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 SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
200 const char *me=linphone_core_get_identity(lc);
201 LinphoneAddress *addr=linphone_address_new(me);
202 const char *username=linphone_address_get_username (addr);
203 SalMediaDescription *md=sal_media_description_new();
205 md->session_id=session_id;
206 md->session_ver=session_ver;
208 strncpy(md->addr,call->localip,sizeof(md->addr));
209 strncpy(md->username,username,sizeof(md->username));
211 if (call->params.down_bw)
212 md->bandwidth=call->params.down_bw;
213 else md->bandwidth=linphone_core_get_download_bandwidth(lc);
215 /*set audio capabilities */
216 strncpy(md->streams[0].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
217 strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
218 md->streams[0].rtp_port=call->audio_port;
219 md->streams[0].rtcp_port=call->audio_port+1;
220 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
221 SalProtoRtpSavp : SalProtoRtpAvp;
222 md->streams[0].type=SalAudio;
223 if (call->params.down_ptime)
224 md->streams[0].ptime=call->params.down_ptime;
226 md->streams[0].ptime=linphone_core_get_download_ptime(lc);
227 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
228 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
229 l=ms_list_append(l,pt);
230 md->streams[0].payloads=l;
234 if (call->params.has_video){
236 md->streams[1].rtp_port=call->video_port;
237 md->streams[1].rtcp_port=call->video_port+1;
238 md->streams[1].proto=md->streams[0].proto;
239 md->streams[1].type=SalVideo;
240 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
241 md->streams[1].payloads=l;
244 for(i=0; i<md->nstreams; i++) {
245 if (md->streams[i].proto == SalProtoRtpSavp) {
246 md->streams[i].crypto[0].tag = 1;
247 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
248 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
249 md->streams[i].crypto[0].algo = 0;
250 md->streams[i].crypto[1].tag = 2;
251 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
252 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
253 md->streams[i].crypto[1].algo = 0;
254 md->streams[i].crypto[2].algo = 0;
256 if ((linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, i) == NULL)) {
257 ice_session_add_check_list(call->ice_session, ice_check_list_new());
261 linphone_address_destroy(addr);
265 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
266 SalMediaDescription *md=call->localdesc;
268 call->localdesc = create_local_media_description(lc,call);
270 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
271 sal_media_description_unref(md);
275 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
276 unsigned int id=rand() & 0xfff;
277 return _create_local_media_description(lc,call,id,id);
280 static int find_port_offset(LinphoneCore *lc){
284 bool_t already_used=FALSE;
285 for(offset=0;offset<100;offset+=2){
286 audio_port=linphone_core_get_audio_port (lc)+offset;
288 for(elem=lc->calls;elem!=NULL;elem=elem->next){
289 LinphoneCall *call=(LinphoneCall*)elem->data;
290 if (call->audio_port==audio_port) {
295 if (!already_used) break;
298 ms_error("Could not find any free port !");
304 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
306 call->magic=linphone_call_magic;
308 call->state=LinphoneCallIdle;
309 call->transfer_state = LinphoneCallIdle;
310 call->start_time=time(NULL);
311 call->media_start_time=0;
312 call->log=linphone_call_log_new(call, from, to);
313 call->owns_call_log=TRUE;
314 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
315 port_offset=find_port_offset (call->core);
316 if (port_offset==-1) return;
317 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
318 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
319 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
320 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
323 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
325 stats->received_rtcp = NULL;
326 stats->sent_rtcp = NULL;
329 static void update_media_description_from_stun(SalMediaDescription *md, const StunCandidate *ac, const StunCandidate *vc){
331 strcpy(md->streams[0].rtp_addr,ac->addr);
332 md->streams[0].rtp_port=ac->port;
333 if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->nstreams==1){
334 strcpy(md->addr,ac->addr);
338 strcpy(md->streams[1].rtp_addr,vc->addr);
339 md->streams[1].rtp_port=vc->port;
344 static void discover_mtu(LinphoneCore *lc, const char *remote){
346 if (lc->net_conf.mtu==0 ){
347 /*attempt to discover mtu*/
348 mtu=ms_discover_mtu(remote);
351 ms_message("Discovered mtu is %i, RTP payload max size is %i",
352 mtu, ms_get_payload_max_size());
357 #define STUN_CANDIDATE_INIT {{0},0}
359 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
361 LinphoneCall *call=ms_new0(LinphoneCall,1);
362 StunCandidate ac=STUN_CANDIDATE_INIT,vc=STUN_CANDIDATE_INIT;
364 call->dir=LinphoneCallOutgoing;
365 call->op=sal_op_new(lc->sal);
366 sal_op_set_user_pointer(call->op,call);
368 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
369 linphone_call_init_common(call,from,to);
370 call->params=*params;
371 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
372 call->ice_session = ice_session_new();
373 ice_session_set_role(call->ice_session, IR_Controlling);
375 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
376 ping_time=linphone_core_run_stun_tests(call->core,call,&ac, &vc);
379 linphone_core_adapt_to_network(lc,ping_time,&call->params);
