp=linphone_core_get_audio_dscp(lc);
980 audio_stream_set_dscp(audiostream,dscp);
981 if (linphone_core_echo_limiter_enabled(lc)){
982 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
983 if (strcasecmp(type,"mic")==0)
984 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
985 else if (strcasecmp(type,"full")==0)
986 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
988 audio_stream_enable_gain_control(audiostream,TRUE);
989 if (linphone_core_echo_cancellation_enabled(lc)){
990 int len,delay,framesize;
991 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
992 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
993 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
994 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
995 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
996 if (statestr && audiostream->ec){
997 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
1000 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
1002 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
1003 audio_stream_enable_noise_gate(audiostream,enabled);
1006 audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
1009 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
1010 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
1011 rtp_session_set_transports(audiostream->session,artp,artcp);
1013 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
1014 rtp_session_set_pktinfo(audiostream->session, TRUE);
1015 rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
1016 if (ice_session_check_list(call->ice_session, 0) == NULL) {
1017 ice_session_add_check_list(call->ice_session, ice_check_list_new());
1019 audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
1020 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
1023 call->audiostream_app_evq = ortp_ev_queue_new();
1024 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
1027 void linphone_call_init_video_stream(LinphoneCall *call){
1028 #ifdef VIDEO_ENABLED
1029 LinphoneCore *lc=call->core;
1031 if (!call->params.has_video) {
1032 linphone_call_stop_video_stream(call);
1035 if (call->videostream != NULL) return;
1036 if ((lc->video_conf.display || lc->video_conf.capture) && call->params.has_video){
1037 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
1038 int dscp=linphone_core_get_video_dscp(lc);
1040 call->videostream=video_stream_new(call->video_port,call->video_port+1,linphone_core_ipv6_enabled(lc));
1042 video_stream_set_dscp(call->videostream,dscp);
1043 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
1044 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
1046 if( lc->video_conf.displaytype != NULL)
1047 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
1048 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
1050 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
1051 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
1052 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
1054 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
1055 rtp_session_set_pktinfo(call->videostream->session, TRUE);
1056 rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
1057 if (ice_session_check_list(call->ice_session, 1) == NULL) {
1058 ice_session_add_check_list(call->ice_session, ice_check_list_new());
1060 call->videostream->ice_check_list = ice_session_check_list(call->ice_session, 1);
1061 ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
1063 call->videostream_app_evq = ortp_ev_queue_new();
1064 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
1065 #ifdef TEST_EXT_RENDERER
1066 video_stream_set_render_callback(call->videostream,rendercb,NULL);
1070 call->videostream=NULL;
1074 void linphone_call_init_media_streams(LinphoneCall *call){
1075 linphone_call_init_audio_stream(call);
1076 linphone_call_init_video_stream(call);
1080 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1082 static void linphone_core_dtmf_received(LinphoneCore *lc, int dtmf){
1083 if (dtmf<0 || dtmf>15){
1084 ms_warning("Bad dtmf value %i",dtmf);
1087 if (lc->vtable.dtmf_received != NULL)
1088 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1091 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1093 MSFilter *f=st->equalizer;
1094 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1095 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1096 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1102 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1103 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1104 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1113 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1114 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1117 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1118 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1119 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1122 audio_stream_set_mic_gain(st,mic_gain);
1124 audio_stream_set_mic_gain(st,0);
1126 recv_gain = lc->sound_conf.soft_play_lev;
1127 if (recv_gain != 0) {
1128 linphone_core_set_playback_gain_db (lc,recv_gain);
1132 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1133 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1134 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1135 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1136 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1137 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1140 if (speed==-1) speed=0.03;
1141 if (force==-1) force=25;
