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);
281 linphone_address_destroy(addr);
285 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
286 SalMediaDescription *md=call->localdesc;
288 call->localdesc = create_local_media_description(lc,call);
290 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
291 sal_media_description_unref(md);
295 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
296 unsigned int id=rand() & 0xfff;
297 return _create_local_media_description(lc,call,id,id);
300 static int find_port_offset(LinphoneCore *lc){
304 bool_t already_used=FALSE;
305 for(offset=0;offset<100;offset+=2){
306 audio_port=linphone_core_get_audio_port (lc)+offset;
308 for(elem=lc->calls;elem!=NULL;elem=elem->next){
309 LinphoneCall *call=(LinphoneCall*)elem->data;
310 if (call->audio_port==audio_port) {
315 if (!already_used) break;
318 ms_error("Could not find any free port !");
324 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
326 call->magic=linphone_call_magic;
328 call->state=LinphoneCallIdle;
329 call->transfer_state = LinphoneCallIdle;
330 call->start_time=time(NULL);
331 call->media_start_time=0;
332 call->log=linphone_call_log_new(call, from, to);
333 call->owns_call_log=TRUE;
334 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
335 port_offset=find_port_offset (call->core);
336 if (port_offset==-1) return;
337 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
338 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
339 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
340 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
343 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
345 stats->received_rtcp = NULL;
346 stats->sent_rtcp = NULL;
347 stats->ice_state = LinphoneIceStateNotActivated;
351 static void discover_mtu(LinphoneCore *lc, const char *remote){
353 if (lc->net_conf.mtu==0 ){
354 /*attempt to discover mtu*/
355 mtu=ms_discover_mtu(remote);
358 ms_message("Discovered mtu is %i, RTP payload max size is %i",
359 mtu, ms_get_payload_max_size());
364 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
366 LinphoneCall *call=ms_new0(LinphoneCall,1);
367 call->dir=LinphoneCallOutgoing;
368 call->op=sal_op_new(lc->sal);
369 sal_op_set_user_pointer(call->op,call);
371 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
372 linphone_call_init_common(call,from,to);
373 call->params=*params;
374 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
375 call->ice_session = ice_session_new();
376 ice_session_set_role(call->ice_session, IR_Controlling);
378 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
379 call->ping_time=linphone_core_run_stun_tests(call->core,call);
381 call->camera_active=params->has_video;
383 discover_mtu(lc,linphone_address_get_domain (to));
384 if (params->referer){
385 sal_call_set_referer(call->op,params->referer->op);
386 call->referer=linphone_call_ref(params->referer);
391 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
392 LinphoneCall *call=ms_new0(LinphoneCall,1);
395 call->dir=LinphoneCallIncoming;
396 sal_op_set_user_pointer(op,call);
400 if (lc->sip_conf.ping_with_options){
401 /*the following sends an option request back to the caller so that
402 we get a chance to discover our nat'd address before answering.*/
403 call->ping_op=sal_op_new(lc->sal);
404 from_str=linphone_address_as_string_uri_only(from);
405 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
406 sal_op_set_user_pointer(call->ping_op,call);
407 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
411 linphone_address_clean(from);
412 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
413 linphone_call_init_common(call, from, to);
414 linphone_core_init_default_params(lc, &call->params);
415 call->params.has_video &= !!lc->video_policy.automatically_accept;
416 call->params.has_video &= linphone_core_media_description_contains_video_stream(sal_call_get_remote_media_description(op));
417 switch (linphone_core_get_firewall_policy(call->core)) {
418 case LinphonePolicyUseIce:
419 call->ice_session = ice_session_new();
420 ice_session_set_role(call->ice_session, IR_Controlled);
421 linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
422 if (call->ice_session != NULL) {
423 linphone_call_init_media_streams(call);
424 linphone_call_start_media_streams_for_ice_gathering(call);
425 if (linphone_core_gather_ice_candidates(call->core,call)<0) {
426 /* Ice candidates gathering failed, proceed with the call anyway. */
427 linphone_call_delete_ice_session(call);
428 linphone_call_stop_media_streams_for_ice_gathering(call);
432 case LinphonePolicyUseStun:
433 call->ping_time=linphone_core_run_stun_tests(call->core,call);
434 /* No break to also destroy ice session in this case. */
438 call->camera_active=call->params.has_video;
440 discover_mtu(lc,linphone_address_get_domain(from));
444 /* this function is called internally to get rid of a call.
