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 switch(call->reason){
501 case LinphoneReasonDeclined:
502 call->log->status=LinphoneCallDeclined;
503 case LinphoneReasonNotAnswered:
504 call->log->status=LinphoneCallMissed;
509 linphone_call_set_terminated (call);
511 if (cstate == LinphoneCallConnected) {
512 call->log->status=LinphoneCallSuccess;
513 call->media_start_time=time(NULL);
516 if (lc->vtable.call_state_changed)
517 lc->vtable.call_state_changed(lc,call,cstate,message);
518 if (cstate==LinphoneCallReleased){
519 if (call->op!=NULL) {
520 /* so that we cannot have anymore upcalls for SAL
521 concerning this call*/
522 sal_op_release(call->op);
525 linphone_call_unref(call);
530 static void linphone_call_destroy(LinphoneCall *obj)
533 sal_op_release(obj->op);
536 if (obj->resultdesc!=NULL) {
537 sal_media_description_unref(obj->resultdesc);
538 obj->resultdesc=NULL;
540 if (obj->localdesc!=NULL) {
541 sal_media_description_unref(obj->localdesc);
545 sal_op_release(obj->ping_op);
548 ms_free(obj->refer_to);
550 if (obj->owns_call_log)
551 linphone_call_log_destroy(obj->log);
552 if (obj->auth_token) {
553 ms_free(obj->auth_token);
560 * @addtogroup call_control
565 * Increments the call 's reference count.
566 * An application that wishes to retain a pointer to call object
567 * must use this function to unsure the pointer remains
568 * valid. Once the application no more needs this pointer,
569 * it must call linphone_call_unref().
571 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
577 * Decrements the call object reference count.
578 * See linphone_call_ref().
580 void linphone_call_unref(LinphoneCall *obj){
583 linphone_call_destroy(obj);
588 * Returns current parameters associated to the call.
590 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
591 return &call->current_params;
594 static bool_t is_video_active(const SalStreamDescription *sd){
595 return sd->port!=0 && sd->dir!=SalStreamInactive;
599 * Returns call parameters proposed by remote.
601 * This is useful when receiving an incoming call, to know whether the remote party
602 * supports video, encryption or whatever.
604 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
605 LinphoneCallParams *cp=&call->remote_params;
606 memset(cp,0,sizeof(*cp));
608 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
610 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
612 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
613 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
614 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
615 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
617 cp->has_video=is_video_active(secure_vsd);
618 if (secure_asd || asd==NULL)
619 cp->media_encryption=LinphoneMediaEncryptionSRTP;
621 cp->has_video=is_video_active(vsd);
630 * Returns the remote address associated to this call
633 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
634 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
638 * Returns the remote address associated to this call as a string.
640 * The result string must be freed by user using ms_free().
642 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
643 return linphone_address_as_string(linphone_call_get_remote_address(call));
647 * Retrieves the call's current state.
649 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
654 * Returns the reason for a call termination (either error or normal termination)
656 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
661 * Get the user_pointer in the LinphoneCall
663 * @ingroup call_control
665 * return user_pointer an opaque user pointer that can be retrieved at any time
667 void *linphone_call_get_user_pointer(LinphoneCall *call)
669 return call->user_pointer;
673 * Set the user_pointer in the LinphoneCall
675 * @ingroup call_control
677 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
679 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
681 call->user_pointer = user_pointer;
685 * Returns the call log associated to this call.
687 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
692 * Returns the refer-to uri (if the call was transfered).
694 const char *linphone_call_get_refer_to(const LinphoneCall *call){
695 return call->refer_to;
699 * Returns direction of the call (incoming or outgoing).
701 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
702 return call->log->dir;
706 * Returns the far end's user agent description string, if available.
708 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
710 return sal_op_get_remote_ua (call->op);
716 * Returns true if this calls has received a transfer that has not been
718 * Pending transfers are executed when this call is being paused or closed,
719 * locally or by remote endpoint.
720 * If the call is already paused while receiving the transfer request, the
721 * transfer immediately occurs.
723 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
724 return call->refer_pending;
728 * Returns call's duration in seconds.