381 call->localdesc=create_local_media_description(lc,call);
382 update_media_description_from_stun(call->localdesc,&ac,&vc);
383 call->camera_active=params->has_video;
385 discover_mtu(lc,linphone_address_get_domain (to));
386 if (params->referer){
387 sal_call_set_referer(call->op,params->referer->op);
388 call->referer=linphone_call_ref(params->referer);
393 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
394 LinphoneCall *call=ms_new0(LinphoneCall,1);
397 StunCandidate ac=STUN_CANDIDATE_INIT,vc=STUN_CANDIDATE_INIT;
399 call->dir=LinphoneCallIncoming;
400 sal_op_set_user_pointer(op,call);
404 if (lc->sip_conf.ping_with_options){
405 /*the following sends an option request back to the caller so that
406 we get a chance to discover our nat'd address before answering.*/
407 call->ping_op=sal_op_new(lc->sal);
408 from_str=linphone_address_as_string_uri_only(from);
409 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
410 sal_op_set_user_pointer(call->ping_op,call);
411 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
415 linphone_address_clean(from);
416 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
417 linphone_call_init_common(call, from, to);
418 linphone_core_init_default_params(lc, &call->params);
419 call->params.has_video &= !!lc->video_policy.automatically_accept;
420 switch (linphone_core_get_firewall_policy(call->core)) {
421 case LinphonePolicyUseIce:
422 call->ice_session = ice_session_new();
423 ice_session_set_role(call->ice_session, IR_Controlled);
424 linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
425 if (call->ice_session != NULL) {
426 linphone_call_init_media_streams(call);
427 linphone_call_start_media_streams_for_ice_gathering(call);
428 if (linphone_core_gather_ice_candidates(call->core,call)<0) {
429 /* Ice candidates gathering failed, proceed with the call anyway. */
430 linphone_call_delete_ice_session(call);
431 linphone_call_stop_media_streams(call);
435 case LinphonePolicyUseStun:
436 ping_time=linphone_core_run_stun_tests(call->core,call,&ac, &vc);
437 /* No break to also destroy ice session in this case. */
442 linphone_core_adapt_to_network(lc,ping_time,&call->params);
444 call->localdesc=create_local_media_description(lc,call);
445 update_media_description_from_stun(call->localdesc,&ac,&vc);
446 call->camera_active=call->params.has_video;
448 discover_mtu(lc,linphone_address_get_domain(from));
452 /* this function is called internally to get rid of a call.
453 It performs the following tasks:
454 - remove the call from the internal list of calls
455 - update the call logs accordingly
458 static void linphone_call_set_terminated(LinphoneCall *call){
459 LinphoneCore *lc=call->core;
461 linphone_core_update_allocated_audio_bandwidth(lc);
463 call->owns_call_log=FALSE;
464 linphone_call_log_completed(call);
467 if (call == lc->current_call){
468 ms_message("Resetting the current call");
469 lc->current_call=NULL;
472 if (linphone_core_del_call(lc,call) != 0){
473 ms_error("Could not remove the call from the list !!!");
476 if (ms_list_size(lc->calls)==0)
477 linphone_core_notify_all_friends(lc,lc->presence_mode);
479 linphone_core_conference_check_uninit(lc);
480 if (call->ringing_beep){
481 linphone_core_stop_dtmf(lc);
482 call->ringing_beep=FALSE;
485 linphone_call_unref(call->referer);
490 void linphone_call_fix_call_parameters(LinphoneCall *call){
491 call->params.has_video=call->current_params.has_video;
492 call->params.media_encryption=call->current_params.media_encryption;
495 const char *linphone_call_state_to_string(LinphoneCallState cs){
497 case LinphoneCallIdle:
498 return "LinphoneCallIdle";
499 case LinphoneCallIncomingReceived:
500 return "LinphoneCallIncomingReceived";
501 case LinphoneCallOutgoingInit:
502 return "LinphoneCallOutgoingInit";
503 case LinphoneCallOutgoingProgress:
504 return "LinphoneCallOutgoingProgress";
505 case LinphoneCallOutgoingRinging:
506 return "LinphoneCallOutgoingRinging";
507 case LinphoneCallOutgoingEarlyMedia:
508 return "LinphoneCallOutgoingEarlyMedia";
509 case LinphoneCallConnected:
510 return "LinphoneCallConnected";
511 case LinphoneCallStreamsRunning:
512 return "LinphoneCallStreamsRunning";
513 case LinphoneCallPausing:
514 return "LinphoneCallPausing";
515 case LinphoneCallPaused:
516 return "LinphoneCallPaused";
517 case LinphoneCallResuming:
518 return "LinphoneCallResuming";
519 case LinphoneCallRefered:
520 return "LinphoneCallRefered";
521 case LinphoneCallError:
522 return "LinphoneCallError";
523 case LinphoneCallEnd:
524 return "LinphoneCallEnd";
525 case LinphoneCallPausedByRemote:
526 return "LinphoneCallPausedByRemote";
527 case LinphoneCallUpdatedByRemote:
528 return "LinphoneCallUpdatedByRemote";
529 case LinphoneCallIncomingEarlyMedia:
530 return "LinphoneCallIncomingEarlyMedia";
531 case LinphoneCallUpdated:
532 return "LinphoneCallUpdated";
533 case LinphoneCallReleased:
534 return "LinphoneCallReleased";
536 return "undefined state";
539 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
540 LinphoneCore *lc=call->core;
542 if (call->state!=cstate){
543 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
544 if (cstate!=LinphoneCallReleased){
545 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
546 linphone_call_state_to_string(cstate));
550 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
551 linphone_call_state_to_string(cstate));
552 if (cstate!=LinphoneCallRefered){
553 /*LinphoneCallRefered is rather an event, not a state.
