]> sjero.net Git - linphone/blob - coreapi/linphonecall.c
Asynchronous ICE candidates gathering.
[linphone] / coreapi / linphonecall.c
1
2 /*
3 linphone
4 Copyright (C) 2010  Belledonne Communications SARL
5  (simon.morlat@linphone.org)
6
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.
11
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.
16
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.
20 */
21 #ifdef WIN32
22 #include <time.h>
23 #endif
24 #include "linphonecore.h"
25 #include "sipsetup.h"
26 #include "lpconfig.h"
27 #include "private.h"
28 #include <ortp/event.h>
29 #include <ortp/b64.h>
30
31
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"
39
40 #ifdef VIDEO_ENABLED
41 static MSWebCam *get_nowebcam_device(){
42         return ms_web_cam_manager_get_cam(ms_web_cam_manager_get(),"StaticImage: Static picture");
43 }
44 #endif
45
46 static bool_t generate_b64_crypto_key(int key_length, char* key_out) {
47         int b64_size;
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");
51                 free(tmp);
52                 return FALSE;
53         }
54         
55         b64_size = b64_encode((const char*)tmp, key_length, NULL, 0);
56         if (b64_size == 0) {
57                 ms_error("Failed to b64 encode key");
58                 free(tmp);
59                 return FALSE;
60         }
61         key_out[b64_size] = '\0';
62         b64_encode((const char*)tmp, key_length, key_out, 40);
63         free(tmp);
64         return TRUE;
65 }
66
67 LinphoneCore *linphone_call_get_core(const LinphoneCall *call){
68         return call->core;
69 }
70
71 const char* linphone_call_get_authentication_token(LinphoneCall *call){
72         return call->auth_token;
73 }
74
75 bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
76         return call->auth_token_verified;
77 }
78
79 static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
80         // Check ZRTP encryption in audiostream
81         if (!call->audiostream_encrypted) {
82                 return FALSE;
83         }
84
85 #ifdef VIDEO_ENABLED
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) {
89                 return FALSE;
90         }
91 #endif
92
93         return TRUE;
94 }
95
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);
103         } else {
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);
108         }
109 }
110
111 #ifdef VIDEO_ENABLED
112 static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
113         ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
114
115         LinphoneCall *call = (LinphoneCall *)data;
116         call->videostream_encrypted=encrypted;
117         propagate_encryption_changed(call);
118 }
119 #endif
120
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");
124
125         LinphoneCall *call = (LinphoneCall *)data;
126         call->audiostream_encrypted=encrypted;
127         
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);
131         }
132
133         propagate_encryption_changed(call);
134
135
136 #ifdef VIDEO_ENABLED
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,&params);
144         }
145 #endif
146 }
147
148
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);
153
154         call->auth_token=ms_strdup(auth_token);
155         call->auth_token_verified=verified;
156
157         ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
158 }
159
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");
163         }
164         if (call->audiostream->ortpZrtpContext==NULL){
165                 ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
166         }
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);
171         }
172         call->auth_token_verified=verified;
173         propagate_encryption_changed(call);
174 }
175
176 static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
177         MSList *l=NULL;
178         const MSList *it;
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);
185                                 continue;
186                         }
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);
190                         }
191                 }
192         }
193         return l;
194 }
195
196 static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
197         MSList *l;
198         PayloadType *pt;
199         int i;
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();
204         IceSession *ice_session=sal_op_get_ice_session(call->op);
205
206         md->session_id=session_id;
207         md->session_ver=session_ver;
208         md->nstreams=1;
209         strncpy(md->addr,call->localip,sizeof(md->addr));
210         strncpy(md->username,username,sizeof(md->username));
211         md->bandwidth=linphone_core_get_download_bandwidth(lc);
212
213         /*set audio capabilities */
214         strncpy(md->streams[0].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
215         strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
216         md->streams[0].rtp_port=call->audio_port;
217         md->streams[0].rtcp_port=call->audio_port+1;
218         md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ? 
219                 SalProtoRtpSavp : SalProtoRtpAvp;
220         md->streams[0].type=SalAudio;
221         md->streams[0].ptime=lc->net_conf.down_ptime;
222         l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
223         pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
224         l=ms_list_append(l,pt);
225         md->streams[0].payloads=l;
226         
227
228
229         if (call->params.has_video){
230                 md->nstreams++;
231                 md->streams[1].rtp_port=call->video_port;
232                 md->streams[1].rtcp_port=call->video_port+1;
233                 md->streams[1].proto=md->streams[0].proto;
234                 md->streams[1].type=SalVideo;
235                 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
236                 md->streams[1].payloads=l;
237         }
238         
239         for(i=0; i<md->nstreams; i++) {
240                 if (md->streams[i].proto == SalProtoRtpSavp) {
241                         md->streams[i].crypto[0].tag = 1;
242                         md->streams[i].crypto[0].algo = AES_128_SHA1_80;
243                         if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
244                                 md->streams[i].crypto[0].algo = 0;
245                         md->streams[i].crypto[1].tag = 2;
246                         md->streams[i].crypto[1].algo = AES_128_SHA1_32;
247                         if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
248                                 md->streams[i].crypto[1].algo = 0;
249                         md->streams[i].crypto[2].algo = 0;
250                 }
251                 if ((call->dir == LinphoneCallOutgoing) && (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (ice_session != NULL)){
252                         ice_session_add_check_list(ice_session, ice_check_list_new());
253                 }
254         }
255         
256         linphone_address_destroy(addr);
257         return md;
258 }
259
260 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
261         SalMediaDescription *md=call->localdesc;
262         if (md== NULL) {
263                 call->localdesc = create_local_media_description(lc,call);
264         } else {
265                 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
266                 sal_media_description_unref(md);
267         }
268 }
269
270 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
271         unsigned int id=rand() & 0xfff;
272         return _create_local_media_description(lc,call,id,id);
273 }
274
275 static int find_port_offset(LinphoneCore *lc){
276         int offset;
277         MSList *elem;
278         int audio_port;
279         bool_t already_used=FALSE;
280         for(offset=0;offset<100;offset+=2){
281                 audio_port=linphone_core_get_audio_port (lc)+offset;
282                 already_used=FALSE;
283                 for(elem=lc->calls;elem!=NULL;elem=elem->next){
284                         LinphoneCall *call=(LinphoneCall*)elem->data;
285                         if (call->audio_port==audio_port) {
286                                 already_used=TRUE;
287                                 break;
288                         }
289                 }
290                 if (!already_used) break;
291         }
292         if (offset==100){
293                 ms_error("Could not find any free port !");
294                 return -1;
295         }
296         return offset;
297 }
298
299 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
300         int port_offset;
301         call->magic=linphone_call_magic;
302         call->refcnt=1;
303         call->state=LinphoneCallIdle;
304         call->transfer_state = LinphoneCallIdle;
305         call->start_time=time(NULL);
306         call->media_start_time=0;
307         call->log=linphone_call_log_new(call, from, to);
308         call->owns_call_log=TRUE;
309         linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