1142 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1143 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1145 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1147 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1148 if (transmit_thres!=-1)
1149 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1151 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1152 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1155 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1156 float floorgain = 1/mic_gain;
1157 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1158 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1159 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1160 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1162 parametrize_equalizer(lc,st);
1165 static void post_configure_audio_streams(LinphoneCall*call){
1166 AudioStream *st=call->audiostream;
1167 LinphoneCore *lc=call->core;
1168 _post_configure_audio_stream(st,lc,call->audio_muted);
1169 if (lc->vtable.dtmf_received!=NULL){
1170 /* replace by our default action*/
1171 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1172 /*rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);*/
1176 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1179 RtpProfile *prof=rtp_profile_new("Call profile");
1182 LinphoneCore *lc=call->core;
1184 const LinphoneCallParams *params=&call->params;
1187 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1188 PayloadType *pt=(PayloadType*)elem->data;
1191 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1192 if (desc->type==SalAudio){
1193 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1194 if (params->up_ptime)
1195 up_ptime=params->up_ptime;
1196 else up_ptime=linphone_core_get_upload_ptime(lc);
1198 *used_pt=payload_type_get_number(pt);
1201 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1202 else if (md->bandwidth>0) {
1203 /*case where b=AS is given globally, not per stream*/
1204 remote_bw=md->bandwidth;
1205 if (desc->type==SalVideo){
1206 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1210 if (desc->type==SalAudio){
1211 int audio_bw=call->audio_bw;
1213 if (params->up_bw< audio_bw)
1214 audio_bw=params->up_bw;
1216 bw=get_min_bandwidth(audio_bw,remote_bw);
1217 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1218 if (bw>0) pt->normal_bitrate=bw*1000;
1219 else if (desc->type==SalAudio){
1220 pt->normal_bitrate=-1;
1223 up_ptime=desc->ptime;
1227 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1228 payload_type_append_send_fmtp(pt,tmp);
1230 number=payload_type_get_number(pt);
1231 if (rtp_profile_get_payload(prof,number)!=NULL){
1232 ms_warning("A payload type with number %i already exists in profile !",number);
1234 rtp_profile_set_payload(prof,number,pt);
1240 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1241 int pause_time=3000;
1242 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1243 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1246 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1247 LinphoneCore *lc=call->core;
1248 LinphoneCall *current=linphone_core_get_current_call(lc);
1249 return !linphone_core_is_in_conference(lc) &&
1250 (current==NULL || current==call);
1252 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1254 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1255 if (crypto[i].tag == tag) {
1261 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1262 LinphoneCore *lc=call->core;
1264 char rtcp_tool[128]={0};
1265 snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s",linphone_core_get_user_agent_name(),linphone_core_get_user_agent_version());
1266 /* look for savp stream first */
1267 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1268 SalProtoRtpSavp,SalAudio);
1269 /* no savp audio stream, use avp */
1271 stream=sal_media_description_find_stream(call->resultdesc,
1272 SalProtoRtpAvp,SalAudio);
1274 if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1275 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1276 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1277 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1278 const char *playfile=lc->play_file;
1279 const char *recfile=lc->rec_file;
1280 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1284 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1285 if (playcard==NULL) {
1286 ms_warning("No card defined for playback !");
1288 if (captcard==NULL) {
1289 ms_warning("No card defined for capture !");
1291 /*Replace soundcard filters by inactive file players or recorders
1292 when placed in recvonly or sendonly mode*/
1293 if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1296 }else if (stream->dir==SalStreamSendOnly){
1300 /*And we will eventually play "playfile" if set by the user*/
1303 if (send_ringbacktone){
1305 playfile=NULL;/* it is setup later*/
1307 /*if playfile are supplied don't use soundcards*/
1308 if (lc->use_files) {
1312 if (call->params.in_conference){
1313 /* first create the graph without soundcard resources*/
1314 captcard=playcard=NULL;
1316 if (!linphone_call_sound_resources_available(call)){
1317 ms_message("Sound resources are used by another call, not using soundcard.");
1318 captcard=playcard=NULL;
1320 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1321 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1322 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1323 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1324 audio_stream_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