554 Indeed it does not change the state of the call (still paused or running)*/
557 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
558 switch(call->reason){
559 case LinphoneReasonDeclined:
560 call->log->status=LinphoneCallDeclined;
562 case LinphoneReasonNotAnswered:
563 call->log->status=LinphoneCallMissed;
568 linphone_call_set_terminated (call);
570 if (cstate == LinphoneCallConnected) {
571 call->log->status=LinphoneCallSuccess;
572 call->media_start_time=time(NULL);
575 if (lc->vtable.call_state_changed)
576 lc->vtable.call_state_changed(lc,call,cstate,message);
577 if (cstate==LinphoneCallReleased){
578 if (call->op!=NULL) {
579 /* so that we cannot have anymore upcalls for SAL
580 concerning this call*/
581 sal_op_release(call->op);
584 linphone_call_unref(call);
589 static void linphone_call_destroy(LinphoneCall *obj)
592 sal_op_release(obj->op);
595 if (obj->resultdesc!=NULL) {
596 sal_media_description_unref(obj->resultdesc);
597 obj->resultdesc=NULL;
599 if (obj->localdesc!=NULL) {
600 sal_media_description_unref(obj->localdesc);
604 sal_op_release(obj->ping_op);
607 ms_free(obj->refer_to);
609 if (obj->owns_call_log)
610 linphone_call_log_destroy(obj->log);
611 if (obj->auth_token) {
612 ms_free(obj->auth_token);
614 if (obj->ice_session) {
615 ice_session_destroy(obj->ice_session);
622 * @addtogroup call_control
627 * Increments the call 's reference count.
628 * An application that wishes to retain a pointer to call object
629 * must use this function to unsure the pointer remains
630 * valid. Once the application no more needs this pointer,
631 * it must call linphone_call_unref().
633 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
639 * Decrements the call object reference count.
640 * See linphone_call_ref().
642 void linphone_call_unref(LinphoneCall *obj){
645 linphone_call_destroy(obj);
650 * Returns current parameters associated to the call.
652 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
653 return &call->current_params;
656 static bool_t is_video_active(const SalStreamDescription *sd){
657 return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
661 * Returns call parameters proposed by remote.
663 * This is useful when receiving an incoming call, to know whether the remote party
664 * supports video, encryption or whatever.
666 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
667 LinphoneCallParams *cp=&call->remote_params;
668 memset(cp,0,sizeof(*cp));
670 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
672 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
674 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
675 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
676 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
677 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
679 cp->has_video=is_video_active(secure_vsd);
680 if (secure_asd || asd==NULL)
681 cp->media_encryption=LinphoneMediaEncryptionSRTP;
683 cp->has_video=is_video_active(vsd);
692 * Returns the remote address associated to this call
695 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
696 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
700 * Returns the remote address associated to this call as a string.
702 * The result string must be freed by user using ms_free().
704 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
705 return linphone_address_as_string(linphone_call_get_remote_address(call));
709 * Retrieves the call's current state.
711 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
716 * Returns the reason for a call termination (either error or normal termination)
718 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
723 * Get the user_pointer in the LinphoneCall
725 * @ingroup call_control
727 * return user_pointer an opaque user pointer that can be retrieved at any time
729 void *linphone_call_get_user_pointer(LinphoneCall *call)
731 return call->user_pointer;
735 * Set the user_pointer in the LinphoneCall
737 * @ingroup call_control
739 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
741 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
743 call->user_pointer = user_pointer;
747 * Returns the call log associated to this call.
749 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
754 * Returns the refer-to uri (if the call was transfered).
756 const char *linphone_call_get_refer_to(const LinphoneCall *call){
757 return call->refer_to;
761 * Returns direction of the call (incoming or outgoing).
763 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
764 return call->log->dir;
768 * Returns the far end's user agent description string, if available.
770 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
772 return sal_op_get_remote_ua (call->op);
778 * Returns true if this calls has received a transfer that has not been
780 * Pending transfers are executed when this call is being paused or closed,
781 * locally or by remote endpoint.
782 * If the call is already paused while receiving the transfer request, the
783 * transfer immediately occurs.
785 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
786 return call->refer_pending;
790 * Returns call's duration in seconds.
792 int linphone_call_get_duration(const LinphoneCall *call){
793 if (call->media_start_time==0) return 0;
794 return time(NULL)-call->media_start_time;
798 * Returns the call object this call is replacing, if any.
799 * Call replacement can occur during call transfers.
800 * By default, the core automatically terminates the replaced call and accept the new one.
801 * This function allows the application to know whether a new incoming call is a one that replaces another one.
803 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
804 SalOp *op=sal_call_get_replaces(call->op);
806 return (LinphoneCall*)sal_op_get_user_pointer(op);
812 * Indicate whether camera input should be sent to remote end.