310         port_offset=find_port_offset (call->core);
311         if (port_offset==-1) return;
312         call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
313         call->video_port=linphone_core_get_video_port(call->core)+port_offset;
314         linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
315         linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
316 }
317
318 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
319         stats->type = type;
320         stats->received_rtcp = NULL;
321         stats->sent_rtcp = NULL;
322 }
323
324 static void discover_mtu(LinphoneCore *lc, const char *remote){
325         int mtu;
326         if (lc->net_conf.mtu==0 ){
327                 /*attempt to discover mtu*/
328                 mtu=ms_discover_mtu(remote);
329                 if (mtu>0){
330                         ms_set_mtu(mtu);
331                         ms_message("Discovered mtu is %i, RTP payload max size is %i",
332                                 mtu, ms_get_payload_max_size());
333                 }
334         }
335 }
336
337 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
338 {
339         LinphoneCall *call=ms_new0(LinphoneCall,1);
340         call->dir=LinphoneCallOutgoing;
341         call->op=sal_op_new(lc->sal);
342         sal_op_set_user_pointer(call->op,call);
343         call->core=lc;
344         linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
345         linphone_call_init_common(call,from,to);
346         call->params=*params;
347         if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
348                 sal_op_set_ice_session(call->op, ice_session_new());
349                 ice_session_set_role(sal_op_get_ice_session(call->op), IR_Controlling);
350         }
351         call->localdesc=create_local_media_description (lc,call);
352         call->camera_active=params->has_video;
353         if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
354                 linphone_core_run_stun_tests(call->core,call);
355         }
356         discover_mtu(lc,linphone_address_get_domain (to));
357         if (params->referer){
358                 sal_call_set_referer(call->op,params->referer->op);
359                 call->referer=linphone_call_ref(params->referer);
360         }
361         return call;
362 }
363
364 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
365         LinphoneCall *call=ms_new0(LinphoneCall,1);
366         char *from_str;
367
368         call->dir=LinphoneCallIncoming;
369         sal_op_set_user_pointer(op,call);
370         call->op=op;
371         call->core=lc;
372
373         if (lc->sip_conf.ping_with_options){
374                 /*the following sends an option request back to the caller so that
375                  we get a chance to discover our nat'd address before answering.*/
376                 call->ping_op=sal_op_new(lc->sal);
377                 from_str=linphone_address_as_string_uri_only(from);
378                 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
379                 sal_op_set_user_pointer(call->ping_op,call);
380                 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
381                 ms_free(from_str);
382         }
383
384         linphone_address_clean(from);
385         linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
386         linphone_call_init_common(call, from, to);
387         linphone_core_init_default_params(lc, &call->params);
388         call->params.has_video &= !!lc->video_policy.automatically_accept;
389         call->localdesc=create_local_media_description (lc,call);
390         call->camera_active=call->params.has_video;
391         switch (linphone_core_get_firewall_policy(call->core)) {
392                 case LinphonePolicyUseIce:
393                         linphone_core_gather_ice_candidates(call->core, call);
394                         break;
395                 case LinphonePolicyUseStun:
396                         linphone_core_run_stun_tests(call->core,call);
397                         /* No break to also destroy ice session in this case. */
398                 default:
399                         if (sal_op_get_ice_session(call->op) != NULL) {
400                                 ice_session_destroy(sal_op_get_ice_session(call->op));
401                                 sal_op_set_ice_session(call->op, NULL);
402                         }
403                         break;
404         }
405         discover_mtu(lc,linphone_address_get_domain(from));
406         return call;
407 }
408
409 /* this function is called internally to get rid of a call.
410  It performs the following tasks:
411  - remove the call from the internal list of calls
412  - update the call logs accordingly
413 */
414
415 static void linphone_call_set_terminated(LinphoneCall *call){
416         LinphoneCore *lc=call->core;
417
418         linphone_core_update_allocated_audio_bandwidth(lc);
419
420         call->owns_call_log=FALSE;
421         linphone_call_log_completed(call);
422
423
424         if (call == lc->current_call){
425                 ms_message("Resetting the current call");
426                 lc->current_call=NULL;
427         }
428
429         if (linphone_core_del_call(lc,call) != 0){
430                 ms_error("Could not remove the call from the list !!!");
431         }
432
433         if (ms_list_size(lc->calls)==0)
434                 linphone_core_notify_all_friends(lc,lc->presence_mode);
435
436         linphone_core_conference_check_uninit(lc);
437         if (call->ringing_beep){
438                 linphone_core_stop_dtmf(lc);
439                 call->ringing_beep=FALSE;
440         }
441         if (call->referer){
442                 linphone_call_unref(call->referer);
443                 call->referer=NULL;
444         }
445 }
446
447 void linphone_call_fix_call_parameters(LinphoneCall *call){
448         call->params.has_video=call->current_params.has_video;
449         call->params.media_encryption=call->current_params.media_encryption;
450 }
451
452 const char *linphone_call_state_to_string(LinphoneCallState cs){
453         switch (cs){
454                 case LinphoneCallIdle:
455                         return "LinphoneCallIdle";
456                 case LinphoneCallIncomingReceived:
457                         return "LinphoneCallIncomingReceived";
458                 case LinphoneCallOutgoingInit:
459                         return "LinphoneCallOutgoingInit";
460                 case LinphoneCallOutgoingProgress:
461                         return "LinphoneCallOutgoingProgress";
462                 case LinphoneCallOutgoingRinging:
463                         return "LinphoneCallOutgoingRinging";
464                 case LinphoneCallOutgoingEarlyMedia:
465                         return "LinphoneCallOutgoingEarlyMedia";
466                 case LinphoneCallConnected:
467                         return "LinphoneCallConnected";
468                 case LinphoneCallStreamsRunning:
469                         return "LinphoneCallStreamsRunning";
470                 case LinphoneCallPausing:
471                         return "LinphoneCallPausing";
472                 case LinphoneCallPaused:
473                         return "LinphoneCallPaused";
474                 case LinphoneCallResuming:
475                         return "LinphoneCallResuming";
476                 case LinphoneCallRefered:
477                         return "LinphoneCallRefered";
478                 case LinphoneCallError:
479                         return "LinphoneCallError";
480                 case LinphoneCallEnd:
481                         return "LinphoneCallEnd";
482                 case LinphoneCallPausedByRemote:
483                         return "LinphoneCallPausedByRemote";
484                 case LinphoneCallUpdatedByRemote:
485                         return "LinphoneCallUpdatedByRemote";
486                 case LinphoneCallIncomingEarlyMedia:
487                         return "LinphoneCallIncomingEarlyMedia";
488                 case LinphoneCallUpdated:
489                         return "LinphoneCallUpdated";
490                 case LinphoneCallReleased:
491                         return "LinphoneCallReleased";
492         }
493         return "undefined state";
494 }
495
496 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
497         LinphoneCore *lc=call->core;
498
499         if (call->state!=cstate){
500                 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
501                         if (cstate!=LinphoneCallReleased){
502                                 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
503                                    linphone_call_state_to_string(cstate));
504                                 return;
505                         }
506                 }
507                 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
508                            linphone_call_state_to_string(cstate));
509                 if (cstate!=LinphoneCallRefered){
510                         /*LinphoneCallRefered is rather an event, not a state.