1325 audio_stream_start_full(
1327 call->audio_profile,
1328 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1330 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1331 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1333 linphone_core_get_audio_jittcomp(lc),
1340 post_configure_audio_streams(call);
1341 if (muted && !send_ringbacktone){
1342 audio_stream_set_mic_gain(call->audiostream,0);
1344 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1346 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1348 if (send_ringbacktone){
1349 setup_ring_player(lc,call);
1351 audio_stream_set_rtcp_information(call->audiostream, cname, rtcp_tool);
1353 /* valid local tags are > 0 */
1354 if (stream->proto == SalProtoRtpSavp) {
1355 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1356 SalProtoRtpSavp,SalAudio);
1357 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1359 if (crypto_idx >= 0) {
1360 audio_stream_enable_strp(
1362 stream->crypto[0].algo,
1363 local_st_desc->crypto[crypto_idx].master_key,
1364 stream->crypto[0].master_key);
1365 call->audiostream_encrypted=TRUE;
1367 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1368 call->audiostream_encrypted=FALSE;
1370 }else call->audiostream_encrypted=FALSE;
1371 if (call->params.in_conference){
1372 /*transform the graph to connect it to the conference filter */
1373 bool_t mute=stream->dir==SalStreamRecvOnly;
1374 linphone_call_add_to_conf(call, mute);
1376 call->current_params.in_conference=call->params.in_conference;
1377 }else ms_warning("No audio stream accepted ?");
1381 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1382 #ifdef VIDEO_ENABLED
1383 LinphoneCore *lc=call->core;
1385 /* look for savp stream first */
1386 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1387 SalProtoRtpSavp,SalVideo);
1388 char rtcp_tool[128]={0};
1389 snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s",linphone_core_get_user_agent_name(),linphone_core_get_user_agent_version());
1391 /* no savp audio stream, use avp */
1393 vstream=sal_media_description_find_stream(call->resultdesc,
1394 SalProtoRtpAvp,SalVideo);
1396 /* shutdown preview */
1397 if (lc->previewstream!=NULL) {
1398 video_preview_stop(lc->previewstream);
1399 lc->previewstream=NULL;
1402 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1403 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1404 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1405 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1407 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1408 VideoStreamDir dir=VideoStreamSendRecv;
1409 MSWebCam *cam=lc->video_conf.device;
1410 bool_t is_inactive=FALSE;
1412 call->current_params.has_video=TRUE;
1414 video_stream_enable_adaptive_bitrate_control(call->videostream,
1415 linphone_core_adaptive_rate_control_enabled(lc));
1416 video_stream_enable_adaptive_jittcomp(call->videostream, linphone_core_video_adaptive_jittcomp_enabled(lc));
1417 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1418 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1419 if (lc->video_window_id!=0)
1420 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1421 if (lc->preview_window_id!=0)
1422 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1423 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1425 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1426 cam=get_nowebcam_device();
1427 dir=VideoStreamSendOnly;
1428 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1429 dir=VideoStreamRecvOnly;
1430 }else if (vstream->dir==SalStreamSendRecv){
1431 if (lc->video_conf.display && lc->video_conf.capture)
1432 dir=VideoStreamSendRecv;
1433 else if (lc->video_conf.display)
1434 dir=VideoStreamRecvOnly;
1436 dir=VideoStreamSendOnly;
1438 ms_warning("video stream is inactive.");
1439 /*either inactive or incompatible with local capabilities*/
1442 if (call->camera_active==FALSE || all_inputs_muted){
1443 cam=get_nowebcam_device();
1446 call->log->video_enabled = TRUE;
1447 video_stream_set_direction (call->videostream, dir);
1448 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1449 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1450 video_stream_start(call->videostream,
1451 call->video_profile, rtp_addr, vstream->rtp_port,
1452 rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1453 used_pt, linphone_core_get_video_jittcomp(lc), cam);
1454 video_stream_set_rtcp_information(call->videostream, cname,rtcp_tool);
1457 if (vstream->proto == SalProtoRtpSavp) {
1458 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1459 SalProtoRtpSavp,SalVideo);
1461 video_stream_enable_strp(
1463 vstream->crypto[0].algo,
1464 local_st_desc->crypto[0].master_key,
1465 vstream->crypto[0].master_key
1467 call->videostream_encrypted=TRUE;
1469 call->videostream_encrypted=FALSE;
1471 }else ms_warning("No video stream accepted.");
1473 ms_warning("No valid video stream defined.");
1478 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1479 LinphoneCore *lc=call->core;
1481 call->current_params.audio_codec = NULL;
1482 call->current_params.video_codec = NULL;
1484 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1486 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1487 #ifdef VIDEO_ENABLED
1488 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1489 SalProtoRtpAvp,SalVideo);
1492 if ((call->audiostream == NULL) && (call->videostream == NULL)) {
1493 ms_fatal("start_media_stream() called without prior init !");