730 int linphone_call_get_duration(const LinphoneCall *call){
731 if (call->media_start_time==0) return 0;
732 return time(NULL)-call->media_start_time;
736 * Returns the call object this call is replacing, if any.
737 * Call replacement can occur during call transfers.
738 * By default, the core automatically terminates the replaced call and accept the new one.
739 * This function allows the application to know whether a new incoming call is a one that replaces another one.
741 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
742 SalOp *op=sal_call_get_replaces(call->op);
744 return (LinphoneCall*)sal_op_get_user_pointer(op);
750 * Indicate whether camera input should be sent to remote end.
752 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
754 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
755 LinphoneCore *lc=call->core;
756 MSWebCam *nowebcam=get_nowebcam_device();
757 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
758 video_stream_change_camera(call->videostream,
759 enable ? lc->video_conf.device : nowebcam);
762 call->camera_active=enable;
767 * Take a photo of currently received video and write it into a jpeg file.
769 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
771 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
772 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
774 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
781 * Returns TRUE if camera pictures are sent to the remote party.
783 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
784 return call->camera_active;
788 * Enable video stream.
790 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
791 cp->has_video=enabled;
794 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
795 return cp->audio_codec;
798 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
799 return cp->video_codec;
803 * Returns whether video is enabled.
805 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
806 return cp->has_video;
809 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
810 return cp->media_encryption;
813 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
814 cp->media_encryption = e;
819 * Enable sending of real early media (during outgoing calls).
821 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
822 cp->real_early_media=enabled;
825 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
826 return cp->real_early_media;
830 * Returns true if the call is part of the locally managed conference.
832 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
833 return cp->in_conference;
837 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
838 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
840 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
841 cp->audio_bw=bandwidth;
846 * Request remote side to send us a Video Fast Update.
848 void linphone_call_send_vfu_request(LinphoneCall *call)
850 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
851 sal_call_send_vfu_request(call->op);
858 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
859 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
860 memcpy(ncp,cp,sizeof(LinphoneCallParams));
867 void linphone_call_params_destroy(LinphoneCallParams *p){
876 #ifdef TEST_EXT_RENDERER
877 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
878 ms_message("rendercb, local buffer=%p, remote buffer=%p",
879 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
884 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
885 LinphoneCall* call = (LinphoneCall*) user_pointer;
886 ms_warning("In linphonecall.c: video_stream_event_cb");
888 case MS_VIDEO_DECODER_DECODING_ERRORS:
889 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
890 linphone_call_send_vfu_request(call);
892 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
893 ms_message("First video frame decoded successfully");
894 if (call->nextVideoFrameDecoded._func != NULL)
895 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
898 ms_warning("Unhandled event %i", event_id);
904 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
905 call->nextVideoFrameDecoded._func = cb;
906 call->nextVideoFrameDecoded._user_data = user_data;
908 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
912 void linphone_call_init_media_streams(LinphoneCall *call){
913 LinphoneCore *lc=call->core;
914 SalMediaDescription *md=call->localdesc;
915 AudioStream *audiostream;
917 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
918 if (linphone_core_echo_limiter_enabled(lc)){
919 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
920 if (strcasecmp(type,"mic")==0)
921 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
922 else if (strcasecmp(type,"full")==0)
923 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
925 audio_stream_enable_gain_control(audiostream,TRUE);
926 if (linphone_core_echo_cancellation_enabled(lc)){
927 int len,delay,framesize;
928 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
929 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
930 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
931 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
932 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
933 if (statestr && audiostream->ec){
934 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
937 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
939 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
940 audio_stream_enable_noise_gate(audiostream,enabled);
944 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
945 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
946 rtp_session_set_transports(audiostream->session,artp,artcp);
949 call->audiostream_app_evq = ortp_ev_queue_new();
950 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
954 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
955 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
