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->start_time=time(NULL);
299 call->media_start_time=0;
300 call->log=linphone_call_log_new(call, from, to);
301 call->owns_call_log=TRUE;
302 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
303 port_offset=find_port_offset (call->core);
304 if (port_offset==-1) return;
305 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
306 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
310 static void discover_mtu(LinphoneCore *lc, const char *remote){
312 if (lc->net_conf.mtu==0 ){
313 /*attempt to discover mtu*/
314 mtu=ms_discover_mtu(remote);
317 ms_message("Discovered mtu is %i, RTP payload max size is %i",
318 mtu, ms_get_payload_max_size());
323 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
325 LinphoneCall *call=ms_new0(LinphoneCall,1);
326 call->dir=LinphoneCallOutgoing;
327 call->op=sal_op_new(lc->sal);
328 sal_op_set_user_pointer(call->op,call);
330 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
331 linphone_call_init_common(call,from,to);
332 call->params=*params;
333 call->localdesc=create_local_media_description (lc,call);
334 call->camera_active=params->has_video;
335 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
336 linphone_core_run_stun_tests(call->core,call);
337 discover_mtu(lc,linphone_address_get_domain (to));
338 if (params->referer){
339 sal_call_set_referer(call->op,params->referer->op);
340 call->referer=linphone_call_ref(params->referer);
345 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
346 LinphoneCall *call=ms_new0(LinphoneCall,1);
349 call->dir=LinphoneCallIncoming;
350 sal_op_set_user_pointer(op,call);
354 if (lc->sip_conf.ping_with_options){
355 /*the following sends an option request back to the caller so that
356 we get a chance to discover our nat'd address before answering.*/
357 call->ping_op=sal_op_new(lc->sal);
358 from_str=linphone_address_as_string_uri_only(from);
359 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
360 sal_op_set_user_pointer(call->ping_op,call);
361 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
365 linphone_address_clean(from);
366 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
367 linphone_call_init_common(call, from, to);
368 linphone_core_init_default_params(lc, &call->params);
369 call->params.has_video &= !!lc->video_policy.automatically_accept;
370 call->localdesc=create_local_media_description (lc,call);
371 call->camera_active=call->params.has_video;
372 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
373 linphone_core_run_stun_tests(call->core,call);
374 discover_mtu(lc,linphone_address_get_domain(from));
378 /* this function is called internally to get rid of a call.
379 It performs the following tasks:
380 - remove the call from the internal list of calls
381 - update the call logs accordingly
384 static void linphone_call_set_terminated(LinphoneCall *call){
385 LinphoneCore *lc=call->core;
387 linphone_core_update_allocated_audio_bandwidth(lc);
389 call->owns_call_log=FALSE;
390 linphone_call_log_completed(call);
393 if (call == lc->current_call){
394 ms_message("Resetting the current call");
395 lc->current_call=NULL;
398 if (linphone_core_del_call(lc,call) != 0){
399 ms_error("Could not remove the call from the list !!!");
402 if (ms_list_size(lc->calls)==0)
403 linphone_core_notify_all_friends(lc,lc->presence_mode);
405 linphone_core_conference_check_uninit(lc);
406 if (call->ringing_beep){
407 linphone_core_stop_dtmf(lc);
408 call->ringing_beep=FALSE;
411 linphone_call_unref(call->referer);
416 void linphone_call_fix_call_parameters(LinphoneCall *call){
417 call->params.has_video=call->current_params.has_video;
418 call->params.media_encryption=call->current_params.media_encryption;
421 const char *linphone_call_state_to_string(LinphoneCallState cs){
423 case LinphoneCallIdle:
424 return "LinphoneCallIdle";
425 case LinphoneCallIncomingReceived:
426 return "LinphoneCallIncomingReceived";