445 It performs the following tasks:
446 - remove the call from the internal list of calls
447 - update the call logs accordingly
450 static void linphone_call_set_terminated(LinphoneCall *call){
451 LinphoneCore *lc=call->core;
453 linphone_core_update_allocated_audio_bandwidth(lc);
455 call->owns_call_log=FALSE;
456 linphone_call_log_completed(call);
459 if (call == lc->current_call){
460 ms_message("Resetting the current call");
461 lc->current_call=NULL;
464 if (linphone_core_del_call(lc,call) != 0){
465 ms_error("Could not remove the call from the list !!!");
468 if (ms_list_size(lc->calls)==0)
469 linphone_core_notify_all_friends(lc,lc->presence_mode);
471 linphone_core_conference_check_uninit(lc);
472 if (call->ringing_beep){
473 linphone_core_stop_dtmf(lc);
474 call->ringing_beep=FALSE;
477 linphone_call_unref(call->referer);
482 void linphone_call_fix_call_parameters(LinphoneCall *call){
483 call->params.has_video=call->current_params.has_video;
484 call->params.media_encryption=call->current_params.media_encryption;
487 const char *linphone_call_state_to_string(LinphoneCallState cs){
489 case LinphoneCallIdle:
490 return "LinphoneCallIdle";
491 case LinphoneCallIncomingReceived:
492 return "LinphoneCallIncomingReceived";
493 case LinphoneCallOutgoingInit:
494 return "LinphoneCallOutgoingInit";
495 case LinphoneCallOutgoingProgress:
496 return "LinphoneCallOutgoingProgress";
497 case LinphoneCallOutgoingRinging:
498 return "LinphoneCallOutgoingRinging";
499 case LinphoneCallOutgoingEarlyMedia:
500 return "LinphoneCallOutgoingEarlyMedia";
501 case LinphoneCallConnected:
502 return "LinphoneCallConnected";
503 case LinphoneCallStreamsRunning:
504 return "LinphoneCallStreamsRunning";
505 case LinphoneCallPausing:
506 return "LinphoneCallPausing";
507 case LinphoneCallPaused:
508 return "LinphoneCallPaused";
509 case LinphoneCallResuming:
510 return "LinphoneCallResuming";
511 case LinphoneCallRefered:
512 return "LinphoneCallRefered";
513 case LinphoneCallError:
514 return "LinphoneCallError";
515 case LinphoneCallEnd:
516 return "LinphoneCallEnd";
517 case LinphoneCallPausedByRemote:
518 return "LinphoneCallPausedByRemote";
519 case LinphoneCallUpdatedByRemote:
520 return "LinphoneCallUpdatedByRemote";
521 case LinphoneCallIncomingEarlyMedia:
522 return "LinphoneCallIncomingEarlyMedia";
523 case LinphoneCallUpdated:
524 return "LinphoneCallUpdated";
525 case LinphoneCallReleased:
526 return "LinphoneCallReleased";
528 return "undefined state";
531 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
532 LinphoneCore *lc=call->core;
534 if (call->state!=cstate){
535 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
536 if (cstate!=LinphoneCallReleased){
537 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
538 linphone_call_state_to_string(cstate));
542 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
543 linphone_call_state_to_string(cstate));
544 if (cstate!=LinphoneCallRefered){
545 /*LinphoneCallRefered is rather an event, not a state.
546 Indeed it does not change the state of the call (still paused or running)*/
549 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
550 switch(call->reason){
551 case LinphoneReasonDeclined:
552 call->log->status=LinphoneCallDeclined;
554 case LinphoneReasonNotAnswered:
555 call->log->status=LinphoneCallMissed;
560 linphone_call_set_terminated (call);
562 if (cstate == LinphoneCallConnected) {
563 call->log->status=LinphoneCallSuccess;
564 call->media_start_time=time(NULL);
567 if (lc->vtable.call_state_changed)
568 lc->vtable.call_state_changed(lc,call,cstate,message);
569 if (cstate==LinphoneCallReleased){
570 if (call->op!=NULL) {
571 /* so that we cannot have anymore upcalls for SAL
572 concerning this call*/
573 sal_op_release(call->op);
576 linphone_call_unref(call);
581 static void linphone_call_destroy(LinphoneCall *obj)
584 sal_op_release(obj->op);
587 if (obj->resultdesc!=NULL) {
588 sal_media_description_unref(obj->resultdesc);
589 obj->resultdesc=NULL;
591 if (obj->localdesc!=NULL) {
592 sal_media_description_unref(obj->localdesc);
596 sal_op_release(obj->ping_op);
599 ms_free(obj->refer_to);
601 if (obj->owns_call_log)
602 linphone_call_log_destroy(obj->log);
603 if (obj->auth_token) {
604 ms_free(obj->auth_token);
606 if (obj->ice_session) {
607 ice_session_destroy(obj->ice_session);
614 * @addtogroup call_control
619 * Increments the call 's reference count.
620 * An application that wishes to retain a pointer to call object
621 * must use this function to unsure the pointer remains
622 * valid. Once the application no more needs this pointer,
623 * it must call linphone_call_unref().
625 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
631 * Decrements the call object reference count.
632 * See linphone_call_ref().
634 void linphone_call_unref(LinphoneCall *obj){
637 linphone_call_destroy(obj);
642 * Returns current parameters associated to the call.
644 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
645 return &call->current_params;
648 static bool_t is_video_active(const SalStreamDescription *sd){
649 return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
653 * Returns call parameters proposed by remote.
655 * This is useful when receiving an incoming call, to know whether the remote party
656 * supports video, encryption or whatever.
