4 Copyright (C) 2010 Belledonne Communications SARL
5 (simon.morlat@linphone.org)
7 This program is free software; you can redistribute it and/or
8 modify it under the terms of the GNU General Public License
9 as published by the Free Software Foundation; either version 2
10 of the License, or (at your option) any later version.
12 This program is distributed in the hope that it will be useful,
13 but WITHOUT ANY WARRANTY; without even the implied warranty of
14 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
15 GNU General Public License for more details.
17 You should have received a copy of the GNU General Public License
18 along with this program; if not, write to the Free Software
19 Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
24 #include "linphonecore.h"
28 #include <ortp/event.h>
32 #include "mediastreamer2/mediastream.h"
33 #include "mediastreamer2/msvolume.h"
34 #include "mediastreamer2/msequalizer.h"
35 #include "mediastreamer2/msfileplayer.h"
36 #include "mediastreamer2/msjpegwriter.h"
37 #include "mediastreamer2/mseventqueue.h"
38 #include "mediastreamer2/mssndcard.h"
41 static MSWebCam *get_nowebcam_device(){
42 return ms_web_cam_manager_get_cam(ms_web_cam_manager_get(),"StaticImage: Static picture");
46 static bool_t generate_b64_crypto_key(int key_length, char* key_out) {
48 uint8_t* tmp = (uint8_t*) malloc(key_length);
49 if (ortp_crypto_get_random(tmp, key_length)!=0) {
50 ms_error("Failed to generate random key");
55 b64_size = b64_encode((const char*)tmp, key_length, NULL, 0);
57 ms_error("Failed to b64 encode key");
61 key_out[b64_size] = '\0';
62 b64_encode((const char*)tmp, key_length, key_out, 40);
67 LinphoneCore *linphone_call_get_core(const LinphoneCall *call){
71 const char* linphone_call_get_authentication_token(LinphoneCall *call){
72 return call->auth_token;
75 bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
76 return call->auth_token_verified;
79 static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
80 // Check ZRTP encryption in audiostream
81 if (!call->audiostream_encrypted) {
86 // If video enabled, check ZRTP encryption in videostream
87 const LinphoneCallParams *params=linphone_call_get_current_params(call);
88 if (params->has_video && !call->videostream_encrypted) {
96 void propagate_encryption_changed(LinphoneCall *call){
97 LinphoneCore *lc=call->core;
98 if (!linphone_call_are_all_streams_encrypted(call)) {
99 ms_message("Some streams are not encrypted");
100 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
101 if (lc->vtable.call_encryption_changed)
102 lc->vtable.call_encryption_changed(call->core, call, FALSE, call->auth_token);
104 ms_message("All streams are encrypted");
105 call->current_params.media_encryption=LinphoneMediaEncryptionZRTP;
106 if (lc->vtable.call_encryption_changed)
107 lc->vtable.call_encryption_changed(call->core, call, TRUE, call->auth_token);
112 static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
113 ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
115 LinphoneCall *call = (LinphoneCall *)data;
116 call->videostream_encrypted=encrypted;
117 propagate_encryption_changed(call);
121 static void linphone_call_audiostream_encryption_changed(void *data, bool_t encrypted) {
122 char status[255]={0};
123 ms_message("Audio stream is %s ", encrypted ? "encrypted" : "not encrypted");
125 LinphoneCall *call = (LinphoneCall *)data;
126 call->audiostream_encrypted=encrypted;
128 if (encrypted && call->core->vtable.display_status != NULL) {
129 snprintf(status,sizeof(status)-1,_("Authentication token is %s"),call->auth_token);
130 call->core->vtable.display_status(call->core, status);
133 propagate_encryption_changed(call);
137 // Enable video encryption
138 const LinphoneCallParams *params=linphone_call_get_current_params(call);
139 if (params->has_video) {
140 ms_message("Trying to enable encryption on video stream");
141 OrtpZrtpParams params;
142 params.zid_file=NULL; //unused
143 video_stream_enable_zrtp(call->videostream,call->audiostream,¶ms);
149 static void linphone_call_audiostream_auth_token_ready(void *data, const char* auth_token, bool_t verified) {
150 LinphoneCall *call=(LinphoneCall *)data;
151 if (call->auth_token != NULL)
152 ms_free(call->auth_token);
154 call->auth_token=ms_strdup(auth_token);
155 call->auth_token_verified=verified;
157 ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
160 void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t verified){
161 if (call->audiostream==NULL){
162 ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
164 if (call->audiostream->ortpZrtpContext==NULL){
165 ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
167 if (!call->auth_token_verified && verified){
168 ortp_zrtp_sas_verified(call->audiostream->ortpZrtpContext);
169 }else if (call->auth_token_verified && !verified){
170 ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
172 call->auth_token_verified=verified;
173 propagate_encryption_changed(call);
176 static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
179 if (max_sample_rate) *max_sample_rate=0;
180 for(it=codecs;it!=NULL;it=it->next){
181 PayloadType *pt=(PayloadType*)it->data;
182 if (pt->flags & PAYLOAD_TYPE_ENABLED){
183 if (bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,bandwidth_limit)){
184 ms_message("Codec %s/%i eliminated because of audio bandwidth constraint.",pt->mime_type,pt->clock_rate);
187 if (linphone_core_check_payload_type_usability(lc,pt)){
188 l=ms_list_append(l,payload_type_clone(pt));
189 if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
196 static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