511                          Indeed it does not change the state of the call (still paused or running)*/
512                         call->state=cstate;
513                 }
514                 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
515              switch(call->reason){
516                                 case LinphoneReasonDeclined:
517                                         call->log->status=LinphoneCallDeclined;
518                                 case LinphoneReasonNotAnswered:
519                                         call->log->status=LinphoneCallMissed;
520                                 break;
521                                 default:
522                                 break;
523                         }
524                         linphone_call_set_terminated (call);
525                 }
526                 if (cstate == LinphoneCallConnected) {
527                         call->log->status=LinphoneCallSuccess;
528                         call->media_start_time=time(NULL);
529                 }
530
531                 if (lc->vtable.call_state_changed)
532                         lc->vtable.call_state_changed(lc,call,cstate,message);
533                 if (cstate==LinphoneCallReleased){
534                         if (call->op!=NULL) {
535                                 /* so that we cannot have anymore upcalls for SAL
536                                  concerning this call*/
537                                 sal_op_release(call->op);
538                                 call->op=NULL;
539                         }
540                         linphone_call_unref(call);
541                 }
542         }
543 }
544
545 static void linphone_call_destroy(LinphoneCall *obj)
546 {
547         if (obj->op!=NULL) {
548                 sal_op_release(obj->op);
549                 obj->op=NULL;
550         }
551         if (obj->resultdesc!=NULL) {
552                 sal_media_description_unref(obj->resultdesc);
553                 obj->resultdesc=NULL;
554         }
555         if (obj->localdesc!=NULL) {
556                 sal_media_description_unref(obj->localdesc);
557                 obj->localdesc=NULL;
558         }
559         if (obj->ping_op) {
560                 sal_op_release(obj->ping_op);
561         }
562         if (obj->refer_to){
563                 ms_free(obj->refer_to);
564         }
565         if (obj->owns_call_log)
566                 linphone_call_log_destroy(obj->log);
567         if (obj->auth_token) {
568                 ms_free(obj->auth_token);
569         }
570
571         ms_free(obj);
572 }
573
574 /**
575  * @addtogroup call_control
576  * @{
577 **/
578
579 /**
580  * Increments the call 's reference count.
581  * An application that wishes to retain a pointer to call object
582  * must use this function to unsure the pointer remains
583  * valid. Once the application no more needs this pointer,
584  * it must call linphone_call_unref().
585 **/
586 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
587         obj->refcnt++;
588         return obj;
589 }
590
591 /**
592  * Decrements the call object reference count.
593  * See linphone_call_ref().
594 **/
595 void linphone_call_unref(LinphoneCall *obj){
596         obj->refcnt--;
597         if (obj->refcnt==0){
598                 linphone_call_destroy(obj);
599         }
600 }
601
602 /**
603  * Returns current parameters associated to the call.
604 **/
605 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
606         return &call->current_params;
607 }
608
609 static bool_t is_video_active(const SalStreamDescription *sd){
610         return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
611 }
612
613 /**
614  * Returns call parameters proposed by remote.
615  * 
616  * This is useful when receiving an incoming call, to know whether the remote party
617  * supports video, encryption or whatever.
618 **/
619 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
620         LinphoneCallParams *cp=&call->remote_params;
621         memset(cp,0,sizeof(*cp));
622         if (call->op){
623                 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
624                 if (md){
625                         SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
626
627                         asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
628                         vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
629                         secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
630                         secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
631                         if (secure_vsd){
632                                 cp->has_video=is_video_active(secure_vsd);
633                                 if (secure_asd || asd==NULL)
634                                         cp->media_encryption=LinphoneMediaEncryptionSRTP;
635                         }else if (vsd){
636                                 cp->has_video=is_video_active(vsd);
637                         }
638                         return cp;
639                 }
640         }
641         return NULL;
642 }
643
644 /**
645  * Returns the remote address associated to this call
646  *
647 **/
648 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
649         return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
650 }
651
652 /**
653  * Returns the remote address associated to this call as a string.
654  *
655  * The result string must be freed by user using ms_free().
656 **/
657 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
658         return linphone_address_as_string(linphone_call_get_remote_address(call));
659 }
660
661 /**
662  * Retrieves the call's current state.
663 **/
664 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
665         return call->state;
666 }
667
668 /**
669  * Returns the reason for a call termination (either error or normal termination)
670 **/
671 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
672         return call->reason;
673 }
674
675 /**
676  * Get the user_pointer in the LinphoneCall
677  *
678  * @ingroup call_control
679  *
680  * return user_pointer an opaque user pointer that can be retrieved at any time
681 **/
682 void *linphone_call_get_user_pointer(LinphoneCall *call)
683 {
684         return call->user_pointer;
685 }
686
687 /**
688  * Set the user_pointer in the LinphoneCall
689  *
690  * @ingroup call_control
691  *
692  * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
693 **/
694 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
695 {
696         call->user_pointer = user_pointer;
697 }
698
699 /**
700  * Returns the call log associated to this call.
701 **/
702 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
703         return call->log;
704 }
705
706 /**
707  * Returns the refer-to uri (if the call was transfered).
708 **/
709 const char *linphone_call_get_refer_to(const LinphoneCall *call){
710         return call->refer_to;
711 }
712
713 /**
714  * Returns direction of the call (incoming or outgoing).
715 **/
716 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
717         return call->log->dir;
718 }
719
720 /**
721  * Returns the far end's user agent description string, if available.
722 **/
723 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
724         if (call->op){
725                 return sal_op_get_remote_ua (call->op);
726         }
727         return NULL;
728 }
729
730 /**
731  * Returns true if this calls has received a transfer that has not been
732  * executed yet.