658 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
659 LinphoneCallParams *cp=&call->remote_params;
660 memset(cp,0,sizeof(*cp));
662 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
664 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
666 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
667 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
668 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
669 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
671 cp->has_video=is_video_active(secure_vsd);
672 if (secure_asd || asd==NULL)
673 cp->media_encryption=LinphoneMediaEncryptionSRTP;
675 cp->has_video=is_video_active(vsd);
684 * Returns the remote address associated to this call
687 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
688 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
692 * Returns the remote address associated to this call as a string.
694 * The result string must be freed by user using ms_free().
696 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
697 return linphone_address_as_string(linphone_call_get_remote_address(call));
701 * Retrieves the call's current state.
703 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
708 * Returns the reason for a call termination (either error or normal termination)
710 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
715 * Get the user_pointer in the LinphoneCall
717 * @ingroup call_control
719 * return user_pointer an opaque user pointer that can be retrieved at any time
721 void *linphone_call_get_user_pointer(LinphoneCall *call)
723 return call->user_pointer;
727 * Set the user_pointer in the LinphoneCall
729 * @ingroup call_control
731 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
733 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
735 call->user_pointer = user_pointer;
739 * Returns the call log associated to this call.
741 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
746 * Returns the refer-to uri (if the call was transfered).
748 const char *linphone_call_get_refer_to(const LinphoneCall *call){
749 return call->refer_to;
753 * Returns direction of the call (incoming or outgoing).
755 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
756 return call->log->dir;
760 * Returns the far end's user agent description string, if available.
762 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
764 return sal_op_get_remote_ua (call->op);
770 * Returns true if this calls has received a transfer that has not been
772 * Pending transfers are executed when this call is being paused or closed,
773 * locally or by remote endpoint.
774 * If the call is already paused while receiving the transfer request, the
775 * transfer immediately occurs.
777 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
778 return call->refer_pending;
782 * Returns call's duration in seconds.
784 int linphone_call_get_duration(const LinphoneCall *call){
785 if (call->media_start_time==0) return 0;
786 return time(NULL)-call->media_start_time;
790 * Returns the call object this call is replacing, if any.
791 * Call replacement can occur during call transfers.
792 * By default, the core automatically terminates the replaced call and accept the new one.
793 * This function allows the application to know whether a new incoming call is a one that replaces another one.
795 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
796 SalOp *op=sal_call_get_replaces(call->op);
798 return (LinphoneCall*)sal_op_get_user_pointer(op);
804 * Indicate whether camera input should be sent to remote end.
806 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
808 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
809 LinphoneCore *lc=call->core;
810 MSWebCam *nowebcam=get_nowebcam_device();
811 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
812 video_stream_change_camera(call->videostream,
813 enable ? lc->video_conf.device : nowebcam);
816 call->camera_active=enable;
821 * Take a photo of currently received video and write it into a jpeg file.
823 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
825 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
826 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
828 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
835 * Returns TRUE if camera pictures are sent to the remote party.
837 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
838 return call->camera_active;
842 * Enable video stream.
844 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
845 cp->has_video=enabled;
848 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
849 return cp->audio_codec;
852 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
853 return cp->video_codec;
857 * Returns whether video is enabled.
859 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
860 return cp->has_video;
863 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
864 return cp->media_encryption;
867 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
868 cp->media_encryption = e;
873 * Enable sending of real early media (during outgoing calls).
875 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
876 cp->real_early_media=enabled;
879 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
880 return cp->real_early_media;
884 * Returns true if the call is part of the locally managed conference.
886 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
887 return cp->in_conference;
891 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
892 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
894 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
895 cp->audio_bw=bandwidth;
900 * Request remote side to send us a Video Fast Update.
902 void linphone_call_send_vfu_request(LinphoneCall *call)
904 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
905 sal_call_send_vfu_request(call->op);
912 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
913 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
914 memcpy(ncp,cp,sizeof(LinphoneCallParams));
921 void linphone_call_params_destroy(LinphoneCallParams *p){
930 #ifdef TEST_EXT_RENDERER
931 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
932 ms_message("rendercb, local buffer=%p, remote buffer=%p",
933 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
