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 LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
72 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
75 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
76 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
79 const char* linphone_call_get_authentication_token(LinphoneCall *call){
80 return call->auth_token;
83 bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
84 return call->auth_token_verified;
87 static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
88 // Check ZRTP encryption in audiostream
89 if (!call->audiostream_encrypted) {
94 // If video enabled, check ZRTP encryption in videostream
95 const LinphoneCallParams *params=linphone_call_get_current_params(call);
96 if (params->has_video && !call->videostream_encrypted) {
104 void propagate_encryption_changed(LinphoneCall *call){
105 LinphoneCore *lc=call->core;
106 if (!linphone_call_are_all_streams_encrypted(call)) {
107 ms_message("Some streams are not encrypted");
108 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
109 if (lc->vtable.call_encryption_changed)
110 lc->vtable.call_encryption_changed(call->core, call, FALSE, call->auth_token);
112 ms_message("All streams are encrypted");
113 call->current_params.media_encryption=LinphoneMediaEncryptionZRTP;
114 if (lc->vtable.call_encryption_changed)
115 lc->vtable.call_encryption_changed(call->core, call, TRUE, call->auth_token);
120 static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
121 ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
123 LinphoneCall *call = (LinphoneCall *)data;
124 call->videostream_encrypted=encrypted;
125 propagate_encryption_changed(call);
129 static void linphone_call_audiostream_encryption_changed(void *data, bool_t encrypted) {
130 char status[255]={0};
131 ms_message("Audio stream is %s ", encrypted ? "encrypted" : "not encrypted");
133 LinphoneCall *call = (LinphoneCall *)data;
134 call->audiostream_encrypted=encrypted;
136 if (encrypted && call->core->vtable.display_status != NULL) {
137 snprintf(status,sizeof(status)-1,_("Authentication token is %s"),call->auth_token);
138 call->core->vtable.display_status(call->core, status);
141 propagate_encryption_changed(call);
145 // Enable video encryption
146 const LinphoneCallParams *params=linphone_call_get_current_params(call);
147 if (params->has_video) {
148 ms_message("Trying to enable encryption on video stream");
149 OrtpZrtpParams params;
150 params.zid_file=NULL; //unused
151 video_stream_enable_zrtp(call->videostream,call->audiostream,¶ms);
157 static void linphone_call_audiostream_auth_token_ready(void *data, const char* auth_token, bool_t verified) {
158 LinphoneCall *call=(LinphoneCall *)data;
159 if (call->auth_token != NULL)
160 ms_free(call->auth_token);
162 call->auth_token=ms_strdup(auth_token);
163 call->auth_token_verified=verified;
165 ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
168 void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t verified){
169 if (call->audiostream==NULL){
170 ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
172 if (call->audiostream->ortpZrtpContext==NULL){
173 ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
175 if (!call->auth_token_verified && verified){
176 ortp_zrtp_sas_verified(call->audiostream->ortpZrtpContext);
177 }else if (call->auth_token_verified && !verified){
178 ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
180 call->auth_token_verified=verified;
181 propagate_encryption_changed(call);
184 static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
187 if (max_sample_rate) *max_sample_rate=0;
188 for(it=codecs;it!=NULL;it=it->next){
189 PayloadType *pt=(PayloadType*)it->data;
190 if (pt->flags & PAYLOAD_TYPE_ENABLED){
191 if (bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,bandwidth_limit)){
192 ms_message("Codec %s/%i eliminated because of audio bandwidth constraint.",pt->mime_type,pt->clock_rate);
195 if (linphone_core_check_payload_type_usability(lc,pt)){
196 l=ms_list_append(l,payload_type_clone(pt));
197 if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
204 static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
208 const char *me=linphone_core_get_identity(lc);
209 LinphoneAddress *addr=linphone_address_new(me);
210 const char *username=linphone_address_get_username (addr);
211 SalMediaDescription *md=sal_media_description_new();
214 md->session_id=session_id;
215 md->session_ver=session_ver;
217 strncpy(md->addr,call->localip,sizeof(md->addr));
218 strncpy(md->username,username,sizeof(md->username));