200 const char *me=linphone_core_get_identity(lc);
201 LinphoneAddress *addr=linphone_address_new(me);
202 const char *username=linphone_address_get_username (addr);
203 SalMediaDescription *md=sal_media_description_new();
206 md->session_id=session_id;
207 md->session_ver=session_ver;
209 strncpy(md->addr,call->localip,sizeof(md->addr));
210 strncpy(md->username,username,sizeof(md->username));
211 md->bandwidth=linphone_core_get_download_bandwidth(lc);
213 /*set audio capabilities */
214 strncpy(md->streams[0].addr,call->localip,sizeof(md->streams[0].addr));
215 md->streams[0].port=call->audio_port;
216 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
217 SalProtoRtpSavp : SalProtoRtpAvp;
218 md->streams[0].type=SalAudio;
219 md->streams[0].ptime=lc->net_conf.down_ptime;
220 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
221 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
222 l=ms_list_append(l,pt);
223 md->streams[0].payloads=l;
227 if (call->params.has_video){
229 md->streams[1].port=call->video_port;
230 md->streams[1].proto=md->streams[0].proto;
231 md->streams[1].type=SalVideo;
232 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
233 md->streams[1].payloads=l;
236 for(i=0; i<md->nstreams; i++) {
237 if (md->streams[i].proto == SalProtoRtpSavp) {
238 md->streams[i].crypto[0].tag = 1;
239 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
240 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
241 md->streams[i].crypto[0].algo = 0;
242 md->streams[i].crypto[1].tag = 2;
243 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
244 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
245 md->streams[i].crypto[1].algo = 0;
246 md->streams[i].crypto[2].algo = 0;
250 linphone_address_destroy(addr);
254 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
255 SalMediaDescription *md=call->localdesc;
257 call->localdesc = create_local_media_description(lc,call);
259 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
260 sal_media_description_unref(md);
264 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
265 unsigned int id=rand() & 0xfff;
266 return _create_local_media_description(lc,call,id,id);
269 static int find_port_offset(LinphoneCore *lc){
273 bool_t already_used=FALSE;
274 for(offset=0;offset<100;offset+=2){
275 audio_port=linphone_core_get_audio_port (lc)+offset;
277 for(elem=lc->calls;elem!=NULL;elem=elem->next){
278 LinphoneCall *call=(LinphoneCall*)elem->data;
279 if (call->audio_port==audio_port) {
284 if (!already_used) break;
287 ms_error("Could not find any free port !");
293 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
295 call->magic=linphone_call_magic;
297 call->state=LinphoneCallIdle;
298 call->transfer_state = LinphoneCallIdle;
299 call->start_time=time(NULL);
300 call->media_start_time=0;
301 call->log=linphone_call_log_new(call, from, to);
302 call->owns_call_log=TRUE;
303 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
304 port_offset=find_port_offset (call->core);
305 if (port_offset==-1) return;
306 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
307 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
308 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
309 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
312 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
314 stats->received_rtcp = NULL;
315 stats->sent_rtcp = NULL;
318 static void discover_mtu(LinphoneCore *lc, const char *remote){
320 if (lc->net_conf.mtu==0 ){
321 /*attempt to discover mtu*/
322 mtu=ms_discover_mtu(remote);
325 ms_message("Discovered mtu is %i, RTP payload max size is %i",
326 mtu, ms_get_payload_max_size());
331 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
333 LinphoneCall *call=ms_new0(LinphoneCall,1);
334 call->dir=LinphoneCallOutgoing;
335 call->op=sal_op_new(lc->sal);
336 sal_op_set_user_pointer(call->op,call);
338 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
339 linphone_call_init_common(call,from,to);
340 call->params=*params;
341 call->localdesc=create_local_media_description (lc,call);
342 call->camera_active=params->has_video;
343 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
344 linphone_core_run_stun_tests(call->core,call);
345 discover_mtu(lc,linphone_address_get_domain (to));
346 if (params->referer){
347 sal_call_set_referer(call->op,params->referer->op);
348 call->referer=linphone_call_ref(params->referer);
353 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
354 LinphoneCall *call=ms_new0(LinphoneCall,1);
357 call->dir=LinphoneCallIncoming;
358 sal_op_set_user_pointer(op,call);
362 if (lc->sip_conf.ping_with_options){
363 /*the following sends an option request back to the caller so that
364 we get a chance to discover our nat'd address before answering.*/
365 call->ping_op=sal_op_new(lc->sal);
366 from_str=linphone_address_as_string_uri_only(from);
367 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
368 sal_op_set_user_pointer(call->ping_op,call);
369 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
373 linphone_address_clean(from);
374 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
375 linphone_call_init_common(call, from, to);
376 linphone_core_init_default_params(lc, &call->params);
377 call->params.has_video &= !!lc->video_policy.automatically_accept;
378 call->localdesc=create_local_media_description (lc,call);
379 call->camera_active=call->params.has_video;
380 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
381 linphone_core_run_stun_tests(call->core,call);
382 discover_mtu(lc,linphone_address_get_domain(from));
386 /* this function is called internally to get rid of a call.