427 case LinphoneCallOutgoingInit:
428 return "LinphoneCallOutgoingInit";
429 case LinphoneCallOutgoingProgress:
430 return "LinphoneCallOutgoingProgress";
431 case LinphoneCallOutgoingRinging:
432 return "LinphoneCallOutgoingRinging";
433 case LinphoneCallOutgoingEarlyMedia:
434 return "LinphoneCallOutgoingEarlyMedia";
435 case LinphoneCallConnected:
436 return "LinphoneCallConnected";
437 case LinphoneCallStreamsRunning:
438 return "LinphoneCallStreamsRunning";
439 case LinphoneCallPausing:
440 return "LinphoneCallPausing";
441 case LinphoneCallPaused:
442 return "LinphoneCallPaused";
443 case LinphoneCallResuming:
444 return "LinphoneCallResuming";
445 case LinphoneCallRefered:
446 return "LinphoneCallRefered";
447 case LinphoneCallError:
448 return "LinphoneCallError";
449 case LinphoneCallEnd:
450 return "LinphoneCallEnd";
451 case LinphoneCallPausedByRemote:
452 return "LinphoneCallPausedByRemote";
453 case LinphoneCallUpdatedByRemote:
454 return "LinphoneCallUpdatedByRemote";
455 case LinphoneCallIncomingEarlyMedia:
456 return "LinphoneCallIncomingEarlyMedia";
457 case LinphoneCallUpdated:
458 return "LinphoneCallUpdated";
459 case LinphoneCallReleased:
460 return "LinphoneCallReleased";
462 return "undefined state";
465 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
466 LinphoneCore *lc=call->core;
468 if (call->state!=cstate){
469 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
470 if (cstate!=LinphoneCallReleased){
471 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
472 linphone_call_state_to_string(cstate));
476 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
477 linphone_call_state_to_string(cstate));
478 if (cstate!=LinphoneCallRefered){
479 /*LinphoneCallRefered is rather an event, not a state.
480 Indeed it does not change the state of the call (still paused or running)*/
483 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
484 if (call->reason==LinphoneReasonDeclined){
485 call->log->status=LinphoneCallDeclined;
487 linphone_call_set_terminated (call);
489 if (cstate == LinphoneCallConnected) {
490 call->log->status=LinphoneCallSuccess;
491 call->media_start_time=time(NULL);
494 if (lc->vtable.call_state_changed)
495 lc->vtable.call_state_changed(lc,call,cstate,message);
496 if (cstate==LinphoneCallReleased){
497 if (call->op!=NULL) {
498 /* so that we cannot have anymore upcalls for SAL
499 concerning this call*/
500 sal_op_release(call->op);
503 linphone_call_unref(call);
508 static void linphone_call_destroy(LinphoneCall *obj)
511 sal_op_release(obj->op);
514 if (obj->resultdesc!=NULL) {
515 sal_media_description_unref(obj->resultdesc);
516 obj->resultdesc=NULL;
518 if (obj->localdesc!=NULL) {
519 sal_media_description_unref(obj->localdesc);
523 sal_op_release(obj->ping_op);
526 ms_free(obj->refer_to);
528 if (obj->owns_call_log)
529 linphone_call_log_destroy(obj->log);
530 if (obj->auth_token) {
531 ms_free(obj->auth_token);
538 * @addtogroup call_control
543 * Increments the call 's reference count.
544 * An application that wishes to retain a pointer to call object
545 * must use this function to unsure the pointer remains
546 * valid. Once the application no more needs this pointer,
547 * it must call linphone_call_unref().
549 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
555 * Decrements the call object reference count.
556 * See linphone_call_ref().
558 void linphone_call_unref(LinphoneCall *obj){
561 linphone_call_destroy(obj);
566 * Returns current parameters associated to the call.
568 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
569 return &call->current_params;
572 static bool_t is_video_active(const SalStreamDescription *sd){
573 return sd->port!=0 && sd->dir!=SalStreamInactive;
577 * Returns call parameters proposed by remote.
579 * This is useful when receiving an incoming call, to know whether the remote party
580 * supports video, encryption or whatever.