219 md->bandwidth=linphone_core_get_download_bandwidth(lc);
221 /*set audio capabilities */
222 strncpy(md->streams[0].addr,call->localip,sizeof(md->streams[0].addr));
223 md->streams[0].port=call->audio_port;
224 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
225 SalProtoRtpSavp : SalProtoRtpAvp;
226 md->streams[0].type=SalAudio;
227 md->streams[0].ptime=lc->net_conf.down_ptime;
228 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
229 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
230 l=ms_list_append(l,pt);
231 md->streams[0].payloads=l;
235 if (call->params.has_video){
237 md->streams[1].port=call->video_port;
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;
258 linphone_address_destroy(addr);
262 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
263 SalMediaDescription *md=call->localdesc;
265 call->localdesc = create_local_media_description(lc,call);
267 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
268 sal_media_description_unref(md);
272 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
273 unsigned int id=rand() & 0xfff;
274 return _create_local_media_description(lc,call,id,id);
277 static int find_port_offset(LinphoneCore *lc){
281 bool_t already_used=FALSE;
282 for(offset=0;offset<100;offset+=2){
283 audio_port=linphone_core_get_audio_port (lc)+offset;
285 for(elem=lc->calls;elem!=NULL;elem=elem->next){
286 LinphoneCall *call=(LinphoneCall*)elem->data;
287 if (call->audio_port==audio_port) {
292 if (!already_used) break;
295 ms_error("Could not find any free port !");
301 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
303 call->magic=linphone_call_magic;
305 call->state=LinphoneCallIdle;
306 call->transfer_state = LinphoneCallIdle;
307 call->start_time=time(NULL);
308 call->media_start_time=0;
309 call->log=linphone_call_log_new(call, from, to);
310 call->owns_call_log=TRUE;
311 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
312 port_offset=find_port_offset (call->core);
313 if (port_offset==-1) return;
314 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
315 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
316 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
317 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
320 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
322 stats->received_rtcp = NULL;
323 stats->sent_rtcp = NULL;
326 static void discover_mtu(LinphoneCore *lc, const char *remote){
328 if (lc->net_conf.mtu==0 ){
329 /*attempt to discover mtu*/
330 mtu=ms_discover_mtu(remote);
333 ms_message("Discovered mtu is %i, RTP payload max size is %i",
334 mtu, ms_get_payload_max_size());
339 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
341 LinphoneCall *call=ms_new0(LinphoneCall,1);
342 call->dir=LinphoneCallOutgoing;
343 call->op=sal_op_new(lc->sal);
344 sal_op_set_user_pointer(call->op,call);
346 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
347 linphone_call_init_common(call,from,to);
348 call->params=*params;
349 call->localdesc=create_local_media_description (lc,call);
350 call->camera_active=params->has_video;
351 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
352 linphone_core_run_stun_tests(call->core,call);
353 discover_mtu(lc,linphone_address_get_domain (to));
354 if (params->referer){
355 sal_call_set_referer(call->op,params->referer->op);
356 call->referer=linphone_call_ref(params->referer);
361 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
362 LinphoneCall *call=ms_new0(LinphoneCall,1);
365 call->dir=LinphoneCallIncoming;
366 sal_op_set_user_pointer(op,call);
370 if (lc->sip_conf.ping_with_options){
371 /*the following sends an option request back to the caller so that
372 we get a chance to discover our nat'd address before answering.*/
373 call->ping_op=sal_op_new(lc->sal);
374 from_str=linphone_address_as_string_uri_only(from);
375 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
376 sal_op_set_user_pointer(call->ping_op,call);
377 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
381 linphone_address_clean(from);
382 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
383 linphone_call_init_common(call, from, to);
384 linphone_core_init_default_params(lc, &call->params);
385 call->params.has_video &= !!lc->video_policy.automatically_accept;
386 call->localdesc=create_local_media_description (lc,call);
387 call->camera_active=call->params.has_video;
388 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
389 linphone_core_run_stun_tests(call->core,call);
390 discover_mtu(lc,linphone_address_get_domain(from));
394 /* this function is called internally to get rid of a call.