956 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
957 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
958 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
960 if( lc->video_conf.displaytype != NULL)
961 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
962 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
964 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
965 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
966 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
968 call->videostream_app_evq = ortp_ev_queue_new();
969 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
970 #ifdef TEST_EXT_RENDERER
971 video_stream_set_render_callback(call->videostream,rendercb,NULL);
975 call->videostream=NULL;
980 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
982 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
983 LinphoneCore* lc = (LinphoneCore*)user_data;
984 if (dtmf<0 || dtmf>15){
985 ms_warning("Bad dtmf value %i",dtmf);
988 if (lc->vtable.dtmf_received != NULL)
989 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
992 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
994 MSFilter *f=st->equalizer;
995 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
996 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
997 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1003 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1004 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1005 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1014 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1015 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1018 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1019 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1020 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1023 audio_stream_set_mic_gain(st,mic_gain);
1025 audio_stream_set_mic_gain(st,0);
1027 recv_gain = lc->sound_conf.soft_play_lev;
1028 if (recv_gain != 0) {
1029 linphone_core_set_playback_gain_db (lc,recv_gain);
1033 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1034 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1035 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1036 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1037 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1038 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1041 if (speed==-1) speed=0.03;
1042 if (force==-1) force=25;
1043 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1044 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1046 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1048 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1049 if (transmit_thres!=-1)
1050 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1052 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1053 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1056 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1057 float floorgain = 1/mic_gain;
1058 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1059 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1060 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1061 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1063 parametrize_equalizer(lc,st);
1066 static void post_configure_audio_streams(LinphoneCall*call){
1067 AudioStream *st=call->audiostream;
1068 LinphoneCore *lc=call->core;
1069 _post_configure_audio_stream(st,lc,call->audio_muted);
1070 if (lc->vtable.dtmf_received!=NULL){
1071 /* replace by our default action*/
1072 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1073 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1077 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1080 RtpProfile *prof=rtp_profile_new("Call profile");
1083 LinphoneCore *lc=call->core;
1087 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1088 PayloadType *pt=(PayloadType*)elem->data;
1091 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1092 if (desc->type==SalAudio){
1093 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1094 up_ptime=linphone_core_get_upload_ptime(lc);
1096 *used_pt=payload_type_get_number(pt);
1099 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1100 else if (md->bandwidth>0) {
1101 /*case where b=AS is given globally, not per stream*/
1102 remote_bw=md->bandwidth;
1103 if (desc->type==SalVideo){
1104 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1108 if (desc->type==SalAudio){
1109 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1110 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1111 if (bw>0) pt->normal_bitrate=bw*1000;
1112 else if (desc->type==SalAudio){
1113 pt->normal_bitrate=-1;
1116 up_ptime=desc->ptime;
1120 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1121 payload_type_append_send_fmtp(pt,tmp);
1123 number=payload_type_get_number(pt);
1124 if (rtp_profile_get_payload(prof,number)!=NULL){
1125 ms_warning("A payload type with number %i already exists in profile !",number);
1127 rtp_profile_set_payload(prof,number,pt);
1133 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1134 int pause_time=3000;
1135 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1136 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1139 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1141 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1142 LinphoneCore *lc=call->core;
1143 LinphoneCall *current=linphone_core_get_current_call(lc);
1144 return !linphone_core_is_in_conference(lc) &&
1145 (current==NULL || current==call);
1147 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1149 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1150 if (crypto[i].tag == tag) {
1156 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1157 LinphoneCore *lc=call->core;
1158 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1160 /* look for savp stream first */