733  * Pending transfers are executed when this call is being paused or closed,
734  * locally or by remote endpoint.
735  * If the call is already paused while receiving the transfer request, the
736  * transfer immediately occurs.
737 **/
738 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
739         return call->refer_pending;
740 }
741
742 /**
743  * Returns call's duration in seconds.
744 **/
745 int linphone_call_get_duration(const LinphoneCall *call){
746         if (call->media_start_time==0) return 0;
747         return time(NULL)-call->media_start_time;
748 }
749
750 /**
751  * Returns the call object this call is replacing, if any.
752  * Call replacement can occur during call transfers.
753  * By default, the core automatically terminates the replaced call and accept the new one.
754  * This function allows the application to know whether a new incoming call is a one that replaces another one.
755 **/
756 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
757         SalOp *op=sal_call_get_replaces(call->op);
758         if (op){
759                 return (LinphoneCall*)sal_op_get_user_pointer(op);
760         }
761         return NULL;
762 }
763
764 /**
765  * Indicate whether camera input should be sent to remote end.
766 **/
767 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
768 #ifdef VIDEO_ENABLED
769         if (call->videostream!=NULL && call->videostream->ticker!=NULL){
770                 LinphoneCore *lc=call->core;
771                 MSWebCam *nowebcam=get_nowebcam_device();
772                 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
773                         video_stream_change_camera(call->videostream,
774                                      enable ? lc->video_conf.device : nowebcam);
775                 }
776         }
777         call->camera_active=enable;
778 #endif
779 }
780
781 /**
782  * Take a photo of currently received video and write it into a jpeg file.
783 **/
784 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
785 #ifdef VIDEO_ENABLED
786         if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
787                 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
788         }
789         ms_warning("Cannot take snapshot: no currently running video stream on this call.");
790         return -1;
791 #endif
792         return -1;
793 }
794
795 /**
796  * Returns TRUE if camera pictures are sent to the remote party.
797 **/
798 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
799         return call->camera_active;
800 }
801
802 /**
803  * Enable video stream.
804 **/
805 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
806         cp->has_video=enabled;
807 }
808
809 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
810         return cp->audio_codec;
811 }
812
813 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
814         return cp->video_codec;
815 }
816
817 /**
818  * Returns whether video is enabled.
819 **/
820 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
821         return cp->has_video;
822 }
823
824 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
825         return cp->media_encryption;
826 }
827
828 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
829         cp->media_encryption = e;
830 }
831
832
833 /**
834  * Enable sending of real early media (during outgoing calls).
835 **/
836 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
837         cp->real_early_media=enabled;
838 }
839
840 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
841         return cp->real_early_media;
842 }
843
844 /**
845  * Returns true if the call is part of the locally managed conference.
846 **/
847 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
848         return cp->in_conference;
849 }
850
851 /**
852  * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
853  * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
854 **/
855 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
856         cp->audio_bw=bandwidth;
857 }
858
859 #ifdef VIDEO_ENABLED
860 /**
861  * Request remote side to send us a Video Fast Update.
862 **/
863 void linphone_call_send_vfu_request(LinphoneCall *call)
864 {
865         if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
866                 sal_call_send_vfu_request(call->op);
867 }
868 #endif
869
870 /**
871  *
872 **/
873 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
874         LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
875         memcpy(ncp,cp,sizeof(LinphoneCallParams));
876         return ncp;
877 }
878
879 /**
880  *
881 **/
882 void linphone_call_params_destroy(LinphoneCallParams *p){
883         ms_free(p);
884 }
885
886 /**
887  * @}
888 **/
889
890
891 #ifdef TEST_EXT_RENDERER
892 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
893         ms_message("rendercb, local buffer=%p, remote buffer=%p",
894                    local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
895 }
896 #endif
897
898 #ifdef VIDEO_ENABLED
899 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
900     LinphoneCall* call = (LinphoneCall*) user_pointer;
901         ms_warning("In linphonecall.c: video_stream_event_cb");
902         switch (event_id) {
903                 case MS_VIDEO_DECODER_DECODING_ERRORS:
904                         ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
905                         linphone_call_send_vfu_request(call);
906                         break;
907         case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
908             ms_message("First video frame decoded successfully");
909             if (call->nextVideoFrameDecoded._func != NULL)
910                 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
911             break;
912                 default:
913                         ms_warning("Unhandled event %i", event_id);
914                         break;
915         }
916 }
917 #endif
918
919 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
920     call->nextVideoFrameDecoded._func = cb;
921     call->nextVideoFrameDecoded._user_data = user_data;
922 #ifdef VIDEO_ENABLED
923     ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
924 #endif
925 }
926
927 void linphone_call_init_media_streams(LinphoneCall *call){
928         LinphoneCore *lc=call->core;
929         SalMediaDescription *md=call->localdesc;
930         AudioStream *audiostream;
931         IceSession *ice_session = sal_op_get_ice_session(call->op);
932
933         call->audiostream=audiostream=audio_stream_new(md->streams[0].rtp_port,md->streams[0].rtcp_port,linphone_core_ipv6_enabled(lc));
934         if (linphone_core_echo_limiter_enabled(lc)){
935                 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
936                 if (strcasecmp(type,"mic")==0)
937                         audio_stream_enable_echo_limiter(audiostream,ELControlMic);
938                 else if (strcasecmp(type,"full")==0)
939                         audio_stream_enable_echo_limiter(audiostream,ELControlFull);
940         }
941         audio_stream_enable_gain_control(audiostream,TRUE);
942         if (linphone_core_echo_cancellation_enabled(lc)){
943                 int len,delay,framesize;
944                 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
945                 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
946                 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
947                 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
948                 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
949                 if (statestr && audiostream->ec){
950                         ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
951                 }
952         }
953         audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
954         {
955                 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
956                 audio_stream_enable_noise_gate(audiostream,enabled);
957         }
958
959         if (lc->rtptf){
960                 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
961                 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
962                 rtp_session_set_transports(audiostream->session,artp,artcp);
963         }
964         if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (ice_session != NULL)){
965                 rtp_session_set_pktinfo(audiostream->session, TRUE);
966                 audiostream->ice_check_list = ice_session_check_list(ice_session, 0);
967                 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
968         }
969
970         call->audiostream_app_evq = ortp_ev_queue_new();
971         rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
972
973 #ifdef VIDEO_ENABLED
974
975         if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].rtp_port>0){
976                 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
977                 call->videostream=video_stream_new(md->streams[1].rtp_port,md->streams[1].rtcp_port,linphone_core_ipv6_enabled(lc));
978                 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
979                 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
980           
981                 if( lc->video_conf.displaytype != NULL)
982                         video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
983                 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
984                 if (lc->rtptf){
985                         RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
986                         RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
987                         rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
988                 }
989                 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (ice_session != NULL)){
990                         rtp_session_set_pktinfo(call->videostream->session, TRUE);
991                         call->videostream->ice_check_list = ice_session_check_list(ice_session, 1);
992                         ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
993                 }