395 It performs the following tasks:
396 - remove the call from the internal list of calls
397 - update the call logs accordingly
400 static void linphone_call_set_terminated(LinphoneCall *call){
401 LinphoneCore *lc=call->core;
403 linphone_core_update_allocated_audio_bandwidth(lc);
405 call->owns_call_log=FALSE;
406 linphone_call_log_completed(call);
409 if (call == lc->current_call){
410 ms_message("Resetting the current call");
411 lc->current_call=NULL;
414 if (linphone_core_del_call(lc,call) != 0){
415 ms_error("Could not remove the call from the list !!!");
418 if (ms_list_size(lc->calls)==0)
419 linphone_core_notify_all_friends(lc,lc->presence_mode);
421 linphone_core_conference_check_uninit(lc);
422 if (call->ringing_beep){
423 linphone_core_stop_dtmf(lc);
424 call->ringing_beep=FALSE;
427 linphone_call_unref(call->referer);
432 void linphone_call_fix_call_parameters(LinphoneCall *call){
433 call->params.has_video=call->current_params.has_video;
434 call->params.media_encryption=call->current_params.media_encryption;
437 const char *linphone_call_state_to_string(LinphoneCallState cs){
439 case LinphoneCallIdle:
440 return "LinphoneCallIdle";
441 case LinphoneCallIncomingReceived:
442 return "LinphoneCallIncomingReceived";
443 case LinphoneCallOutgoingInit:
444 return "LinphoneCallOutgoingInit";
445 case LinphoneCallOutgoingProgress:
446 return "LinphoneCallOutgoingProgress";
447 case LinphoneCallOutgoingRinging:
448 return "LinphoneCallOutgoingRinging";
449 case LinphoneCallOutgoingEarlyMedia:
450 return "LinphoneCallOutgoingEarlyMedia";
451 case LinphoneCallConnected:
452 return "LinphoneCallConnected";
453 case LinphoneCallStreamsRunning:
454 return "LinphoneCallStreamsRunning";
455 case LinphoneCallPausing:
456 return "LinphoneCallPausing";
457 case LinphoneCallPaused:
458 return "LinphoneCallPaused";
459 case LinphoneCallResuming:
460 return "LinphoneCallResuming";
461 case LinphoneCallRefered:
462 return "LinphoneCallRefered";
463 case LinphoneCallError:
464 return "LinphoneCallError";
465 case LinphoneCallEnd:
466 return "LinphoneCallEnd";
467 case LinphoneCallPausedByRemote:
468 return "LinphoneCallPausedByRemote";
469 case LinphoneCallUpdatedByRemote:
470 return "LinphoneCallUpdatedByRemote";
471 case LinphoneCallIncomingEarlyMedia:
472 return "LinphoneCallIncomingEarlyMedia";
473 case LinphoneCallUpdated:
474 return "LinphoneCallUpdated";
475 case LinphoneCallReleased:
476 return "LinphoneCallReleased";
478 return "undefined state";
481 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
482 LinphoneCore *lc=call->core;
484 if (call->state!=cstate){
485 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
486 if (cstate!=LinphoneCallReleased){
487 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
488 linphone_call_state_to_string(cstate));
492 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
493 linphone_call_state_to_string(cstate));
494 if (cstate!=LinphoneCallRefered){
495 /*LinphoneCallRefered is rather an event, not a state.
496 Indeed it does not change the state of the call (still paused or running)*/
499 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
500 if (call->reason==LinphoneReasonDeclined){
501 call->log->status=LinphoneCallDeclined;
503 linphone_call_set_terminated (call);
505 if (cstate == LinphoneCallConnected) {
506 call->log->status=LinphoneCallSuccess;
507 call->media_start_time=time(NULL);
510 if (lc->vtable.call_state_changed)
511 lc->vtable.call_state_changed(lc,call,cstate,message);
512 if (cstate==LinphoneCallReleased){
513 if (call->op!=NULL) {
514 /* so that we cannot have anymore upcalls for SAL
515 concerning this call*/
516 sal_op_release(call->op);
519 linphone_call_unref(call);
524 static void linphone_call_destroy(LinphoneCall *obj)
527 sal_op_release(obj->op);
530 if (obj->resultdesc!=NULL) {
531 sal_media_description_unref(obj->resultdesc);
532 obj->resultdesc=NULL;
534 if (obj->localdesc!=NULL) {
535 sal_media_description_unref(obj->localdesc);
539 sal_op_release(obj->ping_op);
542 ms_free(obj->refer_to);
544 if (obj->owns_call_log)
545 linphone_call_log_destroy(obj->log);
546 if (obj->auth_token) {
547 ms_free(obj->auth_token);
554 * @addtogroup call_control
559 * Increments the call 's reference count.
560 * An application that wishes to retain a pointer to call object
561 * must use this function to unsure the pointer remains
562 * valid. Once the application no more needs this pointer,
563 * it must call linphone_call_unref().
565 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
571 * Decrements the call object reference count.
572 * See linphone_call_ref().
574 void linphone_call_unref(LinphoneCall *obj){
577 linphone_call_destroy(obj);
582 * Returns current parameters associated to the call.
584 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
585 return &call->current_params;
588 static bool_t is_video_active(const SalStreamDescription *sd){
589 return sd->port!=0 && sd->dir!=SalStreamInactive;
593 * Returns call parameters proposed by remote.
595 * This is useful when receiving an incoming call, to know whether the remote party
596 * supports video, encryption or whatever.