1496 cname=linphone_address_as_string_uri_only(me);
1498 #if defined(VIDEO_ENABLED)
1499 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1500 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1504 if (call->audiostream!=NULL) {
1505 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1507 call->current_params.has_video=FALSE;
1508 if (call->videostream!=NULL) {
1509 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1512 call->all_muted=all_inputs_muted;
1513 call->playing_ringbacktone=send_ringbacktone;
1514 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1516 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1517 OrtpZrtpParams params;
1518 /*will be set later when zrtp is activated*/
1519 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1521 params.zid_file=lc->zrtp_secrets_cache;
1522 audio_stream_enable_zrtp(call->audiostream,¶ms);
1523 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1524 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1525 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1528 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1529 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1530 linphone_call_fix_call_parameters(call);
1531 if ((call->ice_session != NULL) && (ice_session_state(call->ice_session) != IS_Completed)) {
1532 ice_session_start_connectivity_checks(call->ice_session);
1538 linphone_address_destroy(me);
1541 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1542 audio_stream_prepare_sound(call->audiostream, NULL, NULL);
1543 #ifdef VIDEO_ENABLED
1544 if (call->videostream) {
1545 video_stream_prepare_video(call->videostream);
1550 void linphone_call_stop_media_streams_for_ice_gathering(LinphoneCall *call){
1551 audio_stream_unprepare_sound(call->audiostream);
1552 #ifdef VIDEO_ENABLED
1553 if (call->videostream) {
1554 video_stream_unprepare_video(call->videostream);
1559 void linphone_call_delete_ice_session(LinphoneCall *call){
1560 if (call->ice_session != NULL) {
1561 ice_session_destroy(call->ice_session);
1562 call->ice_session = NULL;
1563 if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
1564 if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
1565 call->stats[LINPHONE_CALL_STATS_AUDIO].ice_state = LinphoneIceStateNotActivated;
1566 call->stats[LINPHONE_CALL_STATS_VIDEO].ice_state = LinphoneIceStateNotActivated;
1570 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1571 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1572 log->quality=audio_stream_get_average_quality_rating(st);
1575 void linphone_call_stop_audio_stream(LinphoneCall *call) {
1576 if (call->audiostream!=NULL) {
1577 call->audiostream->ice_check_list = NULL;
1578 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1579 ortp_ev_queue_flush(call->audiostream_app_evq);
1580 ortp_ev_queue_destroy(call->audiostream_app_evq);
1581 call->audiostream_app_evq=NULL;
1583 if (call->audiostream->ec){
1584 const char *state_str=NULL;
1585 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1587 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1588 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1591 linphone_call_log_fill_stats (call->log,call->audiostream);
1592 if (call->endpoint){
1593 linphone_call_remove_from_conf(call);
1595 audio_stream_stop(call->audiostream);
1596 call->audiostream=NULL;
1600 void linphone_call_stop_video_stream(LinphoneCall *call) {
1601 #ifdef VIDEO_ENABLED
1602 if (call->videostream!=NULL){
1603 call->videostream->ice_check_list = NULL;
1604 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1605 ortp_ev_queue_flush(call->videostream_app_evq);
1606 ortp_ev_queue_destroy(call->videostream_app_evq);
1607 call->videostream_app_evq=NULL;
1608 video_stream_stop(call->videostream);
1609 call->videostream=NULL;
1614 void linphone_call_stop_media_streams(LinphoneCall *call){
1615 linphone_call_stop_audio_stream(call);
1616 linphone_call_stop_video_stream(call);
1617 ms_event_queue_skip(call->core->msevq);
1619 if (call->audio_profile){
1620 rtp_profile_clear_all(call->audio_profile);
1621 rtp_profile_destroy(call->audio_profile);
1622 call->audio_profile=NULL;
1624 if (call->video_profile){
1625 rtp_profile_clear_all(call->video_profile);
1626 rtp_profile_destroy(call->video_profile);
1627 call->video_profile=NULL;
1633 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1634 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1635 bool_t bypass_mode = !enable;
1636 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1639 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1640 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1642 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1645 return linphone_core_echo_cancellation_enabled(call->core);
1649 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1650 if (call!=NULL && call->audiostream!=NULL ) {
1652 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1653 if (strcasecmp(type,"mic")==0)
1654 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1655 else if (strcasecmp(type,"full")==0)
1656 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1658 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1663 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1664 if (call!=NULL && call->audiostream!=NULL ){
1665 return call->audiostream->el_type !=ELInactive ;
1667 return linphone_core_echo_limiter_enabled(call->core);
1672 * @addtogroup call_misc
1677 * Returns the measured sound volume played locally (received from remote).