938 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
939 LinphoneCall* call = (LinphoneCall*) user_pointer;
940 ms_warning("In linphonecall.c: video_stream_event_cb");
942 case MS_VIDEO_DECODER_DECODING_ERRORS:
943 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
944 linphone_call_send_vfu_request(call);
946 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
947 ms_message("First video frame decoded successfully");
948 if (call->nextVideoFrameDecoded._func != NULL)
949 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
952 ms_warning("Unhandled event %i", event_id);
958 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
959 call->nextVideoFrameDecoded._func = cb;
960 call->nextVideoFrameDecoded._user_data = user_data;
962 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
966 void linphone_call_init_audio_stream(LinphoneCall *call){
967 LinphoneCore *lc=call->core;
968 AudioStream *audiostream;
971 if (call->audiostream != NULL) return;
972 call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,linphone_core_ipv6_enabled(lc));
973 dscp=linphone_core_get_audio_dscp(lc);
975 audio_stream_set_dscp(audiostream,dscp);
976 if (linphone_core_echo_limiter_enabled(lc)){
977 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
978 if (strcasecmp(type,"mic")==0)
979 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
980 else if (strcasecmp(type,"full")==0)
981 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
983 audio_stream_enable_gain_control(audiostream,TRUE);
984 if (linphone_core_echo_cancellation_enabled(lc)){
985 int len,delay,framesize;
986 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
987 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
988 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
989 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
990 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
991 if (statestr && audiostream->ec){
992 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
995 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
997 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
998 audio_stream_enable_noise_gate(audiostream,enabled);
1001 audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
1004 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
1005 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
1006 rtp_session_set_transports(audiostream->session,artp,artcp);
1008 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
1009 rtp_session_set_pktinfo(audiostream->session, TRUE);
1010 rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
1011 if (ice_session_check_list(call->ice_session, 0) == NULL) {
1012 ice_session_add_check_list(call->ice_session, ice_check_list_new());
1014 audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
1015 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
1018 call->audiostream_app_evq = ortp_ev_queue_new();
1019 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
1022 void linphone_call_init_video_stream(LinphoneCall *call){
1023 #ifdef VIDEO_ENABLED
1024 LinphoneCore *lc=call->core;
1026 if (!call->params.has_video) {
1027 linphone_call_stop_video_stream(call);
1030 if (call->videostream != NULL) return;
1031 if ((lc->video_conf.display || lc->video_conf.capture) && call->params.has_video){
1032 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
1033 int dscp=linphone_core_get_video_dscp(lc);
1035 call->videostream=video_stream_new(call->video_port,call->video_port+1,linphone_core_ipv6_enabled(lc));
1037 video_stream_set_dscp(call->videostream,dscp);
1038 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
1039 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
1041 if( lc->video_conf.displaytype != NULL)
1042 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
1043 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
1045 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
1046 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
1047 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
1049 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
1050 rtp_session_set_pktinfo(call->videostream->session, TRUE);
1051 rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
1052 if (ice_session_check_list(call->ice_session, 1) == NULL) {
1053 ice_session_add_check_list(call->ice_session, ice_check_list_new());
1055 call->videostream->ice_check_list = ice_session_check_list(call->ice_session, 1);
1056 ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
1058 call->videostream_app_evq = ortp_ev_queue_new();
1059 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
1060 #ifdef TEST_EXT_RENDERER
1061 video_stream_set_render_callback(call->videostream,rendercb,NULL);
1065 call->videostream=NULL;
1069 void linphone_call_init_media_streams(LinphoneCall *call){
1070 linphone_call_init_audio_stream(call);
1071 linphone_call_init_video_stream(call);
1075 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1077 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
1078 LinphoneCore* lc = (LinphoneCore*)user_data;
1079 if (dtmf<0 || dtmf>15){
1080 ms_warning("Bad dtmf value %i",dtmf);
1083 if (lc->vtable.dtmf_received != NULL)
1084 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1087 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1089 MSFilter *f=st->equalizer;
1090 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1091 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1092 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1098 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1099 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1100 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1109 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1110 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1113 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1114 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1115 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1118 audio_stream_set_mic_gain(st,mic_gain);
1120 audio_stream_set_mic_gain(st,0);
1122 recv_gain = lc->sound_conf.soft_play_lev;
1123 if (recv_gain != 0) {
1124 linphone_core_set_playback_gain_db (lc,recv_gain);