598 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
599 LinphoneCallParams *cp=&call->remote_params;
600 memset(cp,0,sizeof(*cp));
602 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
604 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
606 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
607 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
608 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
609 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
611 cp->has_video=is_video_active(secure_vsd);
612 if (secure_asd || asd==NULL)
613 cp->media_encryption=LinphoneMediaEncryptionSRTP;
615 cp->has_video=is_video_active(vsd);
624 * Returns the remote address associated to this call
627 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
628 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
632 * Returns the remote address associated to this call as a string.
634 * The result string must be freed by user using ms_free().
636 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
637 return linphone_address_as_string(linphone_call_get_remote_address(call));
641 * Retrieves the call's current state.
643 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
648 * Returns the reason for a call termination (either error or normal termination)
650 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
655 * Get the user_pointer in the LinphoneCall
657 * @ingroup call_control
659 * return user_pointer an opaque user pointer that can be retrieved at any time
661 void *linphone_call_get_user_pointer(LinphoneCall *call)
663 return call->user_pointer;
667 * Set the user_pointer in the LinphoneCall
669 * @ingroup call_control
671 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
673 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
675 call->user_pointer = user_pointer;
679 * Returns the call log associated to this call.
681 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
686 * Returns the refer-to uri (if the call was transfered).
688 const char *linphone_call_get_refer_to(const LinphoneCall *call){
689 return call->refer_to;
693 * Returns direction of the call (incoming or outgoing).
695 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
696 return call->log->dir;
700 * Returns the far end's user agent description string, if available.
702 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
704 return sal_op_get_remote_ua (call->op);
710 * Returns true if this calls has received a transfer that has not been
712 * Pending transfers are executed when this call is being paused or closed,
713 * locally or by remote endpoint.
714 * If the call is already paused while receiving the transfer request, the
715 * transfer immediately occurs.
717 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
718 return call->refer_pending;
722 * Returns call's duration in seconds.
724 int linphone_call_get_duration(const LinphoneCall *call){
725 if (call->media_start_time==0) return 0;
726 return time(NULL)-call->media_start_time;
730 * Returns the call object this call is replacing, if any.
731 * Call replacement can occur during call transfers.
732 * By default, the core automatically terminates the replaced call and accept the new one.
733 * This function allows the application to know whether a new incoming call is a one that replaces another one.
735 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
736 SalOp *op=sal_call_get_replaces(call->op);
738 return (LinphoneCall*)sal_op_get_user_pointer(op);
744 * Indicate whether camera input should be sent to remote end.
746 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
748 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
749 LinphoneCore *lc=call->core;
750 MSWebCam *nowebcam=get_nowebcam_device();
751 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
752 video_stream_change_camera(call->videostream,
753 enable ? lc->video_conf.device : nowebcam);
756 call->camera_active=enable;
761 * Take a photo of currently received video and write it into a jpeg file.
763 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
765 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
766 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
768 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
775 * Returns TRUE if camera pictures are sent to the remote party.
777 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
778 return call->camera_active;
782 * Enable video stream.
784 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
785 cp->has_video=enabled;
788 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
789 return cp->audio_codec;
792 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
793 return cp->video_codec;
797 * Returns whether video is enabled.
799 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
800 return cp->has_video;
803 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
804 return cp->media_encryption;
807 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
808 cp->media_encryption = e;
813 * Enable sending of real early media (during outgoing calls).
815 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
816 cp->real_early_media=enabled;
819 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
820 return cp->real_early_media;
824 * Returns true if the call is part of the locally managed conference.
826 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
827 return cp->in_conference;
831 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
832 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
834 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
835 cp->audio_bw=bandwidth;
840 * Request remote side to send us a Video Fast Update.