1678 * It is expressed in dbm0.
1680 float linphone_call_get_play_volume(LinphoneCall *call){
1681 AudioStream *st=call->audiostream;
1682 if (st && st->volrecv){
1684 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1688 return LINPHONE_VOLUME_DB_LOWEST;
1692 * Returns the measured sound volume recorded locally (sent to remote).
1693 * It is expressed in dbm0.
1695 float linphone_call_get_record_volume(LinphoneCall *call){
1696 AudioStream *st=call->audiostream;
1697 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1699 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1703 return LINPHONE_VOLUME_DB_LOWEST;
1707 * Obtain real-time quality rating of the call
1709 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1710 * during all the duration of the call. This function returns its value at the time of the function call.
1711 * It is expected that the rating is updated at least every 5 seconds or so.
1712 * The rating is a floating point number comprised between 0 and 5.
1714 * 4-5 = good quality <br>
1715 * 3-4 = average quality <br>
1716 * 2-3 = poor quality <br>
1717 * 1-2 = very poor quality <br>
1718 * 0-1 = can't be worse, mostly unusable <br>
1720 * @returns The function returns -1 if no quality measurement is available, for example if no
1721 * active audio stream exist. Otherwise it returns the quality rating.
1723 float linphone_call_get_current_quality(LinphoneCall *call){
1724 if (call->audiostream){
1725 return audio_stream_get_quality_rating(call->audiostream);
1731 * Returns call quality averaged over all the duration of the call.
1733 * See linphone_call_get_current_quality() for more details about quality measurement.
1735 float linphone_call_get_average_quality(LinphoneCall *call){
1736 if (call->audiostream){
1737 return audio_stream_get_average_quality_rating(call->audiostream);
1743 * Access last known statistics for audio stream, for a given call.
1745 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1746 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1750 * Access last known statistics for video stream, for a given call.
1752 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1753 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1761 static void report_bandwidth(LinphoneCall *call, RtpSession *as, RtpSession *vs){
1762 call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth=(as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0;
1763 call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth=(as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0;
1764 call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth=(vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0;
1765 call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth=(vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0;
1766 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1767 call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth,
1768 call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth ,
1769 call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth,
1770 call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth
1774 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1778 from = linphone_call_get_remote_address_as_string(call);
1781 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1786 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1788 if (lc->vtable.display_warning!=NULL)
1789 lc->vtable.display_warning(lc,temp);
1790 linphone_core_terminate_call(lc,call);
1793 static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
1794 OrtpEventType evt=ortp_event_get_type(ev);
1795 OrtpEventData *evd=ortp_event_get_data(ev);
1798 if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1799 switch (ice_session_state(call->ice_session)) {
1801 ice_session_select_candidates(call->ice_session);
1802 if (ice_session_role(call->ice_session) == IR_Controlling) {
1803 linphone_core_update_call(call->core, call, &call->current_params);
1807 if (ice_session_has_completed_check_list(call->ice_session) == TRUE) {
1808 ice_session_select_candidates(call->ice_session);
1809 if (ice_session_role(call->ice_session) == IR_Controlling) {
1810 /* At least one ICE session has succeeded, so perform a call update. */
1811 linphone_core_update_call(call->core, call, &call->current_params);
1818 linphone_core_update_ice_state_in_call_stats(call);
1819 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1821 if (evd->info.ice_processing_successful==TRUE) {
1822 ice_session_compute_candidates_foundations(call->ice_session);
1823 ice_session_eliminate_redundant_candidates(call->ice_session);
1824 ice_session_choose_default_candidates(call->ice_session);
1825 ping_time = ice_session_gathering_duration(call->ice_session);
1826 if (ping_time >=0) {
1827 ping_time /= ice_session_nb_check_lists(call->ice_session);
1828 call->ping_time=ping_time;
1831 ms_warning("No STUN answer from [%s], disabling ICE",linphone_core_get_stun_server(call->core));
1832 linphone_call_delete_ice_session(call);
1834 switch (call->state) {
1835 case LinphoneCallUpdating:
1836 linphone_core_start_update_call(call->core, call);
1838 case LinphoneCallUpdatedByRemote:
1839 linphone_core_start_accept_call_update(call->core, call);
1841 case LinphoneCallOutgoingInit:
1842 linphone_call_stop_media_streams_for_ice_gathering(call);
1843 linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
1845 case LinphoneCallIdle:
1846 linphone_call_stop_media_streams_for_ice_gathering(call);
1847 linphone_core_notify_incoming_call(call->core, call);
1852 } else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
1853 linphone_core_start_accept_call_update(call->core, call);
1854 linphone_core_update_ice_state_in_call_stats(call);
1855 } else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