814 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
816 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
817 LinphoneCore *lc=call->core;
818 MSWebCam *nowebcam=get_nowebcam_device();
819 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
820 video_stream_change_camera(call->videostream,
821 enable ? lc->video_conf.device : nowebcam);
824 call->camera_active=enable;
829 * Take a photo of currently received video and write it into a jpeg file.
831 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
833 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
834 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
836 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
843 * Returns TRUE if camera pictures are sent to the remote party.
845 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
846 return call->camera_active;
850 * Enable video stream.
852 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
853 cp->has_video=enabled;
856 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
857 return cp->audio_codec;
860 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
861 return cp->video_codec;
865 * Returns whether video is enabled.
867 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
868 return cp->has_video;
871 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
872 return cp->media_encryption;
875 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
876 cp->media_encryption = e;
881 * Enable sending of real early media (during outgoing calls).
883 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
884 cp->real_early_media=enabled;
887 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
888 return cp->real_early_media;
892 * Returns true if the call is part of the locally managed conference.
894 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
895 return cp->in_conference;
899 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
900 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
902 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
903 cp->audio_bw=bandwidth;
908 * Request remote side to send us a Video Fast Update.
910 void linphone_call_send_vfu_request(LinphoneCall *call)
912 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
913 sal_call_send_vfu_request(call->op);
920 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
921 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
922 memcpy(ncp,cp,sizeof(LinphoneCallParams));
929 void linphone_call_params_destroy(LinphoneCallParams *p){
938 #ifdef TEST_EXT_RENDERER
939 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
940 ms_message("rendercb, local buffer=%p, remote buffer=%p",
941 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
946 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
947 LinphoneCall* call = (LinphoneCall*) user_pointer;
948 ms_warning("In linphonecall.c: video_stream_event_cb");
950 case MS_VIDEO_DECODER_DECODING_ERRORS:
951 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
952 linphone_call_send_vfu_request(call);
954 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
955 ms_message("First video frame decoded successfully");
956 if (call->nextVideoFrameDecoded._func != NULL)
957 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
960 ms_warning("Unhandled event %i", event_id);
966 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
967 call->nextVideoFrameDecoded._func = cb;
968 call->nextVideoFrameDecoded._user_data = user_data;
970 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
974 void linphone_call_init_audio_stream(LinphoneCall *call){
975 LinphoneCore *lc=call->core;
976 AudioStream *audiostream;
977 int dscp=lp_config_get_int(lc->config,"rtp","audio_dscp",-1);
979 call->audiostream=audiostream=audio_stream_new(call->audio_port,call->audio_port+1,linphone_core_ipv6_enabled(lc));
981 audio_stream_set_dscp(audiostream,dscp);
982 if (linphone_core_echo_limiter_enabled(lc)){
983 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
984 if (strcasecmp(type,"mic")==0)
985 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
986 else if (strcasecmp(type,"full")==0)
987 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
989 audio_stream_enable_gain_control(audiostream,TRUE);
990 if (linphone_core_echo_cancellation_enabled(lc)){
991 int len,delay,framesize;
992 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
993 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
994 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
995 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
996 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
997 if (statestr && audiostream->ec){
998 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
1001 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
1003 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
1004 audio_stream_enable_noise_gate(audiostream,enabled);
1007 audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
1010 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
1011 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
1012 rtp_session_set_transports(audiostream->session,artp,artcp);
1014 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
1015 rtp_session_set_pktinfo(audiostream->session, TRUE);
1016 rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
1017 audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
1018 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
1021 call->audiostream_app_evq = ortp_ev_queue_new();
1022 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
1025 void linphone_call_init_video_stream(LinphoneCall *call){
1026 #ifdef VIDEO_ENABLED
1027 LinphoneCore *lc=call->core;
1029 if ((lc->video_conf.display || lc->video_conf.capture) && call->params.has_video){
1030 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
1031 int dscp=lp_config_get_int(lc->config,"rtp","video_dscp",-1);
1033 call->videostream=video_stream_new(call->video_port,call->video_port+1,linphone_core_ipv6_enabled(lc));
1035 video_stream_set_dscp(call->videostream,dscp);
1036 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
1037 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
1039 if( lc->video_conf.displaytype != NULL)
1040 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
1041 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
1043 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
1044 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
1045 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
1047 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, 1))){
1048 rtp_session_set_pktinfo(call->videostream->session, TRUE);