842 void linphone_call_send_vfu_request(LinphoneCall *call)
844 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
845 sal_call_send_vfu_request(call->op);
852 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
853 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
854 memcpy(ncp,cp,sizeof(LinphoneCallParams));
861 void linphone_call_params_destroy(LinphoneCallParams *p){
870 #ifdef TEST_EXT_RENDERER
871 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
872 ms_message("rendercb, local buffer=%p, remote buffer=%p",
873 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
878 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
879 LinphoneCall* call = (LinphoneCall*) user_pointer;
880 ms_warning("In linphonecall.c: video_stream_event_cb");
882 case MS_VIDEO_DECODER_DECODING_ERRORS:
883 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
884 linphone_call_send_vfu_request(call);
886 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
887 ms_message("First video frame decoded successfully");
888 if (call->nextVideoFrameDecoded._func != NULL)
889 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
892 ms_warning("Unhandled event %i", event_id);
898 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
899 call->nextVideoFrameDecoded._func = cb;
900 call->nextVideoFrameDecoded._user_data = user_data;
902 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
906 void linphone_call_init_media_streams(LinphoneCall *call){
907 LinphoneCore *lc=call->core;
908 SalMediaDescription *md=call->localdesc;
909 AudioStream *audiostream;
911 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
912 if (linphone_core_echo_limiter_enabled(lc)){
913 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
914 if (strcasecmp(type,"mic")==0)
915 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
916 else if (strcasecmp(type,"full")==0)
917 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
919 audio_stream_enable_gain_control(audiostream,TRUE);
920 if (linphone_core_echo_cancellation_enabled(lc)){
921 int len,delay,framesize;
922 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
923 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
924 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
925 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
926 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
927 if (statestr && audiostream->ec){
928 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
931 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
933 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
934 audio_stream_enable_noise_gate(audiostream,enabled);
938 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
939 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
940 rtp_session_set_transports(audiostream->session,artp,artcp);
943 call->audiostream_app_evq = ortp_ev_queue_new();
944 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
948 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
949 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
950 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
951 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
952 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
954 if( lc->video_conf.displaytype != NULL)
955 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
956 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
958 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
959 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
960 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
962 call->videostream_app_evq = ortp_ev_queue_new();
963 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
964 #ifdef TEST_EXT_RENDERER
965 video_stream_set_render_callback(call->videostream,rendercb,NULL);
969 call->videostream=NULL;
974 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
976 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
977 LinphoneCore* lc = (LinphoneCore*)user_data;
978 if (dtmf<0 || dtmf>15){
979 ms_warning("Bad dtmf value %i",dtmf);
982 if (lc->vtable.dtmf_received != NULL)
983 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
986 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
988 MSFilter *f=st->equalizer;
989 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
990 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
991 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
997 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
998 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
999 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1008 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1009 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1012 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1013 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1014 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1017 audio_stream_set_mic_gain(st,mic_gain);
1019 audio_stream_set_mic_gain(st,0);
1021 recv_gain = lc->sound_conf.soft_play_lev;
1022 if (recv_gain != 0) {
1023 linphone_core_set_playback_gain_db (lc,recv_gain);
1027 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1028 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1029 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1030 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1031 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1032 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1035 if (speed==-1) speed=0.03;
1036 if (force==-1) force=25;
1037 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1038 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1040 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1042 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1043 if (transmit_thres!=-1)
1044 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1046 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1047 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1050 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1051 float floorgain = 1/mic_gain;
1052 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1053 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1054 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1055 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1057 parametrize_equalizer(lc,st);
1060 static void post_configure_audio_streams(LinphoneCall*call){
1061 AudioStream *st=call->audiostream;
1062 LinphoneCore *lc=call->core;
1063 _post_configure_audio_stream(st,lc,call->audio_muted);
1064 if (lc->vtable.dtmf_received!=NULL){
1065 /* replace by our default action*/