1856 ice_session_restart(call->ice_session);
1857 ice_session_set_role(call->ice_session, IR_Controlling);
1858 linphone_core_update_call(call->core, call, &call->current_params);
1862 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1863 LinphoneCore* lc = call->core;
1864 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1865 bool_t disconnected=FALSE;
1867 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1868 RtpSession *as=NULL,*vs=NULL;
1869 float audio_load=0, video_load=0;
1870 if (call->audiostream!=NULL){
1871 as=call->audiostream->session;
1872 if (call->audiostream->ticker)
1873 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1875 if (call->videostream!=NULL){
1876 if (call->videostream->ticker)
1877 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1878 vs=call->videostream->session;
1880 report_bandwidth(call,as,vs);
1881 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1883 #ifdef VIDEO_ENABLED
1884 if (call->videostream!=NULL) {
1887 /* Ensure there is no dangling ICE check list. */
1888 if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
1890 // Beware that the application queue should not depend on treatments fron the
1891 // mediastreamer queue.
1892 video_stream_iterate(call->videostream);
1894 while (call->videostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq)))){
1895 OrtpEventType evt=ortp_event_get_type(ev);
1896 OrtpEventData *evd=ortp_event_get_data(ev);
1897 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1898 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1899 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1900 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1901 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1902 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1903 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1905 if (lc->vtable.call_stats_updated)
1906 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1907 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1908 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1909 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1910 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1911 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1913 if (lc->vtable.call_stats_updated)
1914 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1915 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1916 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1917 handle_ice_events(call, ev);
1919 ortp_event_destroy(ev);
1923 if (call->audiostream!=NULL) {
1926 /* Ensure there is no dangling ICE check list. */
1927 if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
1929 // Beware that the application queue should not depend on treatments fron the
1930 // mediastreamer queue.
1931 audio_stream_iterate(call->audiostream);
1933 while (call->audiostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq)))){
1934 OrtpEventType evt=ortp_event_get_type(ev);
1935 OrtpEventData *evd=ortp_event_get_data(ev);
1936 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1937 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1938 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1939 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1940 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1941 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1942 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1943 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1944 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1946 if (lc->vtable.call_stats_updated)
1947 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1948 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1949 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1950 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1951 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1952 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1954 if (lc->vtable.call_stats_updated)
1955 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1956 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1957 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1958 handle_ice_events(call, ev);
1959 } else if (evt==ORTP_EVENT_TELEPHONE_EVENT){
1960 linphone_core_dtmf_received(lc,evd->info.telephone_event);
1962 ortp_event_destroy(ev);
1965 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1966 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1968 linphone_core_disconnected(call->core,call);
1971 void linphone_call_log_completed(LinphoneCall *call){
1972 LinphoneCore *lc=call->core;
1974 call->log->duration=time(NULL)-call->start_time;
1976 if (call->log->status==LinphoneCallMissed){
1979 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1980 "You have missed %i calls.", lc->missed_calls),
1982 if (lc->vtable.display_status!=NULL)
1983 lc->vtable.display_status(lc,info);
1986 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1987 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1988 MSList *elem,*prevelem=NULL;
1989 /*find the last element*/
1990 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1994 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1995 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1997 if (lc->vtable.call_log_updated!=NULL){
1998 lc->vtable.call_log_updated(lc,call->log);
2000 call_logs_write_to_config_file(lc);
2003 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
2004 return call->transfer_state;
2007 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
2008 if (state != call->transfer_state) {
2009 LinphoneCore* lc = call->core;
2010 call->transfer_state = state;
2011 if (lc->vtable.transfer_state_changed)
2012 lc->vtable.transfer_state_changed(lc, call, state);
2016 bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
2017 return call->params.in_conference;