994                 call->videostream_app_evq = ortp_ev_queue_new();
995                 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
996 #ifdef TEST_EXT_RENDERER
997                 video_stream_set_render_callback(call->videostream,rendercb,NULL);
998 #endif
999         }
1000 #else
1001         call->videostream=NULL;
1002 #endif
1003 }
1004
1005
1006 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1007
1008 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
1009         LinphoneCore* lc = (LinphoneCore*)user_data;
1010         if (dtmf<0 || dtmf>15){
1011                 ms_warning("Bad dtmf value %i",dtmf);
1012                 return;
1013         }
1014         if (lc->vtable.dtmf_received != NULL)
1015                 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1016 }
1017
1018 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1019         if (st->equalizer){
1020                 MSFilter *f=st->equalizer;
1021                 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1022                 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1023                 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1024                 if (enabled){
1025                         if (gains){
1026                                 do{
1027                                         int bytes;
1028                                         MSEqualizerGain g;
1029                                         if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1030                                                 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1031                                                 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1032                                                 gains+=bytes;
1033                                         }else break;
1034                                 }while(1);
1035                         }
1036                 }
1037         }
1038 }
1039
1040 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1041         float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1042         float thres = 0;
1043         float recv_gain;
1044         float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1045         float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1046         int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1047
1048         if (!muted)
1049                 audio_stream_set_mic_gain(st,mic_gain);
1050         else
1051                 audio_stream_set_mic_gain(st,0);
1052
1053         recv_gain = lc->sound_conf.soft_play_lev;
1054         if (recv_gain != 0) {
1055                 linphone_core_set_playback_gain_db (lc,recv_gain);
1056         }
1057         
1058         if (st->volsend){
1059                 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1060                 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1061                 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1062                 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1063                 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1064                 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1065                 MSFilter *f=NULL;
1066                 f=st->volsend;
1067                 if (speed==-1) speed=0.03;
1068                 if (force==-1) force=25;
1069                 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1070                 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1071                 if (thres!=-1)
1072                         ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1073                 if (sustain!=-1)
1074                         ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1075                 if (transmit_thres!=-1)
1076                                 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1077
1078                 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1079                 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1080         }
1081         if (st->volrecv){
1082                 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1083                 float floorgain = 1/mic_gain;
1084                 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1085                 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1086                 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1087                 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1088         }
1089         parametrize_equalizer(lc,st);
1090 }
1091
1092 static void post_configure_audio_streams(LinphoneCall*call){
1093         AudioStream *st=call->audiostream;
1094         LinphoneCore *lc=call->core;
1095         _post_configure_audio_stream(st,lc,call->audio_muted);
1096         if (lc->vtable.dtmf_received!=NULL){
1097                 /* replace by our default action*/
1098                 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1099                 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1100         }
1101 }
1102
1103 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1104         int bw;
1105         const MSList *elem;
1106         RtpProfile *prof=rtp_profile_new("Call profile");
1107         bool_t first=TRUE;
1108         int remote_bw=0;
1109         LinphoneCore *lc=call->core;
1110         int up_ptime=0;
1111         *used_pt=-1;
1112
1113         for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1114                 PayloadType *pt=(PayloadType*)elem->data;
1115                 int number;
1116
1117                 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1118                         if (desc->type==SalAudio){
1119                                 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1120                                 up_ptime=linphone_core_get_upload_ptime(lc);
1121                         }
1122                         *used_pt=payload_type_get_number(pt);
1123                         first=FALSE;
1124                 }
1125                 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1126                 else if (md->bandwidth>0) {
1127                         /*case where b=AS is given globally, not per stream*/
1128                         remote_bw=md->bandwidth;
1129                         if (desc->type==SalVideo){
1130                                 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1131                         }
1132                 }
1133
1134                 if (desc->type==SalAudio){
1135                                 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1136                 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1137                 if (bw>0) pt->normal_bitrate=bw*1000;
1138                 else if (desc->type==SalAudio){
1139                         pt->normal_bitrate=-1;
1140                 }
1141                 if (desc->ptime>0){
1142                         up_ptime=desc->ptime;
1143                 }
1144                 if (up_ptime>0){
1145                         char tmp[40];
1146                         snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1147                         payload_type_append_send_fmtp(pt,tmp);
1148                 }
1149                 number=payload_type_get_number(pt);
1150                 if (rtp_profile_get_payload(prof,number)!=NULL){
1151                         ms_warning("A payload type with number %i already exists in profile !",number);
1152                 }else
1153                         rtp_profile_set_payload(prof,number,pt);
1154         }
1155         return prof;
1156 }
1157
1158
1159 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1160         int pause_time=3000;
1161         audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1162         ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1163 }
1164
1165 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1166
1167 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1168         LinphoneCore *lc=call->core;
1169         LinphoneCall *current=linphone_core_get_current_call(lc);
1170         return !linphone_core_is_in_conference(lc) && 
1171                 (current==NULL || current==call);
1172 }
1173 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1174     int i;
1175     for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1176         if (crypto[i].tag == tag) {
1177             return i;
1178         }
1179     }
1180     return -1;
1181 }
1182 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1183         LinphoneCore *lc=call->core;
1184         int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1185         int used_pt=-1;
1186         /* look for savp stream first */
1187         const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1188                                                 SalProtoRtpSavp,SalAudio);
1189         /* no savp audio stream, use avp */
1190         if (!stream)
1191                 stream=sal_media_description_find_stream(call->resultdesc,
1192                                                 SalProtoRtpAvp,SalAudio);
1193
1194         if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1195                 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1196                         lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1197                 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1198                 const char *playfile=lc->play_file;
1199                 const char *recfile=lc->rec_file;
1200                 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1201                 bool_t use_ec;
1202
1203                 if (used_pt!=-1){
1204                         call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1205                         if (playcard==NULL) {
1206                                 ms_warning("No card defined for playback !");
1207                         }
1208                         if (captcard==NULL) {
1209                                 ms_warning("No card defined for capture !");
1210                         }