1128 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1129 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1130 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1131 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1132 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1133 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1136 if (speed==-1) speed=0.03;
1137 if (force==-1) force=25;
1138 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1139 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1141 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1143 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1144 if (transmit_thres!=-1)
1145 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1147 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1148 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1151 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1152 float floorgain = 1/mic_gain;
1153 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1154 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1155 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1156 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1158 parametrize_equalizer(lc,st);
1161 static void post_configure_audio_streams(LinphoneCall*call){
1162 AudioStream *st=call->audiostream;
1163 LinphoneCore *lc=call->core;
1164 _post_configure_audio_stream(st,lc,call->audio_muted);
1165 if (lc->vtable.dtmf_received!=NULL){
1166 /* replace by our default action*/
1167 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1168 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1172 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1175 RtpProfile *prof=rtp_profile_new("Call profile");
1178 LinphoneCore *lc=call->core;
1180 const LinphoneCallParams *params=&call->params;
1183 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1184 PayloadType *pt=(PayloadType*)elem->data;
1187 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1188 if (desc->type==SalAudio){
1189 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1190 if (params->up_ptime)
1191 up_ptime=params->up_ptime;
1192 else up_ptime=linphone_core_get_upload_ptime(lc);
1194 *used_pt=payload_type_get_number(pt);
1197 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1198 else if (md->bandwidth>0) {
1199 /*case where b=AS is given globally, not per stream*/
1200 remote_bw=md->bandwidth;
1201 if (desc->type==SalVideo){
1202 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1206 if (desc->type==SalAudio){
1207 int audio_bw=call->audio_bw;
1209 if (params->up_bw< audio_bw)
1210 audio_bw=params->up_bw;
1212 bw=get_min_bandwidth(audio_bw,remote_bw);
1213 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1214 if (bw>0) pt->normal_bitrate=bw*1000;
1215 else if (desc->type==SalAudio){
1216 pt->normal_bitrate=-1;
1219 up_ptime=desc->ptime;
1223 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1224 payload_type_append_send_fmtp(pt,tmp);
1226 number=payload_type_get_number(pt);
1227 if (rtp_profile_get_payload(prof,number)!=NULL){
1228 ms_warning("A payload type with number %i already exists in profile !",number);
1230 rtp_profile_set_payload(prof,number,pt);
1236 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1237 int pause_time=3000;
1238 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1239 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1242 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1243 LinphoneCore *lc=call->core;
1244 LinphoneCall *current=linphone_core_get_current_call(lc);
1245 return !linphone_core_is_in_conference(lc) &&
1246 (current==NULL || current==call);
1248 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1250 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1251 if (crypto[i].tag == tag) {
1257 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1258 LinphoneCore *lc=call->core;
1260 char rtcp_tool[128]={0};
1261 snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s",linphone_core_get_user_agent_name(),linphone_core_get_user_agent_version());
1262 /* look for savp stream first */
1263 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1264 SalProtoRtpSavp,SalAudio);
1265 /* no savp audio stream, use avp */
1267 stream=sal_media_description_find_stream(call->resultdesc,
1268 SalProtoRtpAvp,SalAudio);
1270 if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1271 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1272 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1273 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1274 const char *playfile=lc->play_file;
1275 const char *recfile=lc->rec_file;
1276 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1280 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1281 if (playcard==NULL) {
1282 ms_warning("No card defined for playback !");
1284 if (captcard==NULL) {
1285 ms_warning("No card defined for capture !");
1287 /*Replace soundcard filters by inactive file players or recorders
1288 when placed in recvonly or sendonly mode*/
1289 if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1292 }else if (stream->dir==SalStreamSendOnly){
1296 /*And we will eventually play "playfile" if set by the user*/
1299 if (send_ringbacktone){
1301 playfile=NULL;/* it is setup later*/
1303 /*if playfile are supplied don't use soundcards*/
1304 if (lc->use_files) {
1308 if (call->params.in_conference){
1309 /* first create the graph without soundcard resources*/
1310 captcard=playcard=NULL;
1312 if (!linphone_call_sound_resources_available(call)){
1313 ms_message("Sound resources are used by another call, not using soundcard.");
1314 captcard=playcard=NULL;
1316 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1317 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1318 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1319 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1320 audio_stream_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
1321 audio_stream_start_full(
1323 call->audio_profile,
1324 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1326 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1327 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1329 linphone_core_get_audio_jittcomp(lc),
1336 post_configure_audio_streams(call);
1337 if (muted && !send_ringbacktone){
1338 audio_stream_set_mic_gain(call->audiostream,0);
1340 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1342 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1344 if (send_ringbacktone){