1049 rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
1050 call->videostream->ice_check_list = ice_session_check_list(call->ice_session, 1);
1051 ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
1053 call->videostream_app_evq = ortp_ev_queue_new();
1054 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
1055 #ifdef TEST_EXT_RENDERER
1056 video_stream_set_render_callback(call->videostream,rendercb,NULL);
1060 call->videostream=NULL;
1064 void linphone_call_init_media_streams(LinphoneCall *call){
1065 linphone_call_init_audio_stream(call);
1066 linphone_call_init_video_stream(call);
1070 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1072 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
1073 LinphoneCore* lc = (LinphoneCore*)user_data;
1074 if (dtmf<0 || dtmf>15){
1075 ms_warning("Bad dtmf value %i",dtmf);
1078 if (lc->vtable.dtmf_received != NULL)
1079 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1082 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1084 MSFilter *f=st->equalizer;
1085 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1086 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1087 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1093 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1094 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1095 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1104 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1105 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1108 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1109 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1110 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1113 audio_stream_set_mic_gain(st,mic_gain);
1115 audio_stream_set_mic_gain(st,0);
1117 recv_gain = lc->sound_conf.soft_play_lev;
1118 if (recv_gain != 0) {
1119 linphone_core_set_playback_gain_db (lc,recv_gain);
1123 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1124 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1125 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1126 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1127 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1128 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1131 if (speed==-1) speed=0.03;
1132 if (force==-1) force=25;
1133 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1134 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1136 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1138 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1139 if (transmit_thres!=-1)
1140 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1142 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1143 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1146 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1147 float floorgain = 1/mic_gain;
1148 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1149 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1150 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1151 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1153 parametrize_equalizer(lc,st);
1156 static void post_configure_audio_streams(LinphoneCall*call){
1157 AudioStream *st=call->audiostream;
1158 LinphoneCore *lc=call->core;
1159 _post_configure_audio_stream(st,lc,call->audio_muted);
1160 if (lc->vtable.dtmf_received!=NULL){
1161 /* replace by our default action*/
1162 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1163 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1167 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1170 RtpProfile *prof=rtp_profile_new("Call profile");
1173 LinphoneCore *lc=call->core;
1175 const LinphoneCallParams *params=&call->params;
1178 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1179 PayloadType *pt=(PayloadType*)elem->data;
1182 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1183 if (desc->type==SalAudio){
1184 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1185 if (params->up_ptime)
1186 up_ptime=params->up_ptime;
1187 else up_ptime=linphone_core_get_upload_ptime(lc);
1189 *used_pt=payload_type_get_number(pt);
1192 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1193 else if (md->bandwidth>0) {
1194 /*case where b=AS is given globally, not per stream*/
1195 remote_bw=md->bandwidth;
1196 if (desc->type==SalVideo){
1197 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1201 if (desc->type==SalAudio){
1202 int audio_bw=call->audio_bw;
1204 if (params->up_bw< audio_bw)
1205 audio_bw=params->up_bw;
1207 bw=get_min_bandwidth(audio_bw,remote_bw);
1208 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1209 if (bw>0) pt->normal_bitrate=bw*1000;
1210 else if (desc->type==SalAudio){
1211 pt->normal_bitrate=-1;
1214 up_ptime=desc->ptime;
1218 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1219 payload_type_append_send_fmtp(pt,tmp);
1221 number=payload_type_get_number(pt);
1222 if (rtp_profile_get_payload(prof,number)!=NULL){
1223 ms_warning("A payload type with number %i already exists in profile !",number);
1225 rtp_profile_set_payload(prof,number,pt);
1231 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1232 int pause_time=3000;
1233 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1234 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1237 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1239 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1240 LinphoneCore *lc=call->core;
1241 LinphoneCall *current=linphone_core_get_current_call(lc);
1242 return !linphone_core_is_in_conference(lc) &&
1243 (current==NULL || current==call);
1245 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1247 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1248 if (crypto[i].tag == tag) {
1254 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1255 LinphoneCore *lc=call->core;
1257 /* look for savp stream first */
1258 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1259 SalProtoRtpSavp,SalAudio);
1260 /* no savp audio stream, use avp */
1262 stream=sal_media_description_find_stream(call->resultdesc,
1263 SalProtoRtpAvp,SalAudio);
1265 if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1266 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1267 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1268 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1269 const char *playfile=lc->play_file;
1270 const char *recfile=lc->rec_file;
1271 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1275 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1276 if (playcard==NULL) {
1277 ms_warning("No card defined for playback !");
1279 if (captcard==NULL) {
1280 ms_warning("No card defined for capture !");