582 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
583 LinphoneCallParams *cp=&call->remote_params;
584 memset(cp,0,sizeof(*cp));
586 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
588 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
590 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
591 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
592 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
593 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
595 cp->has_video=is_video_active(secure_vsd);
596 if (secure_asd || asd==NULL)
597 cp->media_encryption=LinphoneMediaEncryptionSRTP;
599 cp->has_video=is_video_active(vsd);
608 * Returns the remote address associated to this call
611 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
612 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
616 * Returns the remote address associated to this call as a string.
618 * The result string must be freed by user using ms_free().
620 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
621 return linphone_address_as_string(linphone_call_get_remote_address(call));
625 * Retrieves the call's current state.
627 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
632 * Returns the reason for a call termination (either error or normal termination)
634 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
639 * Get the user_pointer in the LinphoneCall
641 * @ingroup call_control
643 * return user_pointer an opaque user pointer that can be retrieved at any time
645 void *linphone_call_get_user_pointer(LinphoneCall *call)
647 return call->user_pointer;
651 * Set the user_pointer in the LinphoneCall
653 * @ingroup call_control
655 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
657 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
659 call->user_pointer = user_pointer;
663 * Returns the call log associated to this call.
665 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
670 * Returns the refer-to uri (if the call was transfered).
672 const char *linphone_call_get_refer_to(const LinphoneCall *call){
673 return call->refer_to;
677 * Returns direction of the call (incoming or outgoing).
679 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
680 return call->log->dir;
684 * Returns the far end's user agent description string, if available.
686 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
688 return sal_op_get_remote_ua (call->op);
694 * Returns true if this calls has received a transfer that has not been
696 * Pending transfers are executed when this call is being paused or closed,
697 * locally or by remote endpoint.
698 * If the call is already paused while receiving the transfer request, the
699 * transfer immediately occurs.
701 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
702 return call->refer_pending;
706 * Returns call's duration in seconds.
708 int linphone_call_get_duration(const LinphoneCall *call){
709 if (call->media_start_time==0) return 0;
710 return time(NULL)-call->media_start_time;
714 * Returns the call object this call is replacing, if any.
715 * Call replacement can occur during call transfers.
716 * By default, the core automatically terminates the replaced call and accept the new one.
717 * This function allows the application to know whether a new incoming call is a one that replaces another one.
719 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
720 SalOp *op=sal_call_get_replaces(call->op);
722 return (LinphoneCall*)sal_op_get_user_pointer(op);
728 * Indicate whether camera input should be sent to remote end.
730 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
732 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
733 LinphoneCore *lc=call->core;
734 MSWebCam *nowebcam=get_nowebcam_device();
735 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
736 video_stream_change_camera(call->videostream,
737 enable ? lc->video_conf.device : nowebcam);
740 call->camera_active=enable;
745 * Take a photo of currently received video and write it into a jpeg file.
747 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
749 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
750 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
752 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
759 * Returns TRUE if camera pictures are sent to the remote party.
761 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
762 return call->camera_active;
766 * Enable video stream.
768 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
769 cp->has_video=enabled;
773 * Returns whether video is enabled.
775 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
776 return cp->has_video;
779 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
780 return cp->media_encryption;
783 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
784 cp->media_encryption = e;
789 * Enable sending of real early media (during outgoing calls).
791 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
792 cp->real_early_media=enabled;
795 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
796 return cp->real_early_media;
800 * Returns true if the call is part of the locally managed conference.
802 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
803 return cp->in_conference;
807 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
808 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
810 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
811 cp->audio_bw=bandwidth;
816 * Request remote side to send us a Video Fast Update.