1211                         /*Replace soundcard filters by inactive file players or recorders
1212                          when placed in recvonly or sendonly mode*/
1213                         if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1214                                 captcard=NULL;
1215                                 playfile=NULL;
1216                         }else if (stream->dir==SalStreamSendOnly){
1217                                 playcard=NULL;
1218                                 captcard=NULL;
1219                                 recfile=NULL;
1220                                 /*And we will eventually play "playfile" if set by the user*/
1221                                 /*playfile=NULL;*/
1222                         }
1223                         if (send_ringbacktone){
1224                                 captcard=NULL;
1225                                 playfile=NULL;/* it is setup later*/
1226                         }
1227                         /*if playfile are supplied don't use soundcards*/
1228                         if (lc->use_files) {
1229                                 captcard=NULL;
1230                                 playcard=NULL;
1231                         }
1232                         if (call->params.in_conference){
1233                                 /* first create the graph without soundcard resources*/
1234                                 captcard=playcard=NULL;
1235                         }
1236                         if (!linphone_call_sound_resources_available(call)){
1237                                 ms_message("Sound resources are used by another call, not using soundcard.");
1238                                 captcard=playcard=NULL;
1239                         }
1240                         use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1241                         if (playcard &&  stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1242                         if (captcard &&  stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1243                         audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1244                         audio_stream_start_full(
1245                                 call->audiostream,
1246                                 call->audio_profile,
1247                                 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1248                                 stream->rtp_port,
1249                                 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1250                                 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1251                                 used_pt,
1252                                 jitt_comp,
1253                                 playfile,
1254                                 recfile,
1255                                 playcard,
1256                                 captcard,
1257                                 use_ec
1258                                 );
1259                         post_configure_audio_streams(call);
1260                         if (muted && !send_ringbacktone){
1261                                 audio_stream_set_mic_gain(call->audiostream,0);
1262                         }
1263                         if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1264                                 int pause_time=500;
1265                                 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1266                         }
1267                         if (send_ringbacktone){
1268                                 setup_ring_player(lc,call);
1269                         }
1270                         audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1271                         
1272             /* valid local tags are > 0 */
1273                         if (stream->proto == SalProtoRtpSavp) {
1274                 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1275                                                                                             SalProtoRtpSavp,SalAudio);
1276                 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1277                 
1278                 if (crypto_idx >= 0) {
1279                     audio_stream_enable_strp(
1280                                              call->audiostream, 
1281                                              stream->crypto[0].algo,
1282                                              local_st_desc->crypto[crypto_idx].master_key,
1283                                              stream->crypto[0].master_key);
1284                     call->audiostream_encrypted=TRUE;
1285                 } else {
1286                     ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1287                     call->audiostream_encrypted=FALSE;
1288                 }
1289                         }else call->audiostream_encrypted=FALSE;
1290                         if (call->params.in_conference){
1291                                 /*transform the graph to connect it to the conference filter */
1292                                 bool_t mute=stream->dir==SalStreamRecvOnly;
1293                                 linphone_call_add_to_conf(call, mute);
1294                         }
1295                         call->current_params.in_conference=call->params.in_conference;
1296                 }else ms_warning("No audio stream accepted ?");
1297         }
1298 }
1299
1300 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1301 #ifdef VIDEO_ENABLED
1302         LinphoneCore *lc=call->core;
1303         int used_pt=-1;
1304         /* look for savp stream first */
1305         const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1306                                                 SalProtoRtpSavp,SalVideo);
1307         /* no savp audio stream, use avp */
1308         if (!vstream)
1309                 vstream=sal_media_description_find_stream(call->resultdesc,
1310                                                 SalProtoRtpAvp,SalVideo);
1311                                                 
1312         /* shutdown preview */
1313         if (lc->previewstream!=NULL) {
1314                 video_preview_stop(lc->previewstream);
1315                 lc->previewstream=NULL;
1316         }
1317         
1318         if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1319                 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1320                 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1321                 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1322                 if (used_pt!=-1){
1323                         call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1324                         VideoStreamDir dir=VideoStreamSendRecv;
1325                         MSWebCam *cam=lc->video_conf.device;
1326                         bool_t is_inactive=FALSE;
1327
1328                         call->current_params.has_video=TRUE;
1329
1330                         video_stream_enable_adaptive_bitrate_control(call->videostream,
1331                                                                   linphone_core_adaptive_rate_control_enabled(lc));
1332                         video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1333                         video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1334                         if (lc->video_window_id!=0)
1335                                 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1336                         if (lc->preview_window_id!=0)
1337                                 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1338                         video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1339                         
1340                         if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1341                                 cam=get_nowebcam_device();
1342                                 dir=VideoStreamSendOnly;
1343                         }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1344                                 dir=VideoStreamRecvOnly;
1345                         }else if (vstream->dir==SalStreamSendRecv){
1346                                 if (lc->video_conf.display && lc->video_conf.capture)
1347                                         dir=VideoStreamSendRecv;
1348                                 else if (lc->video_conf.display)
1349                                         dir=VideoStreamRecvOnly;
1350                                 else
1351                                         dir=VideoStreamSendOnly;
1352                         }else{
1353                                 ms_warning("video stream is inactive.");
1354                                 /*either inactive or incompatible with local capabilities*/
1355                                 is_inactive=TRUE;
1356                         }
1357                         if (call->camera_active==FALSE || all_inputs_muted){
1358                                 cam=get_nowebcam_device();
1359                         }
1360                         if (!is_inactive){
1361                 call->log->video_enabled = TRUE;
1362                                 video_stream_set_direction (call->videostream, dir);
1363                                 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1364                                 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1365                                 video_stream_start(call->videostream,
1366                                         call->video_profile, rtp_addr, vstream->rtp_port,
1367                                         rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1368                                         used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1369                                 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1370                         }
1371                         
1372                         if (vstream->proto == SalProtoRtpSavp) {
1373                                 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1374                                                 SalProtoRtpSavp,SalVideo);