1282 /*Replace soundcard filters by inactive file players or recorders
1283 when placed in recvonly or sendonly mode*/
1284 if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1287 }else if (stream->dir==SalStreamSendOnly){
1291 /*And we will eventually play "playfile" if set by the user*/
1294 if (send_ringbacktone){
1296 playfile=NULL;/* it is setup later*/
1298 /*if playfile are supplied don't use soundcards*/
1299 if (lc->use_files) {
1303 if (call->params.in_conference){
1304 /* first create the graph without soundcard resources*/
1305 captcard=playcard=NULL;
1307 if (!linphone_call_sound_resources_available(call)){
1308 ms_message("Sound resources are used by another call, not using soundcard.");
1309 captcard=playcard=NULL;
1311 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1312 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1313 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1314 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1315 audio_stream_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
1316 audio_stream_start_full(
1318 call->audio_profile,
1319 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1321 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1322 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1324 linphone_core_get_audio_jittcomp(lc),
1331 post_configure_audio_streams(call);
1332 if (muted && !send_ringbacktone){
1333 audio_stream_set_mic_gain(call->audiostream,0);
1335 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1337 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1339 if (send_ringbacktone){
1340 setup_ring_player(lc,call);
1342 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1344 /* valid local tags are > 0 */
1345 if (stream->proto == SalProtoRtpSavp) {
1346 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1347 SalProtoRtpSavp,SalAudio);
1348 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1350 if (crypto_idx >= 0) {
1351 audio_stream_enable_strp(
1353 stream->crypto[0].algo,
1354 local_st_desc->crypto[crypto_idx].master_key,
1355 stream->crypto[0].master_key);
1356 call->audiostream_encrypted=TRUE;
1358 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1359 call->audiostream_encrypted=FALSE;
1361 }else call->audiostream_encrypted=FALSE;
1362 if (call->params.in_conference){
1363 /*transform the graph to connect it to the conference filter */
1364 bool_t mute=stream->dir==SalStreamRecvOnly;
1365 linphone_call_add_to_conf(call, mute);
1367 call->current_params.in_conference=call->params.in_conference;
1368 }else ms_warning("No audio stream accepted ?");
1372 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1373 #ifdef VIDEO_ENABLED
1374 LinphoneCore *lc=call->core;
1376 /* look for savp stream first */
1377 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1378 SalProtoRtpSavp,SalVideo);
1379 /* no savp audio stream, use avp */
1381 vstream=sal_media_description_find_stream(call->resultdesc,
1382 SalProtoRtpAvp,SalVideo);
1384 /* shutdown preview */
1385 if (lc->previewstream!=NULL) {
1386 video_preview_stop(lc->previewstream);
1387 lc->previewstream=NULL;
1390 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1391 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1392 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1393 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1395 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1396 VideoStreamDir dir=VideoStreamSendRecv;
1397 MSWebCam *cam=lc->video_conf.device;
1398 bool_t is_inactive=FALSE;
1400 call->current_params.has_video=TRUE;
1402 video_stream_enable_adaptive_bitrate_control(call->videostream,
1403 linphone_core_adaptive_rate_control_enabled(lc));
1404 video_stream_enable_adaptive_jittcomp(call->videostream, linphone_core_video_adaptive_jittcomp_enabled(lc));
1405 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1406 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1407 if (lc->video_window_id!=0)
1408 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1409 if (lc->preview_window_id!=0)
1410 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1411 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1413 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1414 cam=get_nowebcam_device();
1415 dir=VideoStreamSendOnly;
1416 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1417 dir=VideoStreamRecvOnly;
1418 }else if (vstream->dir==SalStreamSendRecv){
1419 if (lc->video_conf.display && lc->video_conf.capture)
1420 dir=VideoStreamSendRecv;
1421 else if (lc->video_conf.display)
1422 dir=VideoStreamRecvOnly;
1424 dir=VideoStreamSendOnly;
1426 ms_warning("video stream is inactive.");
1427 /*either inactive or incompatible with local capabilities*/
1430 if (call->camera_active==FALSE || all_inputs_muted){
1431 cam=get_nowebcam_device();
1434 call->log->video_enabled = TRUE;
1435 video_stream_set_direction (call->videostream, dir);
1436 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1437 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1438 video_stream_start(call->videostream,
1439 call->video_profile, rtp_addr, vstream->rtp_port,
1440 rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1441 used_pt, linphone_core_get_video_jittcomp(lc), cam);
1442 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1445 if (vstream->proto == SalProtoRtpSavp) {
1446 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1447 SalProtoRtpSavp,SalVideo);
1449 video_stream_enable_strp(
1451 vstream->crypto[0].algo,
1452 local_st_desc->crypto[0].master_key,
1453 vstream->crypto[0].master_key
1455 call->videostream_encrypted=TRUE;
1457 call->videostream_encrypted=FALSE;
1459 }else ms_warning("No video stream accepted.");
1461 ms_warning("No valid video stream defined.");
1466 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1467 LinphoneCore *lc=call->core;
1469 call->current_params.audio_codec = NULL;
1470 call->current_params.video_codec = NULL;
1472 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1474 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1475 #ifdef VIDEO_ENABLED
1476 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1477 SalProtoRtpAvp,SalVideo);
1480 if ((call->audiostream == NULL) && (call->videostream == NULL)) {
1481 ms_fatal("start_media_stream() called without prior init !");
1484 cname=linphone_address_as_string_uri_only(me);
1486 #if defined(VIDEO_ENABLED)
1487 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1488 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1492 if (call->audiostream!=NULL) {
1493 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1495 call->current_params.has_video=FALSE;