387 It performs the following tasks:
388 - remove the call from the internal list of calls
389 - update the call logs accordingly
392 static void linphone_call_set_terminated(LinphoneCall *call){
393 LinphoneCore *lc=call->core;
395 linphone_core_update_allocated_audio_bandwidth(lc);
397 call->owns_call_log=FALSE;
398 linphone_call_log_completed(call);
401 if (call == lc->current_call){
402 ms_message("Resetting the current call");
403 lc->current_call=NULL;
406 if (linphone_core_del_call(lc,call) != 0){
407 ms_error("Could not remove the call from the list !!!");
410 if (ms_list_size(lc->calls)==0)
411 linphone_core_notify_all_friends(lc,lc->presence_mode);
413 linphone_core_conference_check_uninit(lc);
414 if (call->ringing_beep){
415 linphone_core_stop_dtmf(lc);
416 call->ringing_beep=FALSE;
419 linphone_call_unref(call->referer);
424 void linphone_call_fix_call_parameters(LinphoneCall *call){
425 call->params.has_video=call->current_params.has_video;
426 call->params.media_encryption=call->current_params.media_encryption;
429 const char *linphone_call_state_to_string(LinphoneCallState cs){
431 case LinphoneCallIdle:
432 return "LinphoneCallIdle";
433 case LinphoneCallIncomingReceived:
434 return "LinphoneCallIncomingReceived";
435 case LinphoneCallOutgoingInit:
436 return "LinphoneCallOutgoingInit";
437 case LinphoneCallOutgoingProgress:
438 return "LinphoneCallOutgoingProgress";
439 case LinphoneCallOutgoingRinging:
440 return "LinphoneCallOutgoingRinging";
441 case LinphoneCallOutgoingEarlyMedia:
442 return "LinphoneCallOutgoingEarlyMedia";
443 case LinphoneCallConnected:
444 return "LinphoneCallConnected";
445 case LinphoneCallStreamsRunning:
446 return "LinphoneCallStreamsRunning";
447 case LinphoneCallPausing:
448 return "LinphoneCallPausing";
449 case LinphoneCallPaused:
450 return "LinphoneCallPaused";
451 case LinphoneCallResuming:
452 return "LinphoneCallResuming";
453 case LinphoneCallRefered:
454 return "LinphoneCallRefered";
455 case LinphoneCallError:
456 return "LinphoneCallError";
457 case LinphoneCallEnd:
458 return "LinphoneCallEnd";
459 case LinphoneCallPausedByRemote:
460 return "LinphoneCallPausedByRemote";
461 case LinphoneCallUpdatedByRemote:
462 return "LinphoneCallUpdatedByRemote";
463 case LinphoneCallIncomingEarlyMedia:
464 return "LinphoneCallIncomingEarlyMedia";
465 case LinphoneCallUpdated:
466 return "LinphoneCallUpdated";
467 case LinphoneCallReleased:
468 return "LinphoneCallReleased";
470 return "undefined state";
473 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
474 LinphoneCore *lc=call->core;
476 if (call->state!=cstate){
477 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
478 if (cstate!=LinphoneCallReleased){
479 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
480 linphone_call_state_to_string(cstate));
484 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
485 linphone_call_state_to_string(cstate));
486 if (cstate!=LinphoneCallRefered){
487 /*LinphoneCallRefered is rather an event, not a state.
488 Indeed it does not change the state of the call (still paused or running)*/
491 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
492 switch(call->reason){
493 case LinphoneReasonDeclined:
494 call->log->status=LinphoneCallDeclined;
495 case LinphoneReasonNotAnswered:
496 call->log->status=LinphoneCallMissed;
501 linphone_call_set_terminated (call);
503 if (cstate == LinphoneCallConnected) {
504 call->log->status=LinphoneCallSuccess;
505 call->media_start_time=time(NULL);
508 if (lc->vtable.call_state_changed)
509 lc->vtable.call_state_changed(lc,call,cstate,message);
510 if (cstate==LinphoneCallReleased){
511 if (call->op!=NULL) {
512 /* so that we cannot have anymore upcalls for SAL
513 concerning this call*/
514 sal_op_release(call->op);
517 linphone_call_unref(call);
522 static void linphone_call_destroy(LinphoneCall *obj)
525 sal_op_release(obj->op);
528 if (obj->resultdesc!=NULL) {
529 sal_media_description_unref(obj->resultdesc);
530 obj->resultdesc=NULL;
532 if (obj->localdesc!=NULL) {
533 sal_media_description_unref(obj->localdesc);
537 sal_op_release(obj->ping_op);
540 ms_free(obj->refer_to);
542 if (obj->owns_call_log)
543 linphone_call_log_destroy(obj->log);
544 if (obj->auth_token) {
545 ms_free(obj->auth_token);
552 * @addtogroup call_control
557 * Increments the call 's reference count.