818 void linphone_call_send_vfu_request(LinphoneCall *call)
820 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
821 sal_call_send_vfu_request(call->op);
828 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
829 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
830 memcpy(ncp,cp,sizeof(LinphoneCallParams));
837 void linphone_call_params_destroy(LinphoneCallParams *p){
846 #ifdef TEST_EXT_RENDERER
847 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
848 ms_message("rendercb, local buffer=%p, remote buffer=%p",
849 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
854 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
855 LinphoneCall* call = (LinphoneCall*) user_pointer;
856 ms_warning("In linphonecall.c: video_stream_event_cb");
858 case MS_VIDEO_DECODER_DECODING_ERRORS:
859 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
860 linphone_call_send_vfu_request(call);
862 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
863 ms_message("First video frame decoded successfully");
864 if (call->nextVideoFrameDecoded._func != NULL)
865 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
868 ms_warning("Unhandled event %i", event_id);
874 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
875 call->nextVideoFrameDecoded._func = cb;
876 call->nextVideoFrameDecoded._user_data = user_data;
878 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
882 void linphone_call_init_media_streams(LinphoneCall *call){
883 LinphoneCore *lc=call->core;
884 SalMediaDescription *md=call->localdesc;
885 AudioStream *audiostream;
887 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
888 if (linphone_core_echo_limiter_enabled(lc)){
889 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
890 if (strcasecmp(type,"mic")==0)
891 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
892 else if (strcasecmp(type,"full")==0)
893 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
895 audio_stream_enable_gain_control(audiostream,TRUE);
896 if (linphone_core_echo_cancellation_enabled(lc)){
897 int len,delay,framesize;
898 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
899 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
900 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
901 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
902 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
903 if (statestr && audiostream->ec){
904 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
907 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
909 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
910 audio_stream_enable_noise_gate(audiostream,enabled);
914 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
915 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
916 rtp_session_set_transports(audiostream->session,artp,artcp);
919 call->audiostream_app_evq = ortp_ev_queue_new();
920 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
924 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
925 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
926 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
927 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
928 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
930 if( lc->video_conf.displaytype != NULL)
931 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
932 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
934 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
935 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
936 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
938 call->videostream_app_evq = ortp_ev_queue_new();
939 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
940 #ifdef TEST_EXT_RENDERER
941 video_stream_set_render_callback(call->videostream,rendercb,NULL);
945 call->videostream=NULL;
950 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
952 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
953 LinphoneCore* lc = (LinphoneCore*)user_data;
954 if (dtmf<0 || dtmf>15){
955 ms_warning("Bad dtmf value %i",dtmf);
958 if (lc->vtable.dtmf_received != NULL)
959 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
962 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
964 MSFilter *f=st->equalizer;
965 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
966 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
967 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
973 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
974 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
975 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
984 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
985 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
988 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
989 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
990 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
993 audio_stream_set_mic_gain(st,mic_gain);
995 audio_stream_set_mic_gain(st,0);
997 recv_gain = lc->sound_conf.soft_play_lev;
998 if (recv_gain != 0) {
999 linphone_core_set_playback_gain_db (lc,recv_gain);
1003 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1004 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1005 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1006 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1007 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1008 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1011 if (speed==-1) speed=0.03;
1012 if (force==-1) force=25;
1013 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1014 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1016 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1018 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1019 if (transmit_thres!=-1)