1161 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1162 SalProtoRtpSavp,SalAudio);
1163 /* no savp audio stream, use avp */
1165 stream=sal_media_description_find_stream(call->resultdesc,
1166 SalProtoRtpAvp,SalAudio);
1168 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1169 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1170 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1171 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1172 const char *playfile=lc->play_file;
1173 const char *recfile=lc->rec_file;
1174 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1178 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1179 if (playcard==NULL) {
1180 ms_warning("No card defined for playback !");
1182 if (captcard==NULL) {
1183 ms_warning("No card defined for capture !");
1185 /*Replace soundcard filters by inactive file players or recorders
1186 when placed in recvonly or sendonly mode*/
1187 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1190 }else if (stream->dir==SalStreamSendOnly){
1194 /*And we will eventually play "playfile" if set by the user*/
1197 if (send_ringbacktone){
1199 playfile=NULL;/* it is setup later*/
1201 /*if playfile are supplied don't use soundcards*/
1202 if (lc->use_files) {
1206 if (call->params.in_conference){
1207 /* first create the graph without soundcard resources*/
1208 captcard=playcard=NULL;
1210 if (!linphone_call_sound_resources_available(call)){
1211 ms_message("Sound resources are used by another call, not using soundcard.");
1212 captcard=playcard=NULL;
1214 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1215 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1216 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1217 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1218 audio_stream_start_full(
1220 call->audio_profile,
1221 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1223 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1232 post_configure_audio_streams(call);
1233 if (muted && !send_ringbacktone){
1234 audio_stream_set_mic_gain(call->audiostream,0);
1236 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1238 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1240 if (send_ringbacktone){
1241 setup_ring_player(lc,call);
1243 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1245 /* valid local tags are > 0 */
1246 if (stream->proto == SalProtoRtpSavp) {
1247 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1248 SalProtoRtpSavp,SalAudio);
1249 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1251 if (crypto_idx >= 0) {
1252 audio_stream_enable_strp(
1254 stream->crypto[0].algo,
1255 local_st_desc->crypto[crypto_idx].master_key,
1256 stream->crypto[0].master_key);
1257 call->audiostream_encrypted=TRUE;
1259 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1260 call->audiostream_encrypted=FALSE;
1262 }else call->audiostream_encrypted=FALSE;
1263 if (call->params.in_conference){
1264 /*transform the graph to connect it to the conference filter */
1265 bool_t mute=stream->dir==SalStreamRecvOnly;
1266 linphone_call_add_to_conf(call, mute);
1268 call->current_params.in_conference=call->params.in_conference;
1269 }else ms_warning("No audio stream accepted ?");
1273 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1274 #ifdef VIDEO_ENABLED
1275 LinphoneCore *lc=call->core;
1277 /* look for savp stream first */
1278 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1279 SalProtoRtpSavp,SalVideo);
1280 /* no savp audio stream, use avp */
1282 vstream=sal_media_description_find_stream(call->resultdesc,
1283 SalProtoRtpAvp,SalVideo);
1285 /* shutdown preview */
1286 if (lc->previewstream!=NULL) {
1287 video_preview_stop(lc->previewstream);
1288 lc->previewstream=NULL;
1291 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1292 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1293 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1295 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1296 VideoStreamDir dir=VideoStreamSendRecv;
1297 MSWebCam *cam=lc->video_conf.device;
1298 bool_t is_inactive=FALSE;
1300 call->current_params.has_video=TRUE;
1302 video_stream_enable_adaptive_bitrate_control(call->videostream,
1303 linphone_core_adaptive_rate_control_enabled(lc));
1304 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1305 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1306 if (lc->video_window_id!=0)
1307 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1308 if (lc->preview_window_id!=0)
1309 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1310 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1312 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1313 cam=get_nowebcam_device();
1314 dir=VideoStreamSendOnly;
1315 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1316 dir=VideoStreamRecvOnly;
1317 }else if (vstream->dir==SalStreamSendRecv){
1318 if (lc->video_conf.display && lc->video_conf.capture)
1319 dir=VideoStreamSendRecv;
1320 else if (lc->video_conf.display)
1321 dir=VideoStreamRecvOnly;
1323 dir=VideoStreamSendOnly;
1325 ms_warning("video stream is inactive.");
1326 /*either inactive or incompatible with local capabilities*/
1329 if (call->camera_active==FALSE || all_inputs_muted){
1330 cam=get_nowebcam_device();
1333 call->log->video_enabled = TRUE;
1334 video_stream_set_direction (call->videostream, dir);
1335 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1336 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1337 video_stream_start(call->videostream,
1338 call->video_profile, addr, vstream->port,
1339 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1340 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1341 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1344 if (vstream->proto == SalProtoRtpSavp) {