1496 if (call->videostream!=NULL) {
1497 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1500 call->all_muted=all_inputs_muted;
1501 call->playing_ringbacktone=send_ringbacktone;
1502 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1504 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1505 OrtpZrtpParams params;
1506 /*will be set later when zrtp is activated*/
1507 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1509 params.zid_file=lc->zrtp_secrets_cache;
1510 audio_stream_enable_zrtp(call->audiostream,¶ms);
1511 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1512 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1513 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1516 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1517 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1518 linphone_call_fix_call_parameters(call);
1519 if ((call->ice_session != NULL) && (ice_session_state(call->ice_session) != IS_Completed)) {
1520 ice_session_start_connectivity_checks(call->ice_session);
1526 linphone_address_destroy(me);
1529 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1530 audio_stream_prepare_sound(call->audiostream, NULL, NULL);
1531 #ifdef VIDEO_ENABLED
1532 if (call->videostream) {
1533 video_stream_prepare_video(call->videostream);
1538 void linphone_call_delete_ice_session(LinphoneCall *call){
1539 if (call->ice_session != NULL) {
1540 ice_session_destroy(call->ice_session);
1541 call->ice_session = NULL;
1542 if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
1543 if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
1547 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1548 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1549 log->quality=audio_stream_get_average_quality_rating(st);
1552 void linphone_call_stop_media_streams(LinphoneCall *call){
1553 if (call->audiostream!=NULL) {
1554 call->audiostream->ice_check_list = NULL;
1555 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1556 ortp_ev_queue_flush(call->audiostream_app_evq);
1557 ortp_ev_queue_destroy(call->audiostream_app_evq);
1558 call->audiostream_app_evq=NULL;
1560 if (call->audiostream->ec){
1561 const char *state_str=NULL;
1562 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1564 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1565 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1568 linphone_call_log_fill_stats (call->log,call->audiostream);
1569 if (call->endpoint){
1570 linphone_call_remove_from_conf(call);
1572 audio_stream_stop(call->audiostream);
1573 call->audiostream=NULL;
1577 #ifdef VIDEO_ENABLED
1578 if (call->videostream!=NULL){
1579 call->videostream->ice_check_list = NULL;
1580 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1581 ortp_ev_queue_flush(call->videostream_app_evq);
1582 ortp_ev_queue_destroy(call->videostream_app_evq);
1583 call->videostream_app_evq=NULL;
1584 video_stream_stop(call->videostream);
1585 call->videostream=NULL;
1588 ms_event_queue_skip(call->core->msevq);
1590 if (call->audio_profile){
1591 rtp_profile_clear_all(call->audio_profile);
1592 rtp_profile_destroy(call->audio_profile);
1593 call->audio_profile=NULL;
1595 if (call->video_profile){
1596 rtp_profile_clear_all(call->video_profile);
1597 rtp_profile_destroy(call->video_profile);
1598 call->video_profile=NULL;
1604 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1605 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1606 bool_t bypass_mode = !enable;
1607 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1610 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1611 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1613 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1616 return linphone_core_echo_cancellation_enabled(call->core);
1620 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1621 if (call!=NULL && call->audiostream!=NULL ) {
1623 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1624 if (strcasecmp(type,"mic")==0)
1625 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1626 else if (strcasecmp(type,"full")==0)
1627 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1629 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1634 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1635 if (call!=NULL && call->audiostream!=NULL ){
1636 return call->audiostream->el_type !=ELInactive ;
1638 return linphone_core_echo_limiter_enabled(call->core);
1643 * @addtogroup call_misc
1648 * Returns the measured sound volume played locally (received from remote).
1649 * It is expressed in dbm0.
1651 float linphone_call_get_play_volume(LinphoneCall *call){
1652 AudioStream *st=call->audiostream;
1653 if (st && st->volrecv){
1655 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1659 return LINPHONE_VOLUME_DB_LOWEST;
1663 * Returns the measured sound volume recorded locally (sent to remote).
1664 * It is expressed in dbm0.
1666 float linphone_call_get_record_volume(LinphoneCall *call){
1667 AudioStream *st=call->audiostream;
1668 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1670 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1674 return LINPHONE_VOLUME_DB_LOWEST;
1678 * Obtain real-time quality rating of the call
1680 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1681 * during all the duration of the call. This function returns its value at the time of the function call.
1682 * It is expected that the rating is updated at least every 5 seconds or so.
1683 * The rating is a floating point number comprised between 0 and 5.
1685 * 4-5 = good quality <br>
1686 * 3-4 = average quality <br>
1687 * 2-3 = poor quality <br>
1688 * 1-2 = very poor quality <br>
1689 * 0-1 = can't be worse, mostly unusable <br>
1691 * @returns The function returns -1 if no quality measurement is available, for example if no
1692 * active audio stream exist. Otherwise it returns the quality rating.
1694 float linphone_call_get_current_quality(LinphoneCall *call){
1695 if (call->audiostream){
1696 return audio_stream_get_quality_rating(call->audiostream);
1702 * Returns call quality averaged over all the duration of the call.
1704 * See linphone_call_get_current_quality() for more details about quality measurement.
1706 float linphone_call_get_average_quality(LinphoneCall *call){
1707 if (call->audiostream){
1708 return audio_stream_get_average_quality_rating(call->audiostream);
1714 * Access last known statistics for audio stream, for a given call.
1716 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1717 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1721 * Access last known statistics for video stream, for a given call.