558 * An application that wishes to retain a pointer to call object
559 * must use this function to unsure the pointer remains
560 * valid. Once the application no more needs this pointer,
561 * it must call linphone_call_unref().
563 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
569 * Decrements the call object reference count.
570 * See linphone_call_ref().
572 void linphone_call_unref(LinphoneCall *obj){
575 linphone_call_destroy(obj);
580 * Returns current parameters associated to the call.
582 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
583 return &call->current_params;
586 static bool_t is_video_active(const SalStreamDescription *sd){
587 return sd->port!=0 && sd->dir!=SalStreamInactive;
591 * Returns call parameters proposed by remote.
593 * This is useful when receiving an incoming call, to know whether the remote party
594 * supports video, encryption or whatever.
596 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
597 LinphoneCallParams *cp=&call->remote_params;
598 memset(cp,0,sizeof(*cp));
600 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
602 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
604 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
605 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
606 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
607 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
609 cp->has_video=is_video_active(secure_vsd);
610 if (secure_asd || asd==NULL)
611 cp->media_encryption=LinphoneMediaEncryptionSRTP;
613 cp->has_video=is_video_active(vsd);
622 * Returns the remote address associated to this call
625 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
626 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
630 * Returns the remote address associated to this call as a string.
632 * The result string must be freed by user using ms_free().
634 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
635 return linphone_address_as_string(linphone_call_get_remote_address(call));
639 * Retrieves the call's current state.
641 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
646 * Returns the reason for a call termination (either error or normal termination)
648 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
653 * Get the user_pointer in the LinphoneCall
655 * @ingroup call_control
657 * return user_pointer an opaque user pointer that can be retrieved at any time
659 void *linphone_call_get_user_pointer(LinphoneCall *call)
661 return call->user_pointer;
665 * Set the user_pointer in the LinphoneCall
667 * @ingroup call_control
669 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
671 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
673 call->user_pointer = user_pointer;
677 * Returns the call log associated to this call.
679 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
684 * Returns the refer-to uri (if the call was transfered).
686 const char *linphone_call_get_refer_to(const LinphoneCall *call){
687 return call->refer_to;
691 * Returns direction of the call (incoming or outgoing).
693 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
694 return call->log->dir;
698 * Returns the far end's user agent description string, if available.
700 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
702 return sal_op_get_remote_ua (call->op);
708 * Returns true if this calls has received a transfer that has not been
710 * Pending transfers are executed when this call is being paused or closed,
711 * locally or by remote endpoint.
712 * If the call is already paused while receiving the transfer request, the
713 * transfer immediately occurs.
715 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
716 return call->refer_pending;
720 * Returns call's duration in seconds.
722 int linphone_call_get_duration(const LinphoneCall *call){
723 if (call->media_start_time==0) return 0;
724 return time(NULL)-call->media_start_time;
728 * Returns the call object this call is replacing, if any.
729 * Call replacement can occur during call transfers.
730 * By default, the core automatically terminates the replaced call and accept the new one.
731 * This function allows the application to know whether a new incoming call is a one that replaces another one.
733 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
734 SalOp *op=sal_call_get_replaces(call->op);
736 return (LinphoneCall*)sal_op_get_user_pointer(op);
742 * Indicate whether camera input should be sent to remote end.
744 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
746 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
747 LinphoneCore *lc=call->core;
748 MSWebCam *nowebcam=get_nowebcam_device();
749 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
750 video_stream_change_camera(call->videostream,
751 enable ? lc->video_conf.device : nowebcam);
754 call->camera_active=enable;
759 * Take a photo of currently received video and write it into a jpeg file.
761 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
763 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
764 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
766 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
773 * Returns TRUE if camera pictures are sent to the remote party.
775 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
776 return call->camera_active;
780 * Enable video stream.
782 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
783 cp->has_video=enabled;
786 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
787 return cp->audio_codec;
790 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
791 return cp->video_codec;
795 * Returns whether video is enabled.
797 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
798 return cp->has_video;
801 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
802 return cp->media_encryption;
805 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
806 cp->media_encryption = e;
811 * Enable sending of real early media (during outgoing calls).
813 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
814 cp->real_early_media=enabled;
817 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
818 return cp->real_early_media;
822 * Returns true if the call is part of the locally managed conference.