1020 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1022 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1023 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1026 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1027 float floorgain = 1/mic_gain;
1028 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1029 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1030 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1031 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1033 parametrize_equalizer(lc,st);
1036 static void post_configure_audio_streams(LinphoneCall*call){
1037 AudioStream *st=call->audiostream;
1038 LinphoneCore *lc=call->core;
1039 _post_configure_audio_stream(st,lc,call->audio_muted);
1040 if (lc->vtable.dtmf_received!=NULL){
1041 /* replace by our default action*/
1042 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1043 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1047 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1050 RtpProfile *prof=rtp_profile_new("Call profile");
1053 LinphoneCore *lc=call->core;
1057 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1058 PayloadType *pt=(PayloadType*)elem->data;
1061 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1062 if (desc->type==SalAudio){
1063 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1064 up_ptime=linphone_core_get_upload_ptime(lc);
1066 *used_pt=payload_type_get_number(pt);
1069 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1070 else if (md->bandwidth>0) {
1071 /*case where b=AS is given globally, not per stream*/
1072 remote_bw=md->bandwidth;
1073 if (desc->type==SalVideo){
1074 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1078 if (desc->type==SalAudio){
1079 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1080 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1081 if (bw>0) pt->normal_bitrate=bw*1000;
1082 else if (desc->type==SalAudio){
1083 pt->normal_bitrate=-1;
1086 up_ptime=desc->ptime;
1090 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1091 payload_type_append_send_fmtp(pt,tmp);
1093 number=payload_type_get_number(pt);
1094 if (rtp_profile_get_payload(prof,number)!=NULL){
1095 ms_warning("A payload type with number %i already exists in profile !",number);
1097 rtp_profile_set_payload(prof,number,pt);
1103 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1104 int pause_time=3000;
1105 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1106 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1109 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1111 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1112 LinphoneCore *lc=call->core;
1113 LinphoneCall *current=linphone_core_get_current_call(lc);
1114 return !linphone_core_is_in_conference(lc) &&
1115 (current==NULL || current==call);
1117 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1119 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1120 if (crypto[i].tag == tag) {
1126 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1127 LinphoneCore *lc=call->core;
1128 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1130 /* look for savp stream first */
1131 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1132 SalProtoRtpSavp,SalAudio);
1133 /* no savp audio stream, use avp */
1135 stream=sal_media_description_find_stream(call->resultdesc,
1136 SalProtoRtpAvp,SalAudio);
1138 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1139 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1140 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1141 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1142 const char *playfile=lc->play_file;
1143 const char *recfile=lc->rec_file;
1144 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1148 if (playcard==NULL) {
1149 ms_warning("No card defined for playback !");
1151 if (captcard==NULL) {
1152 ms_warning("No card defined for capture !");
1154 /*Replace soundcard filters by inactive file players or recorders
1155 when placed in recvonly or sendonly mode*/
1156 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1159 }else if (stream->dir==SalStreamSendOnly){
1163 /*And we will eventually play "playfile" if set by the user*/
1166 if (send_ringbacktone){
1168 playfile=NULL;/* it is setup later*/
1170 /*if playfile are supplied don't use soundcards*/
1171 if (lc->use_files) {
1175 if (call->params.in_conference){
1176 /* first create the graph without soundcard resources*/
1177 captcard=playcard=NULL;
1179 if (!linphone_call_sound_resources_available(call)){
1180 ms_message("Sound resources are used by another call, not using soundcard.");
1181 captcard=playcard=NULL;
1183 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1184 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1185 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1186 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1187 audio_stream_start_full(
1189 call->audio_profile,
1190 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1192 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1201 post_configure_audio_streams(call);
1202 if (muted && !send_ringbacktone){
1203 audio_stream_set_mic_gain(call->audiostream,0);
1205 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1207 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1209 if (send_ringbacktone){
1210 setup_ring_player(lc,call);
1212 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1214 /* valid local tags are > 0 */
1215 if (stream->proto == SalProtoRtpSavp) {