1345 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1346 SalProtoRtpSavp,SalVideo);
1348 video_stream_enable_strp(
1350 vstream->crypto[0].algo,
1351 local_st_desc->crypto[0].master_key,
1352 vstream->crypto[0].master_key
1354 call->videostream_encrypted=TRUE;
1356 call->videostream_encrypted=FALSE;
1358 }else ms_warning("No video stream accepted.");
1360 ms_warning("No valid video stream defined.");
1365 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1366 LinphoneCore *lc=call->core;
1368 call->current_params.audio_codec = NULL;
1369 call->current_params.video_codec = NULL;
1371 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1373 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1374 #ifdef VIDEO_ENABLED
1375 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1376 SalProtoRtpAvp,SalVideo);
1379 if(call->audiostream == NULL)
1381 ms_fatal("start_media_stream() called without prior init !");
1384 cname=linphone_address_as_string_uri_only(me);
1386 #if defined(VIDEO_ENABLED)
1387 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1388 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1392 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1393 call->current_params.has_video=FALSE;
1394 if (call->videostream!=NULL) {
1395 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1398 call->all_muted=all_inputs_muted;
1399 call->playing_ringbacktone=send_ringbacktone;
1400 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1402 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1403 OrtpZrtpParams params;
1404 /*will be set later when zrtp is activated*/
1405 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1407 params.zid_file=lc->zrtp_secrets_cache;
1408 audio_stream_enable_zrtp(call->audiostream,¶ms);
1409 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1410 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1411 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1414 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1415 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1416 linphone_call_fix_call_parameters(call);
1421 linphone_address_destroy(me);
1424 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1425 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1426 log->quality=audio_stream_get_average_quality_rating(st);
1429 void linphone_call_stop_media_streams(LinphoneCall *call){
1430 if (call->audiostream!=NULL) {
1431 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1432 ortp_ev_queue_flush(call->audiostream_app_evq);
1433 ortp_ev_queue_destroy(call->audiostream_app_evq);
1435 if (call->audiostream->ec){
1436 const char *state_str=NULL;
1437 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1439 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1440 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1443 linphone_call_log_fill_stats (call->log,call->audiostream);
1444 if (call->endpoint){
1445 linphone_call_remove_from_conf(call);
1447 audio_stream_stop(call->audiostream);
1448 call->audiostream=NULL;
1452 #ifdef VIDEO_ENABLED
1453 if (call->videostream!=NULL){
1454 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1455 ortp_ev_queue_flush(call->videostream_app_evq);
1456 ortp_ev_queue_destroy(call->videostream_app_evq);
1457 video_stream_stop(call->videostream);
1458 call->videostream=NULL;
1461 ms_event_queue_skip(call->core->msevq);
1463 if (call->audio_profile){
1464 rtp_profile_clear_all(call->audio_profile);
1465 rtp_profile_destroy(call->audio_profile);
1466 call->audio_profile=NULL;
1468 if (call->video_profile){
1469 rtp_profile_clear_all(call->video_profile);
1470 rtp_profile_destroy(call->video_profile);
1471 call->video_profile=NULL;
1477 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1478 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1479 bool_t bypass_mode = !enable;
1480 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1483 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1484 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1486 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1489 return linphone_core_echo_cancellation_enabled(call->core);
1493 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1494 if (call!=NULL && call->audiostream!=NULL ) {
1496 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1497 if (strcasecmp(type,"mic")==0)
1498 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1499 else if (strcasecmp(type,"full")==0)
1500 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1502 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1507 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1508 if (call!=NULL && call->audiostream!=NULL ){
1509 return call->audiostream->el_type !=ELInactive ;
1511 return linphone_core_echo_limiter_enabled(call->core);
1516 * @addtogroup call_misc
1521 * Returns the measured sound volume played locally (received from remote)
1522 * It is expressed in dbm0.
1524 float linphone_call_get_play_volume(LinphoneCall *call){
1525 AudioStream *st=call->audiostream;
1526 if (st && st->volrecv){
1528 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1532 return LINPHONE_VOLUME_DB_LOWEST;
1536 * Returns the measured sound volume recorded locally (sent to remote)
1537 * It is expressed in dbm0.
1539 float linphone_call_get_record_volume(LinphoneCall *call){
1540 AudioStream *st=call->audiostream;
1541 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1543 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1547 return LINPHONE_VOLUME_DB_LOWEST;
1551 * Obtain real-time quality rating of the call
1553 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1554 * during all the duration of the call. This function returns its value at the time of the function call.
1555 * It is expected that the rating is updated at least every 5 seconds or so.
1556 * The rating is a floating point number comprised between 0 and 5.