824 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
825 return cp->in_conference;
829 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
830 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
832 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
833 cp->audio_bw=bandwidth;
838 * Request remote side to send us a Video Fast Update.
840 void linphone_call_send_vfu_request(LinphoneCall *call)
842 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
843 sal_call_send_vfu_request(call->op);
850 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
851 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
852 memcpy(ncp,cp,sizeof(LinphoneCallParams));
859 void linphone_call_params_destroy(LinphoneCallParams *p){
868 #ifdef TEST_EXT_RENDERER
869 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
870 ms_message("rendercb, local buffer=%p, remote buffer=%p",
871 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
876 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
877 LinphoneCall* call = (LinphoneCall*) user_pointer;
878 ms_warning("In linphonecall.c: video_stream_event_cb");
880 case MS_VIDEO_DECODER_DECODING_ERRORS:
881 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
882 linphone_call_send_vfu_request(call);
884 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
885 ms_message("First video frame decoded successfully");
886 if (call->nextVideoFrameDecoded._func != NULL)
887 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
890 ms_warning("Unhandled event %i", event_id);
896 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
897 call->nextVideoFrameDecoded._func = cb;
898 call->nextVideoFrameDecoded._user_data = user_data;
900 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
904 void linphone_call_init_media_streams(LinphoneCall *call){
905 LinphoneCore *lc=call->core;
906 SalMediaDescription *md=call->localdesc;
907 AudioStream *audiostream;
909 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
910 if (linphone_core_echo_limiter_enabled(lc)){
911 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
912 if (strcasecmp(type,"mic")==0)
913 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
914 else if (strcasecmp(type,"full")==0)
915 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
917 audio_stream_enable_gain_control(audiostream,TRUE);
918 if (linphone_core_echo_cancellation_enabled(lc)){
919 int len,delay,framesize;
920 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
921 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
922 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
923 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
924 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
925 if (statestr && audiostream->ec){
926 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
929 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
931 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
932 audio_stream_enable_noise_gate(audiostream,enabled);
936 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
937 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
938 rtp_session_set_transports(audiostream->session,artp,artcp);
941 call->audiostream_app_evq = ortp_ev_queue_new();
942 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
946 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
947 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
948 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
949 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
950 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
952 if( lc->video_conf.displaytype != NULL)
953 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
954 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
956 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
957 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
958 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
960 call->videostream_app_evq = ortp_ev_queue_new();
961 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
962 #ifdef TEST_EXT_RENDERER
963 video_stream_set_render_callback(call->videostream,rendercb,NULL);
967 call->videostream=NULL;
972 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
974 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
975 LinphoneCore* lc = (LinphoneCore*)user_data;
976 if (dtmf<0 || dtmf>15){
977 ms_warning("Bad dtmf value %i",dtmf);
980 if (lc->vtable.dtmf_received != NULL)
981 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
984 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
986 MSFilter *f=st->equalizer;
987 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
988 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
989 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
995 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
996 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
997 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1006 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1007 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1010 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1011 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1012 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1015 audio_stream_set_mic_gain(st,mic_gain);
1017 audio_stream_set_mic_gain(st,0);
1019 recv_gain = lc->sound_conf.soft_play_lev;
1020 if (recv_gain != 0) {
1021 linphone_core_set_playback_gain_db (lc,recv_gain);
1025 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1026 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1027 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1028 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1029 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1030 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1033 if (speed==-1) speed=0.03;
1034 if (force==-1) force=25;
1035 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1036 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1038 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1040 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1041 if (transmit_thres!=-1)
1042 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1044 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1045 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1048 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1049 float floorgain = 1/mic_gain;
1050 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1051 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1052 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1053 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1055 parametrize_equalizer(lc,st);
1058 static void post_configure_audio_streams(LinphoneCall*call){
1059 AudioStream *st=call->audiostream;
1060 LinphoneCore *lc=call->core;