1345 setup_ring_player(lc,call);
1347 audio_stream_set_rtcp_information(call->audiostream, cname, rtcp_tool);
1349 /* valid local tags are > 0 */
1350 if (stream->proto == SalProtoRtpSavp) {
1351 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1352 SalProtoRtpSavp,SalAudio);
1353 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1355 if (crypto_idx >= 0) {
1356 audio_stream_enable_strp(
1358 stream->crypto[0].algo,
1359 local_st_desc->crypto[crypto_idx].master_key,
1360 stream->crypto[0].master_key);
1361 call->audiostream_encrypted=TRUE;
1363 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1364 call->audiostream_encrypted=FALSE;
1366 }else call->audiostream_encrypted=FALSE;
1367 if (call->params.in_conference){
1368 /*transform the graph to connect it to the conference filter */
1369 bool_t mute=stream->dir==SalStreamRecvOnly;
1370 linphone_call_add_to_conf(call, mute);
1372 call->current_params.in_conference=call->params.in_conference;
1373 }else ms_warning("No audio stream accepted ?");
1377 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1378 #ifdef VIDEO_ENABLED
1379 LinphoneCore *lc=call->core;
1381 /* look for savp stream first */
1382 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1383 SalProtoRtpSavp,SalVideo);
1384 char rtcp_tool[128]={0};
1385 snprintf(rtcp_tool,sizeof(rtcp_tool)-1,"%s-%s",linphone_core_get_user_agent_name(),linphone_core_get_user_agent_version());
1387 /* no savp audio stream, use avp */
1389 vstream=sal_media_description_find_stream(call->resultdesc,
1390 SalProtoRtpAvp,SalVideo);
1392 /* shutdown preview */
1393 if (lc->previewstream!=NULL) {
1394 video_preview_stop(lc->previewstream);
1395 lc->previewstream=NULL;
1398 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1399 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1400 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1401 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1403 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1404 VideoStreamDir dir=VideoStreamSendRecv;
1405 MSWebCam *cam=lc->video_conf.device;
1406 bool_t is_inactive=FALSE;
1408 call->current_params.has_video=TRUE;
1410 video_stream_enable_adaptive_bitrate_control(call->videostream,
1411 linphone_core_adaptive_rate_control_enabled(lc));
1412 video_stream_enable_adaptive_jittcomp(call->videostream, linphone_core_video_adaptive_jittcomp_enabled(lc));
1413 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1414 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1415 if (lc->video_window_id!=0)
1416 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1417 if (lc->preview_window_id!=0)
1418 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1419 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1421 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1422 cam=get_nowebcam_device();
1423 dir=VideoStreamSendOnly;
1424 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1425 dir=VideoStreamRecvOnly;
1426 }else if (vstream->dir==SalStreamSendRecv){
1427 if (lc->video_conf.display && lc->video_conf.capture)
1428 dir=VideoStreamSendRecv;
1429 else if (lc->video_conf.display)
1430 dir=VideoStreamRecvOnly;
1432 dir=VideoStreamSendOnly;
1434 ms_warning("video stream is inactive.");
1435 /*either inactive or incompatible with local capabilities*/
1438 if (call->camera_active==FALSE || all_inputs_muted){
1439 cam=get_nowebcam_device();
1442 call->log->video_enabled = TRUE;
1443 video_stream_set_direction (call->videostream, dir);
1444 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1445 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1446 video_stream_start(call->videostream,
1447 call->video_profile, rtp_addr, vstream->rtp_port,
1448 rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1449 used_pt, linphone_core_get_video_jittcomp(lc), cam);
1450 video_stream_set_rtcp_information(call->videostream, cname,rtcp_tool);
1453 if (vstream->proto == SalProtoRtpSavp) {
1454 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1455 SalProtoRtpSavp,SalVideo);
1457 video_stream_enable_strp(
1459 vstream->crypto[0].algo,
1460 local_st_desc->crypto[0].master_key,
1461 vstream->crypto[0].master_key
1463 call->videostream_encrypted=TRUE;
1465 call->videostream_encrypted=FALSE;
1467 }else ms_warning("No video stream accepted.");
1469 ms_warning("No valid video stream defined.");
1474 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1475 LinphoneCore *lc=call->core;
1477 call->current_params.audio_codec = NULL;
1478 call->current_params.video_codec = NULL;
1480 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1482 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1483 #ifdef VIDEO_ENABLED
1484 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1485 SalProtoRtpAvp,SalVideo);
1488 if ((call->audiostream == NULL) && (call->videostream == NULL)) {
1489 ms_fatal("start_media_stream() called without prior init !");
1492 cname=linphone_address_as_string_uri_only(me);
1494 #if defined(VIDEO_ENABLED)
1495 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1496 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1500 if (call->audiostream!=NULL) {
1501 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1503 call->current_params.has_video=FALSE;
1504 if (call->videostream!=NULL) {
1505 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1508 call->all_muted=all_inputs_muted;
1509 call->playing_ringbacktone=send_ringbacktone;
1510 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1512 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1513 OrtpZrtpParams params;
1514 /*will be set later when zrtp is activated*/
1515 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1517 params.zid_file=lc->zrtp_secrets_cache;
1518 audio_stream_enable_zrtp(call->audiostream,¶ms);
1519 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1520 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1521 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1524 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1525 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1526 linphone_call_fix_call_parameters(call);
1527 if ((call->ice_session != NULL) && (ice_session_state(call->ice_session) != IS_Completed)) {