1066 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1067 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1071 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1074 RtpProfile *prof=rtp_profile_new("Call profile");
1077 LinphoneCore *lc=call->core;
1081 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1082 PayloadType *pt=(PayloadType*)elem->data;
1085 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1086 if (desc->type==SalAudio){
1087 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1088 up_ptime=linphone_core_get_upload_ptime(lc);
1090 *used_pt=payload_type_get_number(pt);
1093 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1094 else if (md->bandwidth>0) {
1095 /*case where b=AS is given globally, not per stream*/
1096 remote_bw=md->bandwidth;
1097 if (desc->type==SalVideo){
1098 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1102 if (desc->type==SalAudio){
1103 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1104 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1105 if (bw>0) pt->normal_bitrate=bw*1000;
1106 else if (desc->type==SalAudio){
1107 pt->normal_bitrate=-1;
1110 up_ptime=desc->ptime;
1114 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1115 payload_type_append_send_fmtp(pt,tmp);
1117 number=payload_type_get_number(pt);
1118 if (rtp_profile_get_payload(prof,number)!=NULL){
1119 ms_warning("A payload type with number %i already exists in profile !",number);
1121 rtp_profile_set_payload(prof,number,pt);
1127 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1128 int pause_time=3000;
1129 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1130 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1133 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1135 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1136 LinphoneCore *lc=call->core;
1137 LinphoneCall *current=linphone_core_get_current_call(lc);
1138 return !linphone_core_is_in_conference(lc) &&
1139 (current==NULL || current==call);
1141 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1143 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1144 if (crypto[i].tag == tag) {
1150 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1151 LinphoneCore *lc=call->core;
1152 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1154 /* look for savp stream first */
1155 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1156 SalProtoRtpSavp,SalAudio);
1157 /* no savp audio stream, use avp */
1159 stream=sal_media_description_find_stream(call->resultdesc,
1160 SalProtoRtpAvp,SalAudio);
1162 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1163 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1164 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1165 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1166 const char *playfile=lc->play_file;
1167 const char *recfile=lc->rec_file;
1168 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1172 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1173 if (playcard==NULL) {
1174 ms_warning("No card defined for playback !");
1176 if (captcard==NULL) {
1177 ms_warning("No card defined for capture !");
1179 /*Replace soundcard filters by inactive file players or recorders
1180 when placed in recvonly or sendonly mode*/
1181 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1184 }else if (stream->dir==SalStreamSendOnly){
1188 /*And we will eventually play "playfile" if set by the user*/
1191 if (send_ringbacktone){
1193 playfile=NULL;/* it is setup later*/
1195 /*if playfile are supplied don't use soundcards*/
1196 if (lc->use_files) {
1200 if (call->params.in_conference){
1201 /* first create the graph without soundcard resources*/
1202 captcard=playcard=NULL;
1204 if (!linphone_call_sound_resources_available(call)){
1205 ms_message("Sound resources are used by another call, not using soundcard.");
1206 captcard=playcard=NULL;
1208 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1209 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1210 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1211 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1212 audio_stream_start_full(
1214 call->audio_profile,
1215 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1217 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1226 post_configure_audio_streams(call);
1227 if (muted && !send_ringbacktone){
1228 audio_stream_set_mic_gain(call->audiostream,0);
1230 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1232 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1234 if (send_ringbacktone){
1235 setup_ring_player(lc,call);
1237 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1239 /* valid local tags are > 0 */
1240 if (stream->proto == SalProtoRtpSavp) {
1241 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1242 SalProtoRtpSavp,SalAudio);
1243 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1245 if (crypto_idx >= 0) {
1246 audio_stream_enable_strp(
1248 stream->crypto[0].algo,
1249 local_st_desc->crypto[crypto_idx].master_key,
1250 stream->crypto[0].master_key);
1251 call->audiostream_encrypted=TRUE;
1253 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1254 call->audiostream_encrypted=FALSE;
1256 }else call->audiostream_encrypted=FALSE;
1257 if (call->params.in_conference){
1258 /*transform the graph to connect it to the conference filter */
1259 bool_t mute=stream->dir==SalStreamRecvOnly;
1260 linphone_call_add_to_conf(call, mute);
1262 call->current_params.in_conference=call->params.in_conference;
1263 }else ms_warning("No audio stream accepted ?");
1267 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1268 #ifdef VIDEO_ENABLED
1269 LinphoneCore *lc=call->core;
1271 /* look for savp stream first */
1272 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1273 SalProtoRtpSavp,SalVideo);
1274 /* no savp audio stream, use avp */
1276 vstream=sal_media_description_find_stream(call->resultdesc,
1277 SalProtoRtpAvp,SalVideo);
1279 /* shutdown preview */
1280 if (lc->previewstream!=NULL) {
1281 video_preview_stop(lc->previewstream);
1282 lc->previewstream=NULL;
1285 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1286 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1287 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1289 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1290 VideoStreamDir dir=VideoStreamSendRecv;
1291 MSWebCam *cam=lc->video_conf.device;
1292 bool_t is_inactive=FALSE;
1294 call->current_params.has_video=TRUE;
1296 video_stream_enable_adaptive_bitrate_control(call->videostream,
1297 linphone_core_adaptive_rate_control_enabled(lc));
1298 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1299 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1300 if (lc->video_window_id!=0)