1061 _post_configure_audio_stream(st,lc,call->audio_muted);
1062 if (lc->vtable.dtmf_received!=NULL){
1063 /* replace by our default action*/
1064 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1065 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1069 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1072 RtpProfile *prof=rtp_profile_new("Call profile");
1075 LinphoneCore *lc=call->core;
1079 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1080 PayloadType *pt=(PayloadType*)elem->data;
1083 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1084 if (desc->type==SalAudio){
1085 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1086 up_ptime=linphone_core_get_upload_ptime(lc);
1088 *used_pt=payload_type_get_number(pt);
1091 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1092 else if (md->bandwidth>0) {
1093 /*case where b=AS is given globally, not per stream*/
1094 remote_bw=md->bandwidth;
1095 if (desc->type==SalVideo){
1096 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1100 if (desc->type==SalAudio){
1101 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1102 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1103 if (bw>0) pt->normal_bitrate=bw*1000;
1104 else if (desc->type==SalAudio){
1105 pt->normal_bitrate=-1;
1108 up_ptime=desc->ptime;
1112 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1113 payload_type_append_send_fmtp(pt,tmp);
1115 number=payload_type_get_number(pt);
1116 if (rtp_profile_get_payload(prof,number)!=NULL){
1117 ms_warning("A payload type with number %i already exists in profile !",number);
1119 rtp_profile_set_payload(prof,number,pt);
1125 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1126 int pause_time=3000;
1127 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1128 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1131 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1133 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1134 LinphoneCore *lc=call->core;
1135 LinphoneCall *current=linphone_core_get_current_call(lc);
1136 return !linphone_core_is_in_conference(lc) &&
1137 (current==NULL || current==call);
1139 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1141 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1142 if (crypto[i].tag == tag) {
1148 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1149 LinphoneCore *lc=call->core;
1150 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1152 /* look for savp stream first */
1153 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1154 SalProtoRtpSavp,SalAudio);
1155 /* no savp audio stream, use avp */
1157 stream=sal_media_description_find_stream(call->resultdesc,
1158 SalProtoRtpAvp,SalAudio);
1160 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1161 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1162 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1163 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1164 const char *playfile=lc->play_file;
1165 const char *recfile=lc->rec_file;
1166 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1170 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1171 if (playcard==NULL) {
1172 ms_warning("No card defined for playback !");
1174 if (captcard==NULL) {
1175 ms_warning("No card defined for capture !");
1177 /*Replace soundcard filters by inactive file players or recorders
1178 when placed in recvonly or sendonly mode*/
1179 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1182 }else if (stream->dir==SalStreamSendOnly){
1186 /*And we will eventually play "playfile" if set by the user*/
1189 if (send_ringbacktone){
1191 playfile=NULL;/* it is setup later*/
1193 /*if playfile are supplied don't use soundcards*/
1194 if (lc->use_files) {
1198 if (call->params.in_conference){
1199 /* first create the graph without soundcard resources*/
1200 captcard=playcard=NULL;
1202 if (!linphone_call_sound_resources_available(call)){
1203 ms_message("Sound resources are used by another call, not using soundcard.");
1204 captcard=playcard=NULL;
1206 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1207 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1208 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1209 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1210 audio_stream_start_full(
1212 call->audio_profile,
1213 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1215 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1224 post_configure_audio_streams(call);
1225 if (muted && !send_ringbacktone){
1226 audio_stream_set_mic_gain(call->audiostream,0);
1228 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1230 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1232 if (send_ringbacktone){
1233 setup_ring_player(lc,call);
1235 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1237 /* valid local tags are > 0 */
1238 if (stream->proto == SalProtoRtpSavp) {
1239 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1240 SalProtoRtpSavp,SalAudio);
1241 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1243 if (crypto_idx >= 0) {
1244 audio_stream_enable_strp(
1246 stream->crypto[0].algo,
1247 local_st_desc->crypto[crypto_idx].master_key,
1248 stream->crypto[0].master_key);
1249 call->audiostream_encrypted=TRUE;
1251 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1252 call->audiostream_encrypted=FALSE;
1254 }else call->audiostream_encrypted=FALSE;
1255 if (call->params.in_conference){
1256 /*transform the graph to connect it to the conference filter */
1257 bool_t mute=stream->dir==SalStreamRecvOnly;
1258 linphone_call_add_to_conf(call, mute);
1260 call->current_params.in_conference=call->params.in_conference;
1261 }else ms_warning("No audio stream accepted ?");
1265 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1266 #ifdef VIDEO_ENABLED
1267 LinphoneCore *lc=call->core;
1269 /* look for savp stream first */
1270 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1271 SalProtoRtpSavp,SalVideo);
1272 /* no savp audio stream, use avp */
1274 vstream=sal_media_description_find_stream(call->resultdesc,
1275 SalProtoRtpAvp,SalVideo);
1277 /* shutdown preview */
1278 if (lc->previewstream!=NULL) {
1279 video_preview_stop(lc->previewstream);
1280 lc->previewstream=NULL;
1283 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1284 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1285 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1287 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1288 VideoStreamDir dir=VideoStreamSendRecv;
1289 MSWebCam *cam=lc->video_conf.device;
1290 bool_t is_inactive=FALSE;
1292 call->current_params.has_video=TRUE;
1294 video_stream_enable_adaptive_bitrate_control(call->videostream,
1295 linphone_core_adaptive_rate_control_enabled(lc));