1528 ice_session_start_connectivity_checks(call->ice_session);
1534 linphone_address_destroy(me);
1537 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1538 audio_stream_prepare_sound(call->audiostream, NULL, NULL);
1539 #ifdef VIDEO_ENABLED
1540 if (call->videostream) {
1541 video_stream_prepare_video(call->videostream);
1546 void linphone_call_stop_media_streams_for_ice_gathering(LinphoneCall *call){
1547 audio_stream_unprepare_sound(call->audiostream);
1548 #ifdef VIDEO_ENABLED
1549 if (call->videostream) {
1550 video_stream_unprepare_video(call->videostream);
1555 void linphone_call_delete_ice_session(LinphoneCall *call){
1556 if (call->ice_session != NULL) {
1557 ice_session_destroy(call->ice_session);
1558 call->ice_session = NULL;
1559 if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
1560 if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
1561 call->stats[LINPHONE_CALL_STATS_AUDIO].ice_state = LinphoneIceStateNotActivated;
1562 call->stats[LINPHONE_CALL_STATS_VIDEO].ice_state = LinphoneIceStateNotActivated;
1566 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1567 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1568 log->quality=audio_stream_get_average_quality_rating(st);
1571 void linphone_call_stop_audio_stream(LinphoneCall *call) {
1572 if (call->audiostream!=NULL) {
1573 call->audiostream->ice_check_list = NULL;
1574 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1575 ortp_ev_queue_flush(call->audiostream_app_evq);
1576 ortp_ev_queue_destroy(call->audiostream_app_evq);
1577 call->audiostream_app_evq=NULL;
1579 if (call->audiostream->ec){
1580 const char *state_str=NULL;
1581 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1583 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1584 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1587 linphone_call_log_fill_stats (call->log,call->audiostream);
1588 if (call->endpoint){
1589 linphone_call_remove_from_conf(call);
1591 audio_stream_stop(call->audiostream);
1592 call->audiostream=NULL;
1596 void linphone_call_stop_video_stream(LinphoneCall *call) {
1597 #ifdef VIDEO_ENABLED
1598 if (call->videostream!=NULL){
1599 call->videostream->ice_check_list = NULL;
1600 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1601 ortp_ev_queue_flush(call->videostream_app_evq);
1602 ortp_ev_queue_destroy(call->videostream_app_evq);
1603 call->videostream_app_evq=NULL;
1604 video_stream_stop(call->videostream);
1605 call->videostream=NULL;
1610 void linphone_call_stop_media_streams(LinphoneCall *call){
1611 linphone_call_stop_audio_stream(call);
1612 linphone_call_stop_video_stream(call);
1613 ms_event_queue_skip(call->core->msevq);
1615 if (call->audio_profile){
1616 rtp_profile_clear_all(call->audio_profile);
1617 rtp_profile_destroy(call->audio_profile);
1618 call->audio_profile=NULL;
1620 if (call->video_profile){
1621 rtp_profile_clear_all(call->video_profile);
1622 rtp_profile_destroy(call->video_profile);
1623 call->video_profile=NULL;
1629 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1630 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1631 bool_t bypass_mode = !enable;
1632 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1635 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1636 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1638 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1641 return linphone_core_echo_cancellation_enabled(call->core);
1645 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1646 if (call!=NULL && call->audiostream!=NULL ) {
1648 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1649 if (strcasecmp(type,"mic")==0)
1650 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1651 else if (strcasecmp(type,"full")==0)
1652 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1654 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1659 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1660 if (call!=NULL && call->audiostream!=NULL ){
1661 return call->audiostream->el_type !=ELInactive ;
1663 return linphone_core_echo_limiter_enabled(call->core);
1668 * @addtogroup call_misc
1673 * Returns the measured sound volume played locally (received from remote).
1674 * It is expressed in dbm0.
1676 float linphone_call_get_play_volume(LinphoneCall *call){
1677 AudioStream *st=call->audiostream;
1678 if (st && st->volrecv){
1680 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1684 return LINPHONE_VOLUME_DB_LOWEST;
1688 * Returns the measured sound volume recorded locally (sent to remote).
1689 * It is expressed in dbm0.
1691 float linphone_call_get_record_volume(LinphoneCall *call){
1692 AudioStream *st=call->audiostream;
1693 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1695 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1699 return LINPHONE_VOLUME_DB_LOWEST;
1703 * Obtain real-time quality rating of the call
1705 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1706 * during all the duration of the call. This function returns its value at the time of the function call.
1707 * It is expected that the rating is updated at least every 5 seconds or so.
1708 * The rating is a floating point number comprised between 0 and 5.
1710 * 4-5 = good quality <br>
1711 * 3-4 = average quality <br>
1712 * 2-3 = poor quality <br>
1713 * 1-2 = very poor quality <br>
1714 * 0-1 = can't be worse, mostly unusable <br>
1716 * @returns The function returns -1 if no quality measurement is available, for example if no
1717 * active audio stream exist. Otherwise it returns the quality rating.
1719 float linphone_call_get_current_quality(LinphoneCall *call){
1720 if (call->audiostream){
1721 return audio_stream_get_quality_rating(call->audiostream);
1727 * Returns call quality averaged over all the duration of the call.
1729 * See linphone_call_get_current_quality() for more details about quality measurement.
1731 float linphone_call_get_average_quality(LinphoneCall *call){
1732 if (call->audiostream){
1733 return audio_stream_get_average_quality_rating(call->audiostream);
1739 * Access last known statistics for audio stream, for a given call.
1741 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1742 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1746 * Access last known statistics for video stream, for a given call.