1301 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1302 if (lc->preview_window_id!=0)
1303 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1304 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1306 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1307 cam=get_nowebcam_device();
1308 dir=VideoStreamSendOnly;
1309 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1310 dir=VideoStreamRecvOnly;
1311 }else if (vstream->dir==SalStreamSendRecv){
1312 if (lc->video_conf.display && lc->video_conf.capture)
1313 dir=VideoStreamSendRecv;
1314 else if (lc->video_conf.display)
1315 dir=VideoStreamRecvOnly;
1317 dir=VideoStreamSendOnly;
1319 ms_warning("video stream is inactive.");
1320 /*either inactive or incompatible with local capabilities*/
1323 if (call->camera_active==FALSE || all_inputs_muted){
1324 cam=get_nowebcam_device();
1327 call->log->video_enabled = TRUE;
1328 video_stream_set_direction (call->videostream, dir);
1329 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1330 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1331 video_stream_start(call->videostream,
1332 call->video_profile, addr, vstream->port,
1333 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1334 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1335 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1338 if (vstream->proto == SalProtoRtpSavp) {
1339 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1340 SalProtoRtpSavp,SalVideo);
1342 video_stream_enable_strp(
1344 vstream->crypto[0].algo,
1345 local_st_desc->crypto[0].master_key,
1346 vstream->crypto[0].master_key
1348 call->videostream_encrypted=TRUE;
1350 call->videostream_encrypted=FALSE;
1352 }else ms_warning("No video stream accepted.");
1354 ms_warning("No valid video stream defined.");
1359 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1360 LinphoneCore *lc=call->core;
1362 call->current_params.audio_codec = NULL;
1363 call->current_params.video_codec = NULL;
1365 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1367 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1368 #ifdef VIDEO_ENABLED
1369 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1370 SalProtoRtpAvp,SalVideo);
1373 if(call->audiostream == NULL)
1375 ms_fatal("start_media_stream() called without prior init !");
1378 cname=linphone_address_as_string_uri_only(me);
1380 #if defined(VIDEO_ENABLED)
1381 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1382 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1386 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1387 call->current_params.has_video=FALSE;
1388 if (call->videostream!=NULL) {
1389 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1392 call->all_muted=all_inputs_muted;
1393 call->playing_ringbacktone=send_ringbacktone;
1394 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1396 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1397 OrtpZrtpParams params;
1398 /*will be set later when zrtp is activated*/
1399 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1401 params.zid_file=lc->zrtp_secrets_cache;
1402 audio_stream_enable_zrtp(call->audiostream,¶ms);
1403 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1404 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1405 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1408 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1409 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1410 linphone_call_fix_call_parameters(call);
1415 linphone_address_destroy(me);
1418 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1419 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1420 log->quality=audio_stream_get_average_quality_rating(st);
1423 void linphone_call_stop_media_streams(LinphoneCall *call){
1424 if (call->audiostream!=NULL) {
1425 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1426 ortp_ev_queue_flush(call->audiostream_app_evq);
1427 ortp_ev_queue_destroy(call->audiostream_app_evq);
1429 if (call->audiostream->ec){
1430 const char *state_str=NULL;
1431 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1433 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1434 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1437 linphone_call_log_fill_stats (call->log,call->audiostream);
1438 if (call->endpoint){
1439 linphone_call_remove_from_conf(call);
1441 audio_stream_stop(call->audiostream);
1442 call->audiostream=NULL;
1446 #ifdef VIDEO_ENABLED
1447 if (call->videostream!=NULL){
1448 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1449 ortp_ev_queue_flush(call->videostream_app_evq);
1450 ortp_ev_queue_destroy(call->videostream_app_evq);
1451 video_stream_stop(call->videostream);
1452 call->videostream=NULL;
1455 ms_event_queue_skip(call->core->msevq);
1457 if (call->audio_profile){
1458 rtp_profile_clear_all(call->audio_profile);
1459 rtp_profile_destroy(call->audio_profile);
1460 call->audio_profile=NULL;
1462 if (call->video_profile){
1463 rtp_profile_clear_all(call->video_profile);
1464 rtp_profile_destroy(call->video_profile);
1465 call->video_profile=NULL;
1471 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1472 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1473 bool_t bypass_mode = !enable;
1474 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1477 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1478 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1480 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1483 return linphone_core_echo_cancellation_enabled(call->core);
1487 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1488 if (call!=NULL && call->audiostream!=NULL ) {
1490 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1491 if (strcasecmp(type,"mic")==0)
1492 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1493 else if (strcasecmp(type,"full")==0)
1494 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1496 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1501 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1502 if (call!=NULL && call->audiostream!=NULL ){
1503 return call->audiostream->el_type !=ELInactive ;
1505 return linphone_core_echo_limiter_enabled(call->core);
1510 * @addtogroup call_misc
1515 * Returns the measured sound volume played locally (received from remote)
1516 * It is expressed in dbm0.
1518 float linphone_call_get_play_volume(LinphoneCall *call){
1519 AudioStream *st=call->audiostream;
1520 if (st && st->volrecv){
1522 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1526 return LINPHONE_VOLUME_DB_LOWEST;
1530 * Returns the measured sound volume recorded locally (sent to remote)
1531 * It is expressed in dbm0.