1296 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1297 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1298 if (lc->video_window_id!=0)
1299 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1300 if (lc->preview_window_id!=0)
1301 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1302 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1304 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1305 cam=get_nowebcam_device();
1306 dir=VideoStreamSendOnly;
1307 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1308 dir=VideoStreamRecvOnly;
1309 }else if (vstream->dir==SalStreamSendRecv){
1310 if (lc->video_conf.display && lc->video_conf.capture)
1311 dir=VideoStreamSendRecv;
1312 else if (lc->video_conf.display)
1313 dir=VideoStreamRecvOnly;
1315 dir=VideoStreamSendOnly;
1317 ms_warning("video stream is inactive.");
1318 /*either inactive or incompatible with local capabilities*/
1321 if (call->camera_active==FALSE || all_inputs_muted){
1322 cam=get_nowebcam_device();
1325 call->log->video_enabled = TRUE;
1326 video_stream_set_direction (call->videostream, dir);
1327 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1328 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1329 video_stream_start(call->videostream,
1330 call->video_profile, addr, vstream->port,
1331 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1332 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1333 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1336 if (vstream->proto == SalProtoRtpSavp) {
1337 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1338 SalProtoRtpSavp,SalVideo);
1340 video_stream_enable_strp(
1342 vstream->crypto[0].algo,
1343 local_st_desc->crypto[0].master_key,
1344 vstream->crypto[0].master_key
1346 call->videostream_encrypted=TRUE;
1348 call->videostream_encrypted=FALSE;
1350 }else ms_warning("No video stream accepted.");
1352 ms_warning("No valid video stream defined.");
1357 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1358 LinphoneCore *lc=call->core;
1360 call->current_params.audio_codec = NULL;
1361 call->current_params.video_codec = NULL;
1363 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1365 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1366 #ifdef VIDEO_ENABLED
1367 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1368 SalProtoRtpAvp,SalVideo);
1371 if(call->audiostream == NULL)
1373 ms_fatal("start_media_stream() called without prior init !");
1376 cname=linphone_address_as_string_uri_only(me);
1378 #if defined(VIDEO_ENABLED)
1379 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1380 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1384 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1385 call->current_params.has_video=FALSE;
1386 if (call->videostream!=NULL) {
1387 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1390 call->all_muted=all_inputs_muted;
1391 call->playing_ringbacktone=send_ringbacktone;
1392 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1394 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1395 OrtpZrtpParams params;
1396 /*will be set later when zrtp is activated*/
1397 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1399 params.zid_file=lc->zrtp_secrets_cache;
1400 audio_stream_enable_zrtp(call->audiostream,¶ms);
1401 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1402 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1403 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1406 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1407 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1408 linphone_call_fix_call_parameters(call);
1413 linphone_address_destroy(me);
1416 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1417 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1418 log->quality=audio_stream_get_average_quality_rating(st);
1421 void linphone_call_stop_media_streams(LinphoneCall *call){
1422 if (call->audiostream!=NULL) {
1423 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1424 ortp_ev_queue_flush(call->audiostream_app_evq);
1425 ortp_ev_queue_destroy(call->audiostream_app_evq);
1427 if (call->audiostream->ec){
1428 const char *state_str=NULL;
1429 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1431 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1432 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1435 linphone_call_log_fill_stats (call->log,call->audiostream);
1436 if (call->endpoint){
1437 linphone_call_remove_from_conf(call);
1439 audio_stream_stop(call->audiostream);
1440 call->audiostream=NULL;
1444 #ifdef VIDEO_ENABLED
1445 if (call->videostream!=NULL){
1446 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1447 ortp_ev_queue_flush(call->videostream_app_evq);
1448 ortp_ev_queue_destroy(call->videostream_app_evq);
1449 video_stream_stop(call->videostream);
1450 call->videostream=NULL;
1453 ms_event_queue_skip(call->core->msevq);
1455 if (call->audio_profile){
1456 rtp_profile_clear_all(call->audio_profile);
1457 rtp_profile_destroy(call->audio_profile);
1458 call->audio_profile=NULL;
1460 if (call->video_profile){
1461 rtp_profile_clear_all(call->video_profile);
1462 rtp_profile_destroy(call->video_profile);
1463 call->video_profile=NULL;
1469 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1470 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1471 bool_t bypass_mode = !enable;
1472 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1475 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1476 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1478 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1481 return linphone_core_echo_cancellation_enabled(call->core);
1485 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1486 if (call!=NULL && call->audiostream!=NULL ) {
1488 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1489 if (strcasecmp(type,"mic")==0)
1490 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1491 else if (strcasecmp(type,"full")==0)
1492 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1494 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1499 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1500 if (call!=NULL && call->audiostream!=NULL ){
1501 return call->audiostream->el_type !=ELInactive ;
1503 return linphone_core_echo_limiter_enabled(call->core);
1508 * @addtogroup call_misc
1513 * Returns the measured sound volume played locally (received from remote).
1514 * It is expressed in dbm0.
1516 float linphone_call_get_play_volume(LinphoneCall *call){
1517 AudioStream *st=call->audiostream;
1518 if (st && st->volrecv){
1520 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1524 return LINPHONE_VOLUME_DB_LOWEST;
1528 * Returns the measured sound volume recorded locally (sent to remote).