1748 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1749 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1757 static void report_bandwidth(LinphoneCall *call, RtpSession *as, RtpSession *vs){
1758 call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth=(as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0;
1759 call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth=(as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0;
1760 call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth=(vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0;
1761 call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth=(vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0;
1762 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1763 call->stats[LINPHONE_CALL_STATS_AUDIO].download_bandwidth,
1764 call->stats[LINPHONE_CALL_STATS_AUDIO].upload_bandwidth ,
1765 call->stats[LINPHONE_CALL_STATS_VIDEO].download_bandwidth,
1766 call->stats[LINPHONE_CALL_STATS_VIDEO].upload_bandwidth
1770 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1774 from = linphone_call_get_remote_address_as_string(call);
1777 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1782 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1784 if (lc->vtable.display_warning!=NULL)
1785 lc->vtable.display_warning(lc,temp);
1786 linphone_core_terminate_call(lc,call);
1789 static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
1790 OrtpEventType evt=ortp_event_get_type(ev);
1791 OrtpEventData *evd=ortp_event_get_data(ev);
1794 if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1795 switch (ice_session_state(call->ice_session)) {
1797 ice_session_select_candidates(call->ice_session);
1798 if (ice_session_role(call->ice_session) == IR_Controlling) {
1799 linphone_core_update_call(call->core, call, &call->current_params);
1803 if (ice_session_has_completed_check_list(call->ice_session) == TRUE) {
1804 ice_session_select_candidates(call->ice_session);
1805 if (ice_session_role(call->ice_session) == IR_Controlling) {
1806 /* At least one ICE session has succeeded, so perform a call update. */
1807 linphone_core_update_call(call->core, call, &call->current_params);
1814 linphone_core_update_ice_state_in_call_stats(call);
1815 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1817 if (evd->info.ice_processing_successful==TRUE) {
1818 ice_session_compute_candidates_foundations(call->ice_session);
1819 ice_session_eliminate_redundant_candidates(call->ice_session);
1820 ice_session_choose_default_candidates(call->ice_session);
1821 ping_time = ice_session_gathering_duration(call->ice_session);
1822 if (ping_time >=0) {
1823 ping_time /= ice_session_nb_check_lists(call->ice_session);
1824 call->ping_time=ping_time;
1827 ms_warning("No STUN answer from [%s], disabling ICE",linphone_core_get_stun_server(call->core));
1828 linphone_call_delete_ice_session(call);
1830 switch (call->state) {
1831 case LinphoneCallStreamsRunning:
1832 linphone_core_start_update_call(call->core, call);
1834 case LinphoneCallUpdatedByRemote:
1835 linphone_core_start_accept_call_update(call->core, call);
1837 case LinphoneCallOutgoingInit:
1838 linphone_call_stop_media_streams_for_ice_gathering(call);
1839 linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
1842 linphone_call_stop_media_streams_for_ice_gathering(call);
1843 linphone_core_notify_incoming_call(call->core, call);
1846 } else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
1847 linphone_core_start_accept_call_update(call->core, call);
1848 } else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
1849 ice_session_restart(call->ice_session);
1850 ice_session_set_role(call->ice_session, IR_Controlling);
1851 linphone_core_update_call(call->core, call, &call->current_params);
1855 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1856 LinphoneCore* lc = call->core;
1857 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1858 bool_t disconnected=FALSE;
1860 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1861 RtpSession *as=NULL,*vs=NULL;
1862 float audio_load=0, video_load=0;
1863 if (call->audiostream!=NULL){
1864 as=call->audiostream->session;
1865 if (call->audiostream->ticker)
1866 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1868 if (call->videostream!=NULL){
1869 if (call->videostream->ticker)
1870 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1871 vs=call->videostream->session;
1873 report_bandwidth(call,as,vs);
1874 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1876 #ifdef VIDEO_ENABLED
1877 if (call->videostream!=NULL) {
1880 /* Ensure there is no dangling ICE check list. */
1881 if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
1883 // Beware that the application queue should not depend on treatments fron the
1884 // mediastreamer queue.
1885 video_stream_iterate(call->videostream);
1887 while (call->videostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq)))){
1888 OrtpEventType evt=ortp_event_get_type(ev);
1889 OrtpEventData *evd=ortp_event_get_data(ev);
1890 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1891 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1892 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1893 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1894 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1895 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1896 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1898 if (lc->vtable.call_stats_updated)
1899 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1900 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1901 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1902 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1903 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1904 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1906 if (lc->vtable.call_stats_updated)
1907 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1908 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1909 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1910 handle_ice_events(call, ev);
1912 ortp_event_destroy(ev);
1916 if (call->audiostream!=NULL) {
1919 /* Ensure there is no dangling ICE check list. */
1920 if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
1922 // Beware that the application queue should not depend on treatments fron the
1923 // mediastreamer queue.
1924 audio_stream_iterate(call->audiostream);
1926 while (call->audiostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq)))){
1927 OrtpEventType evt=ortp_event_get_type(ev);
1928 OrtpEventData *evd=ortp_event_get_data(ev);
1929 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1930 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1931 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1932 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1933 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1934 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1935 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1936 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1937 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1939 if (lc->vtable.call_stats_updated)
1940 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1941 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1942 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1943 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1944 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1945 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1947 if (lc->vtable.call_stats_updated)
1948 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1949 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1950 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1951 handle_ice_events(call, ev);
1953 ortp_event_destroy(ev);
1956 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1957 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1959 linphone_core_disconnected(call->core,call);
1962 void linphone_call_log_completed(LinphoneCall *call){
1963 LinphoneCore *lc=call->core;
1965 call->log->duration=time(NULL)-call->start_time;
1967 if (call->log->status==LinphoneCallMissed){
1970 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1971 "You have missed %i calls.", lc->missed_calls),
1973 if (lc->vtable.display_status!=NULL)
1974 lc->vtable.display_status(lc,info);
1977 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1978 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1979 MSList *elem,*prevelem=NULL;
1980 /*find the last element*/
1981 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1985 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1986 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1988 if (lc->vtable.call_log_updated!=NULL){
1989 lc->vtable.call_log_updated(lc,call->log);
1991 call_logs_write_to_config_file(lc);
1994 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1995 return call->transfer_state;
1998 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1999 if (state != call->transfer_state) {
2000 LinphoneCore* lc = call->core;
2001 call->transfer_state = state;
2002 if (lc->vtable.transfer_state_changed)
2003 lc->vtable.transfer_state_changed(lc, call, state);
2007 bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
2008 return call->params.in_conference;