1723 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1724 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1732 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1733 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1734 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1735 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1736 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1737 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1740 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1744 from = linphone_call_get_remote_address_as_string(call);
1747 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1752 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1754 if (lc->vtable.display_warning!=NULL)
1755 lc->vtable.display_warning(lc,temp);
1756 linphone_core_terminate_call(lc,call);
1759 static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
1760 OrtpEventType evt=ortp_event_get_type(ev);
1761 OrtpEventData *evd=ortp_event_get_data(ev);
1763 if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1764 switch (ice_session_state(call->ice_session)) {
1766 if (ice_session_role(call->ice_session) == IR_Controlling) {
1767 ice_session_select_candidates(call->ice_session);
1768 linphone_core_update_call(call->core, call, &call->current_params);
1772 if (ice_session_has_completed_check_list(call->ice_session) == TRUE) {
1773 if (ice_session_role(call->ice_session) == IR_Controlling) {
1774 /* At least one ICE session has succeeded, so perform a call update. */
1775 ice_session_select_candidates(call->ice_session);
1776 linphone_core_update_call(call->core, call, &call->current_params);
1783 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1785 if (evd->info.ice_processing_successful==TRUE) {
1786 ice_session_compute_candidates_foundations(call->ice_session);
1787 ice_session_eliminate_redundant_candidates(call->ice_session);
1788 ice_session_choose_default_candidates(call->ice_session);
1789 ping_time = ice_session_gathering_duration(call->ice_session);
1790 if (ping_time >=0) {
1791 ping_time /= ice_session_nb_check_lists(call->ice_session);
1794 ms_warning("No STUN answer from [%s], disabling ICE",linphone_core_get_stun_server(call->core));
1795 linphone_call_delete_ice_session(call);
1797 switch (call->state) {
1798 case LinphoneCallStreamsRunning:
1799 linphone_core_start_update_call(call->core, call);
1801 case LinphoneCallUpdatedByRemote:
1802 linphone_core_start_accept_call_update(call->core, call);
1804 case LinphoneCallOutgoingInit:
1805 if (ping_time >= 0) {
1806 linphone_core_adapt_to_network(call->core, ping_time, &call->params);
1808 linphone_call_stop_media_streams(call);
1809 linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
1812 if (ping_time >= 0) {
1813 linphone_core_adapt_to_network(call->core, ping_time, &call->params);
1815 linphone_call_stop_media_streams(call);
1816 linphone_core_notify_incoming_call(call->core, call);
1819 } else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
1820 linphone_core_start_accept_call_update(call->core, call);
1821 } else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
1822 ice_session_restart(call->ice_session);
1823 ice_session_set_role(call->ice_session, IR_Controlling);
1824 linphone_core_update_call(call->core, call, &call->current_params);
1828 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1829 LinphoneCore* lc = call->core;
1830 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1831 bool_t disconnected=FALSE;
1833 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1834 RtpSession *as=NULL,*vs=NULL;
1835 float audio_load=0, video_load=0;
1836 if (call->audiostream!=NULL){
1837 as=call->audiostream->session;
1838 if (call->audiostream->ticker)
1839 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1841 if (call->videostream!=NULL){
1842 if (call->videostream->ticker)
1843 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1844 vs=call->videostream->session;
1846 display_bandwidth(as,vs);
1847 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1849 #ifdef VIDEO_ENABLED
1850 if (call->videostream!=NULL) {
1853 /* Ensure there is no dangling ICE check list. */
1854 if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
1856 // Beware that the application queue should not depend on treatments fron the
1857 // mediastreamer queue.
1858 video_stream_iterate(call->videostream);
1860 while (call->videostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq)))){
1861 OrtpEventType evt=ortp_event_get_type(ev);
1862 OrtpEventData *evd=ortp_event_get_data(ev);
1863 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1864 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1865 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1866 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1867 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1868 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1869 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1871 if (lc->vtable.call_stats_updated)
1872 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1873 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1874 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1875 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1876 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1877 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1879 if (lc->vtable.call_stats_updated)
1880 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1881 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1882 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1883 handle_ice_events(call, ev);
1885 ortp_event_destroy(ev);
1889 if (call->audiostream!=NULL) {
1892 /* Ensure there is no dangling ICE check list. */
1893 if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
1895 // Beware that the application queue should not depend on treatments fron the
1896 // mediastreamer queue.
1897 audio_stream_iterate(call->audiostream);
1899 while (call->audiostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq)))){
1900 OrtpEventType evt=ortp_event_get_type(ev);
1901 OrtpEventData *evd=ortp_event_get_data(ev);
1902 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1903 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1904 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1905 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1906 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1907 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1908 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1909 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1910 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1912 if (lc->vtable.call_stats_updated)
1913 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1914 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1915 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1916 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1917 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1918 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1920 if (lc->vtable.call_stats_updated)
1921 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1922 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1923 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1924 handle_ice_events(call, ev);
1926 ortp_event_destroy(ev);
1929 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1930 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1932 linphone_core_disconnected(call->core,call);
1935 void linphone_call_log_completed(LinphoneCall *call){
1936 LinphoneCore *lc=call->core;
1938 call->log->duration=time(NULL)-call->start_time;
1940 if (call->log->status==LinphoneCallMissed){
1943 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1944 "You have missed %i calls.", lc->missed_calls),
1946 if (lc->vtable.display_status!=NULL)
1947 lc->vtable.display_status(lc,info);
1950 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1951 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1952 MSList *elem,*prevelem=NULL;
1953 /*find the last element*/
1954 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1958 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1959 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1961 if (lc->vtable.call_log_updated!=NULL){
1962 lc->vtable.call_log_updated(lc,call->log);
1964 call_logs_write_to_config_file(lc);
1967 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1968 return call->transfer_state;
1971 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1972 if (state != call->transfer_state) {
1973 LinphoneCore* lc = call->core;
1974 call->transfer_state = state;
1975 if (lc->vtable.transfer_state_changed)
1976 lc->vtable.transfer_state_changed(lc, call, state);
1980 bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
1981 return call->params.in_conference;