1533 float linphone_call_get_record_volume(LinphoneCall *call){
1534 AudioStream *st=call->audiostream;
1535 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1537 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1541 return LINPHONE_VOLUME_DB_LOWEST;
1545 * Obtain real-time quality rating of the call
1547 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1548 * during all the duration of the call. This function returns its value at the time of the function call.
1549 * It is expected that the rating is updated at least every 5 seconds or so.
1550 * The rating is a floating point number comprised between 0 and 5.
1552 * 4-5 = good quality <br>
1553 * 3-4 = average quality <br>
1554 * 2-3 = poor quality <br>
1555 * 1-2 = very poor quality <br>
1556 * 0-1 = can't be worse, mostly unusable <br>
1558 * @returns The function returns -1 if no quality measurement is available, for example if no
1559 * active audio stream exist. Otherwise it returns the quality rating.
1561 float linphone_call_get_current_quality(LinphoneCall *call){
1562 if (call->audiostream){
1563 return audio_stream_get_quality_rating(call->audiostream);
1569 * Returns call quality averaged over all the duration of the call.
1571 * See linphone_call_get_current_quality() for more details about quality measurement.
1573 float linphone_call_get_average_quality(LinphoneCall *call){
1574 if (call->audiostream){
1575 return audio_stream_get_average_quality_rating(call->audiostream);
1584 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1585 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1586 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1587 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1588 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1589 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1592 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1596 from = linphone_call_get_remote_address_as_string(call);
1599 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1604 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1606 if (lc->vtable.display_warning!=NULL)
1607 lc->vtable.display_warning(lc,temp);
1608 linphone_core_terminate_call(lc,call);
1611 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1612 LinphoneCore* lc = call->core;
1613 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1614 bool_t disconnected=FALSE;
1616 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1617 RtpSession *as=NULL,*vs=NULL;
1618 float audio_load=0, video_load=0;
1619 if (call->audiostream!=NULL){
1620 as=call->audiostream->session;
1621 if (call->audiostream->ticker)
1622 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1624 if (call->videostream!=NULL){
1625 if (call->videostream->ticker)
1626 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1627 vs=call->videostream->session;
1629 display_bandwidth(as,vs);
1630 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1632 #ifdef VIDEO_ENABLED
1633 if (call->videostream!=NULL) {
1634 // Beware that the application queue should not depend on treatments fron the
1635 // mediastreamer queue.
1636 video_stream_iterate(call->videostream);
1638 if (call->videostream_app_evq){
1640 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1641 OrtpEventType evt=ortp_event_get_type(ev);
1642 OrtpEventData *evd=ortp_event_get_data(ev);
1643 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1644 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1645 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1646 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1647 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1648 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1649 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1651 if (lc->vtable.call_stats_updated)
1652 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1653 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1654 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1655 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1656 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1657 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1659 if (lc->vtable.call_stats_updated)
1660 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1662 ortp_event_destroy(ev);
1667 if (call->audiostream!=NULL) {
1668 // Beware that the application queue should not depend on treatments fron the
1669 // mediastreamer queue.
1670 audio_stream_iterate(call->audiostream);
1672 if (call->audiostream_app_evq){
1674 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1675 OrtpEventType evt=ortp_event_get_type(ev);
1676 OrtpEventData *evd=ortp_event_get_data(ev);
1677 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1678 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1679 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1680 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1681 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1682 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1683 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1684 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1685 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1687 if (lc->vtable.call_stats_updated)
1688 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1689 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1690 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1691 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1692 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1693 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1695 if (lc->vtable.call_stats_updated)
1696 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1698 ortp_event_destroy(ev);
1702 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1703 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1705 linphone_core_disconnected(call->core,call);
1708 void linphone_call_log_completed(LinphoneCall *call){
1709 LinphoneCore *lc=call->core;
1711 call->log->duration=time(NULL)-call->start_time;
1713 if (call->log->status==LinphoneCallMissed){
1716 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1717 "You have missed %i calls.", lc->missed_calls),
1719 if (lc->vtable.display_status!=NULL)
1720 lc->vtable.display_status(lc,info);
1723 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1724 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1725 MSList *elem,*prevelem=NULL;
1726 /*find the last element*/
1727 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1731 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1732 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1734 if (lc->vtable.call_log_updated!=NULL){
1735 lc->vtable.call_log_updated(lc,call->log);
1737 call_logs_write_to_config_file(lc);
1740 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1741 return call->transfer_state;
1744 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1745 if (state != call->transfer_state) {
1746 LinphoneCore* lc = call->core;
1747 call->transfer_state = state;
1748 if (lc->vtable.transfer_state_changed)
1749 lc->vtable.transfer_state_changed(lc, call, state);