1529 * It is expressed in dbm0.
1531 float linphone_call_get_record_volume(LinphoneCall *call){
1532 AudioStream *st=call->audiostream;
1533 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1535 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1539 return LINPHONE_VOLUME_DB_LOWEST;
1543 * Obtain real-time quality rating of the call
1545 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1546 * during all the duration of the call. This function returns its value at the time of the function call.
1547 * It is expected that the rating is updated at least every 5 seconds or so.
1548 * The rating is a floating point number comprised between 0 and 5.
1550 * 4-5 = good quality <br>
1551 * 3-4 = average quality <br>
1552 * 2-3 = poor quality <br>
1553 * 1-2 = very poor quality <br>
1554 * 0-1 = can't be worse, mostly unusable <br>
1556 * @returns The function returns -1 if no quality measurement is available, for example if no
1557 * active audio stream exist. Otherwise it returns the quality rating.
1559 float linphone_call_get_current_quality(LinphoneCall *call){
1560 if (call->audiostream){
1561 return audio_stream_get_quality_rating(call->audiostream);
1567 * Returns call quality averaged over all the duration of the call.
1569 * See linphone_call_get_current_quality() for more details about quality measurement.
1571 float linphone_call_get_average_quality(LinphoneCall *call){
1572 if (call->audiostream){
1573 return audio_stream_get_average_quality_rating(call->audiostream);
1579 * Access last known statistics for audio stream, for a given call.
1581 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1582 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1586 * Access last known statistics for video stream, for a given call.
1588 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1589 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1597 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1598 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1599 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1600 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1601 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1602 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1605 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1609 from = linphone_call_get_remote_address_as_string(call);
1612 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1617 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1619 if (lc->vtable.display_warning!=NULL)
1620 lc->vtable.display_warning(lc,temp);
1621 linphone_core_terminate_call(lc,call);
1624 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1625 LinphoneCore* lc = call->core;
1626 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1627 bool_t disconnected=FALSE;
1629 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1630 RtpSession *as=NULL,*vs=NULL;
1631 float audio_load=0, video_load=0;
1632 if (call->audiostream!=NULL){
1633 as=call->audiostream->session;
1634 if (call->audiostream->ticker)
1635 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1637 if (call->videostream!=NULL){
1638 if (call->videostream->ticker)
1639 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1640 vs=call->videostream->session;
1642 display_bandwidth(as,vs);
1643 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1645 #ifdef VIDEO_ENABLED
1646 if (call->videostream!=NULL) {
1647 // Beware that the application queue should not depend on treatments fron the
1648 // mediastreamer queue.
1649 video_stream_iterate(call->videostream);
1651 if (call->videostream_app_evq){
1653 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1654 OrtpEventType evt=ortp_event_get_type(ev);
1655 OrtpEventData *evd=ortp_event_get_data(ev);
1656 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1657 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1658 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1659 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1660 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1661 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1662 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1664 if (lc->vtable.call_stats_updated)
1665 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1666 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1667 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1668 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1669 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1670 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1672 if (lc->vtable.call_stats_updated)
1673 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1675 ortp_event_destroy(ev);
1680 if (call->audiostream!=NULL) {
1681 // Beware that the application queue should not depend on treatments fron the
1682 // mediastreamer queue.
1683 audio_stream_iterate(call->audiostream);
1685 if (call->audiostream_app_evq){
1687 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1688 OrtpEventType evt=ortp_event_get_type(ev);
1689 OrtpEventData *evd=ortp_event_get_data(ev);
1690 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1691 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1692 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1693 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1694 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1695 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1696 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1697 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1698 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1700 if (lc->vtable.call_stats_updated)
1701 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1702 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1703 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1704 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1705 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1706 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1708 if (lc->vtable.call_stats_updated)
1709 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1711 ortp_event_destroy(ev);
1715 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1716 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1718 linphone_core_disconnected(call->core,call);
1721 void linphone_call_log_completed(LinphoneCall *call){
1722 LinphoneCore *lc=call->core;
1724 call->log->duration=time(NULL)-call->start_time;
1726 if (call->log->status==LinphoneCallMissed){
1729 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1730 "You have missed %i calls.", lc->missed_calls),
1732 if (lc->vtable.display_status!=NULL)
1733 lc->vtable.display_status(lc,info);
1736 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1737 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1738 MSList *elem,*prevelem=NULL;
1739 /*find the last element*/
1740 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1744 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1745 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1747 if (lc->vtable.call_log_updated!=NULL){
1748 lc->vtable.call_log_updated(lc,call->log);
1750 call_logs_write_to_config_file(lc);
1753 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1754 return call->transfer_state;
1757 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1758 if (state != call->transfer_state) {
1759 LinphoneCore* lc = call->core;
1760 call->transfer_state = state;
1761 if (lc->vtable.transfer_state_changed)
1762 lc->vtable.transfer_state_changed(lc, call, state);