1375                                                 
1376                                 video_stream_enable_strp(
1377                                         call->videostream, 
1378                                         vstream->crypto[0].algo,
1379                                         local_st_desc->crypto[0].master_key, 
1380                                         vstream->crypto[0].master_key
1381                                         );
1382                                 call->videostream_encrypted=TRUE;
1383                         }else{
1384                                 call->videostream_encrypted=FALSE;
1385                         }
1386                 }else ms_warning("No video stream accepted.");
1387         }else{
1388                 ms_warning("No valid video stream defined.");
1389         }
1390 #endif
1391 }
1392
1393 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1394         LinphoneCore *lc=call->core;
1395
1396         call->current_params.audio_codec = NULL;
1397         call->current_params.video_codec = NULL;
1398
1399         LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1400         char *cname;
1401         bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1402 #ifdef VIDEO_ENABLED
1403         const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1404                                                         SalProtoRtpAvp,SalVideo);
1405 #endif
1406
1407         if(call->audiostream == NULL)
1408         {
1409                 ms_fatal("start_media_stream() called without prior init !");
1410                 return;
1411         }
1412         cname=linphone_address_as_string_uri_only(me);
1413
1414 #if defined(VIDEO_ENABLED)
1415         if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1416                 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1417                 use_arc=FALSE;
1418         }
1419 #endif
1420         linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1421         call->current_params.has_video=FALSE;
1422         if (call->videostream!=NULL) {
1423                 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1424         }
1425
1426         call->all_muted=all_inputs_muted;
1427         call->playing_ringbacktone=send_ringbacktone;
1428         call->up_bw=linphone_core_get_upload_bandwidth(lc);
1429
1430         if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1431                 OrtpZrtpParams params;
1432                 /*will be set later when zrtp is activated*/
1433                 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1434                 
1435                 params.zid_file=lc->zrtp_secrets_cache;
1436                 audio_stream_enable_zrtp(call->audiostream,&params);
1437         }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1438                 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1439                         LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1440         }
1441
1442         /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1443          * further in the call, for example during pause,resume, conferencing reINVITEs*/
1444         linphone_call_fix_call_parameters(call);
1445         if ((sal_op_get_ice_session(call->op) != NULL) && (ice_session_state(sal_op_get_ice_session(call->op)) != IS_Completed)) {
1446                 ice_session_start_connectivity_checks(sal_op_get_ice_session(call->op));
1447         }
1448
1449         goto end;
1450         end:
1451                 ms_free(cname);
1452                 linphone_address_destroy(me);
1453 }
1454
1455 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1456         audio_stream_start_ice_gathering(call->audiostream);
1457         if (call->videostream) {
1458                 video_stream_start_ice_gathering(call->videostream);
1459         }
1460 }
1461
1462 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1463         audio_stream_get_local_rtp_stats (st,&log->local_stats);
1464         log->quality=audio_stream_get_average_quality_rating(st);
1465 }
1466
1467 void linphone_call_stop_media_streams(LinphoneCall *call){
1468         if (call->audiostream!=NULL) {
1469                 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1470                 ortp_ev_queue_flush(call->audiostream_app_evq);
1471                 ortp_ev_queue_destroy(call->audiostream_app_evq);
1472
1473                 if (call->audiostream->ec){
1474                         const char *state_str=NULL;
1475                         ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1476                         if (state_str){
1477                                 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1478                                 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1479                         }
1480                 }
1481                 linphone_call_log_fill_stats (call->log,call->audiostream);
1482                 if (call->endpoint){
1483                         linphone_call_remove_from_conf(call);
1484                 }
1485                 audio_stream_stop(call->audiostream);
1486                 call->audiostream=NULL;
1487         }
1488
1489
1490 #ifdef VIDEO_ENABLED
1491         if (call->videostream!=NULL){
1492                 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1493                 ortp_ev_queue_flush(call->videostream_app_evq);
1494                 ortp_ev_queue_destroy(call->videostream_app_evq);
1495                 video_stream_stop(call->videostream);
1496                 call->videostream=NULL;
1497         }
1498 #endif
1499         ms_event_queue_skip(call->core->msevq);
1500         
1501         if (call->audio_profile){
1502                 rtp_profile_clear_all(call->audio_profile);
1503                 rtp_profile_destroy(call->audio_profile);
1504                 call->audio_profile=NULL;
1505         }
1506         if (call->video_profile){
1507                 rtp_profile_clear_all(call->video_profile);
1508                 rtp_profile_destroy(call->video_profile);
1509                 call->video_profile=NULL;
1510         }
1511 }
1512
1513
1514
1515 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1516         if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1517                 bool_t bypass_mode = !enable;
1518                 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1519         }
1520 }
1521 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1522         if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1523                 bool_t val;
1524                 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1525                 return !val;
1526         } else {
1527                 return linphone_core_echo_cancellation_enabled(call->core);
1528         }
1529 }
1530
1531 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1532         if (call!=NULL && call->audiostream!=NULL ) {
1533                 if (val) {
1534                 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1535                 if (strcasecmp(type,"mic")==0)
1536                         audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1537                 else if (strcasecmp(type,"full")==0)
1538                         audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1539                 } else {
1540                         audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1541                 }
1542         }
1543 }
1544
1545 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1546         if (call!=NULL && call->audiostream!=NULL ){
1547                 return call->audiostream->el_type !=ELInactive ;
1548         } else {
1549                 return linphone_core_echo_limiter_enabled(call->core);
1550         }
1551 }
1552
1553 /**
1554  * @addtogroup call_misc
1555  * @{
1556 **/
1557
1558 /**
1559  * Returns the measured sound volume played locally (received from remote).
1560  * It is expressed in dbm0.
1561 **/
1562 float linphone_call_get_play_volume(LinphoneCall *call){
1563         AudioStream *st=call->audiostream;
1564         if (st && st->volrecv){
1565                 float vol=0;
1566                 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1567                 return vol;
1568
1569         }
1570         return LINPHONE_VOLUME_DB_LOWEST;
1571 }
1572
1573 /**
1574  * Returns the measured sound volume recorded locally (sent to remote).
1575  * It is expressed in dbm0.
1576 **/
1577 float linphone_call_get_record_volume(LinphoneCall *call){
1578         AudioStream *st=call->audiostream;
1579         if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1580                 float vol=0;
1581                 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1582                 return vol;
1583
1584         }
1585         return LINPHONE_VOLUME_DB_LOWEST;
1586 }
1587
1588 /**
1589  * Obtain real-time quality rating of the call
1590  *
1591  * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1592  * during all the duration of the call. This function returns its value at the time of the function call.
1593  * It is expected that the rating is updated at least every 5 seconds or so.
1594  * The rating is a floating point number comprised between 0 and 5.
1595  *
1596  * 4-5 = good quality <br>
1597  * 3-4 = average quality <br>
1598  * 2-3 = poor quality <br>
1599  * 1-2 = very poor quality <br>
1600  * 0-1 = can't be worse, mostly unusable <br>
1601  *
1602  * @returns The function returns -1 if no quality measurement is available, for example if no
1603  * active audio stream exist. Otherwise it returns the quality rating.
1604 **/
1605 float linphone_call_get_current_quality(LinphoneCall *call){
1606         if (call->audiostream){
1607                 return audio_stream_get_quality_rating(call->audiostream);
1608         }
1609         return -1;
1610 }
1611
1612 /**
1613  * Returns call quality averaged over all the duration of the call.
1614  *
1615  * See linphone_call_get_current_quality() for more details about quality measurement.