1558 * 4-5 = good quality <br>
1559 * 3-4 = average quality <br>
1560 * 2-3 = poor quality <br>
1561 * 1-2 = very poor quality <br>
1562 * 0-1 = can't be worse, mostly unusable <br>
1564 * @returns The function returns -1 if no quality measurement is available, for example if no
1565 * active audio stream exist. Otherwise it returns the quality rating.
1567 float linphone_call_get_current_quality(LinphoneCall *call){
1568 if (call->audiostream){
1569 return audio_stream_get_quality_rating(call->audiostream);
1575 * Returns call quality averaged over all the duration of the call.
1577 * See linphone_call_get_current_quality() for more details about quality measurement.
1579 float linphone_call_get_average_quality(LinphoneCall *call){
1580 if (call->audiostream){
1581 return audio_stream_get_average_quality_rating(call->audiostream);
1590 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1591 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1592 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1593 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1594 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1595 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1598 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1602 from = linphone_call_get_remote_address_as_string(call);
1605 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1610 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1612 if (lc->vtable.display_warning!=NULL)
1613 lc->vtable.display_warning(lc,temp);
1614 linphone_core_terminate_call(lc,call);
1617 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1618 LinphoneCore* lc = call->core;
1619 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1620 bool_t disconnected=FALSE;
1622 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1623 RtpSession *as=NULL,*vs=NULL;
1624 float audio_load=0, video_load=0;
1625 if (call->audiostream!=NULL){
1626 as=call->audiostream->session;
1627 if (call->audiostream->ticker)
1628 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1630 if (call->videostream!=NULL){
1631 if (call->videostream->ticker)
1632 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1633 vs=call->videostream->session;
1635 display_bandwidth(as,vs);
1636 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1638 #ifdef VIDEO_ENABLED
1639 if (call->videostream!=NULL) {
1640 // Beware that the application queue should not depend on treatments fron the
1641 // mediastreamer queue.
1642 video_stream_iterate(call->videostream);
1644 if (call->videostream_app_evq){
1646 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1647 OrtpEventType evt=ortp_event_get_type(ev);
1648 OrtpEventData *evd=ortp_event_get_data(ev);
1649 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1650 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1651 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1652 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1653 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1654 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1655 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1657 if (lc->vtable.call_stats_updated)
1658 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1659 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1660 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1661 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1662 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1663 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1665 if (lc->vtable.call_stats_updated)
1666 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1668 ortp_event_destroy(ev);
1673 if (call->audiostream!=NULL) {
1674 // Beware that the application queue should not depend on treatments fron the
1675 // mediastreamer queue.
1676 audio_stream_iterate(call->audiostream);
1678 if (call->audiostream_app_evq){
1680 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1681 OrtpEventType evt=ortp_event_get_type(ev);
1682 OrtpEventData *evd=ortp_event_get_data(ev);
1683 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1684 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1685 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1686 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1687 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1688 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1689 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1690 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1691 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1693 if (lc->vtable.call_stats_updated)
1694 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1695 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1696 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1697 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1698 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1699 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1701 if (lc->vtable.call_stats_updated)
1702 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1704 ortp_event_destroy(ev);
1708 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1709 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1711 linphone_core_disconnected(call->core,call);
1714 void linphone_call_log_completed(LinphoneCall *call){
1715 LinphoneCore *lc=call->core;
1717 call->log->duration=time(NULL)-call->start_time;
1719 if (call->log->status==LinphoneCallMissed){
1722 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1723 "You have missed %i calls.", lc->missed_calls),
1725 if (lc->vtable.display_status!=NULL)
1726 lc->vtable.display_status(lc,info);
1729 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1730 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1731 MSList *elem,*prevelem=NULL;
1732 /*find the last element*/
1733 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1737 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1738 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1740 if (lc->vtable.call_log_updated!=NULL){
1741 lc->vtable.call_log_updated(lc,call->log);
1743 call_logs_write_to_config_file(lc);
1746 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1747 return call->transfer_state;
1750 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1751 if (state != call->transfer_state) {
1752 LinphoneCore* lc = call->core;
1753 call->transfer_state = state;
1754 if (lc->vtable.transfer_state_changed)
1755 lc->vtable.transfer_state_changed(lc, call, state);