1616 **/
1617 float linphone_call_get_average_quality(LinphoneCall *call){
1618         if (call->audiostream){
1619                 return audio_stream_get_average_quality_rating(call->audiostream);
1620         }
1621         return -1;
1622 }
1623
1624 /**
1625  * Access last known statistics for audio stream, for a given call.
1626 **/
1627 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1628         return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1629 }
1630
1631 /**
1632  * Access last known statistics for video stream, for a given call.
1633 **/
1634 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1635         return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1636 }
1637
1638
1639 /**
1640  * @}
1641 **/
1642
1643 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1644         ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1645         (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1646         (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1647         (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1648         (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1649 }
1650
1651 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1652         char temp[256];
1653         char *from=NULL;
1654         if(call)
1655                 from = linphone_call_get_remote_address_as_string(call);
1656         if (from)
1657         {
1658                 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1659                 free(from);
1660         }
1661         else
1662         {
1663                 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1664         }
1665         if (lc->vtable.display_warning!=NULL)
1666                 lc->vtable.display_warning(lc,temp);
1667         linphone_core_terminate_call(lc,call);
1668 }
1669
1670 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1671         LinphoneCore* lc = call->core;
1672         int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1673         bool_t disconnected=FALSE;
1674
1675         if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1676                 RtpSession *as=NULL,*vs=NULL;
1677                 float audio_load=0, video_load=0;
1678                 if (call->audiostream!=NULL){
1679                         as=call->audiostream->session;
1680                         if (call->audiostream->ticker)
1681                                 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1682                 }
1683                 if (call->videostream!=NULL){
1684                         if (call->videostream->ticker)
1685                                 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1686                         vs=call->videostream->session;
1687                 }
1688                 display_bandwidth(as,vs);
1689                 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1690         }
1691 #ifdef VIDEO_ENABLED
1692         if (call->videostream!=NULL) {
1693                 // Beware that the application queue should not depend on treatments fron the
1694                 // mediastreamer queue.
1695                 video_stream_iterate(call->videostream);
1696
1697                 if (call->videostream_app_evq){
1698                         OrtpEvent *ev;
1699                         while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1700                                 OrtpEventType evt=ortp_event_get_type(ev);
1701                                 OrtpEventData *evd=ortp_event_get_data(ev);
1702                                 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1703                                         linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1704                                 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1705                                         call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1706                                         if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1707                                                 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1708                                         call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1709                                         evd->packet = NULL;
1710                                         if (lc->vtable.call_stats_updated)
1711                                                 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1712                                 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1713                                         memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1714                                         if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1715                                                 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1716                                         call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1717                                         evd->packet = NULL;
1718                                         if (lc->vtable.call_stats_updated)
1719                                                 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1720                                 } else if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1721                                         if (ice_session_role(sal_op_get_ice_session(call->op)) == IR_Controlling) {
1722                                                 linphone_core_update_call(lc, call, &call->current_params);
1723                                         }
1724                                 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1725                                         if (call->state==LinphoneCallOutgoingInit) {
1726                                                 linphone_call_stop_media_streams(call);
1727                                                 if (evd->info.ice_processing_successful==FALSE) {
1728                                                         ice_session_destroy(sal_op_get_ice_session(call->op));
1729                                                         sal_op_set_ice_session(call->op, NULL);
1730                                                 }
1731                                                 linphone_core_start_invite(call->core,call,NULL);
1732                                         }
1733                                 }
1734                                 ortp_event_destroy(ev);
1735                         }
1736                 }
1737         }
1738 #endif
1739         if (call->audiostream!=NULL) {
1740                 // Beware that the application queue should not depend on treatments fron the
1741                 // mediastreamer queue.
1742                 audio_stream_iterate(call->audiostream);
1743
1744                 if (call->audiostream_app_evq){
1745                         OrtpEvent *ev;
1746                         while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1747                                 OrtpEventType evt=ortp_event_get_type(ev);
1748                                 OrtpEventData *evd=ortp_event_get_data(ev);
1749                                 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1750                                         linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1751                                 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1752                                         linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1753                                 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1754                                         call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1755                                         if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1756                                                 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1757                                         call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1758                                         evd->packet = NULL;
1759                                         if (lc->vtable.call_stats_updated)
1760                                                 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1761                                 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1762                                         memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1763                                         if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1764                                                 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1765                                         call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1766                                         evd->packet = NULL;
1767                                         if (lc->vtable.call_stats_updated)
1768                                                 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1769                                 } else if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1770                                         if (ice_session_role(sal_op_get_ice_session(call->op)) == IR_Controlling) {
1771                                                 linphone_core_update_call(lc, call, &call->current_params);
1772                                         }
1773                                 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1774                                         if (call->state==LinphoneCallOutgoingInit) {
1775                                                 linphone_call_stop_media_streams(call);
1776                                                 if (evd->info.ice_processing_successful==FALSE) {
1777                                                         ice_session_destroy(sal_op_get_ice_session(call->op));
1778                                                         sal_op_set_ice_session(call->op, NULL);
1779                                                 }
1780                                                 linphone_core_start_invite(call->core,call,NULL);
1781                                         }
1782                                 }
1783                                 ortp_event_destroy(ev);
1784                         }
1785                 }
1786         }
1787         if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1788                 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1789         if (disconnected)
1790                 linphone_core_disconnected(call->core,call);
1791 }
1792
1793 void linphone_call_log_completed(LinphoneCall *call){
1794         LinphoneCore *lc=call->core;
1795
1796         call->log->duration=time(NULL)-call->start_time;
1797
1798         if (call->log->status==LinphoneCallMissed){
1799                 char *info;
1800                 lc->missed_calls++;
1801                 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1802                                          "You have missed %i calls.", lc->missed_calls),
1803                                 lc->missed_calls);
1804         if (lc->vtable.display_status!=NULL)
1805             lc->vtable.display_status(lc,info);
1806                 ms_free(info);
1807         }
1808         lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1809         if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1810                 MSList *elem,*prevelem=NULL;
1811                 /*find the last element*/
1812                 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1813                         prevelem=elem;
1814                 }
1815                 elem=prevelem;
1816                 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1817                 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1818         }
1819         if (lc->vtable.call_log_updated!=NULL){
1820                 lc->vtable.call_log_updated(lc,call->log);
1821         }
1822         call_logs_write_to_config_file(lc);
1823 }
1824
1825 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1826         return call->transfer_state;
1827 }
1828
1829 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1830         if (state != call->transfer_state) {
1831                 LinphoneCore* lc = call->core;
1832                 call->transfer_state = state;
1833                 if (lc->vtable.transfer_state_changed)
1834                         lc->vtable.transfer_state_changed(lc, call, state);
1835         }
1836 }
1837