1216 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1217 SalProtoRtpSavp,SalAudio);
1218 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1220 if (crypto_idx >= 0) {
1221 audio_stream_enable_strp(
1223 stream->crypto[0].algo,
1224 local_st_desc->crypto[crypto_idx].master_key,
1225 stream->crypto[0].master_key);
1226 call->audiostream_encrypted=TRUE;
1228 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1229 call->audiostream_encrypted=FALSE;
1231 }else call->audiostream_encrypted=FALSE;
1232 if (call->params.in_conference){
1233 /*transform the graph to connect it to the conference filter */
1234 bool_t mute=stream->dir==SalStreamRecvOnly;
1235 linphone_call_add_to_conf(call, mute);
1237 call->current_params.in_conference=call->params.in_conference;
1238 }else ms_warning("No audio stream accepted ?");
1242 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1243 #ifdef VIDEO_ENABLED
1244 LinphoneCore *lc=call->core;
1246 /* look for savp stream first */
1247 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1248 SalProtoRtpSavp,SalVideo);
1249 /* no savp audio stream, use avp */
1251 vstream=sal_media_description_find_stream(call->resultdesc,
1252 SalProtoRtpAvp,SalVideo);
1254 /* shutdown preview */
1255 if (lc->previewstream!=NULL) {
1256 video_preview_stop(lc->previewstream);
1257 lc->previewstream=NULL;
1260 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1261 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1262 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1264 VideoStreamDir dir=VideoStreamSendRecv;
1265 MSWebCam *cam=lc->video_conf.device;
1266 bool_t is_inactive=FALSE;
1268 call->current_params.has_video=TRUE;
1270 video_stream_enable_adaptive_bitrate_control(call->videostream,
1271 linphone_core_adaptive_rate_control_enabled(lc));
1272 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1273 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1274 if (lc->video_window_id!=0)
1275 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1276 if (lc->preview_window_id!=0)
1277 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1278 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1280 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1281 cam=get_nowebcam_device();
1282 dir=VideoStreamSendOnly;
1283 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1284 dir=VideoStreamRecvOnly;
1285 }else if (vstream->dir==SalStreamSendRecv){
1286 if (lc->video_conf.display && lc->video_conf.capture)
1287 dir=VideoStreamSendRecv;
1288 else if (lc->video_conf.display)
1289 dir=VideoStreamRecvOnly;
1291 dir=VideoStreamSendOnly;
1293 ms_warning("video stream is inactive.");
1294 /*either inactive or incompatible with local capabilities*/
1297 if (call->camera_active==FALSE || all_inputs_muted){
1298 cam=get_nowebcam_device();
1301 call->log->video_enabled = TRUE;
1302 video_stream_set_direction (call->videostream, dir);
1303 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1304 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1305 video_stream_start(call->videostream,
1306 call->video_profile, addr, vstream->port,
1307 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1308 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1309 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1312 if (vstream->proto == SalProtoRtpSavp) {
1313 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1314 SalProtoRtpSavp,SalVideo);
1316 video_stream_enable_strp(
1318 vstream->crypto[0].algo,
1319 local_st_desc->crypto[0].master_key,
1320 vstream->crypto[0].master_key
1322 call->videostream_encrypted=TRUE;
1324 call->videostream_encrypted=FALSE;
1326 }else ms_warning("No video stream accepted.");
1328 ms_warning("No valid video stream defined.");
1333 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1334 LinphoneCore *lc=call->core;
1335 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1337 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1338 #ifdef VIDEO_ENABLED
1339 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1340 SalProtoRtpAvp,SalVideo);
1343 if(call->audiostream == NULL)
1345 ms_fatal("start_media_stream() called without prior init !");
1348 cname=linphone_address_as_string_uri_only(me);
1350 #if defined(VIDEO_ENABLED)
1351 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1352 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1356 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1357 call->current_params.has_video=FALSE;
1358 if (call->videostream!=NULL) {
1359 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1362 call->all_muted=all_inputs_muted;
1363 call->playing_ringbacktone=send_ringbacktone;
1364 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1366 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1367 OrtpZrtpParams params;
1368 /*will be set later when zrtp is activated*/
1369 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1371 params.zid_file=lc->zrtp_secrets_cache;
1372 audio_stream_enable_zrtp(call->audiostream,¶ms);
1373 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1374 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1375 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1378 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1379 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1380 linphone_call_fix_call_parameters(call);
1385 linphone_address_destroy(me);
1388 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1389 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1390 log->quality=audio_stream_get_average_quality_rating(st);
1393 void linphone_call_stop_media_streams(LinphoneCall *call){
1394 if (call->audiostream!=NULL) {
1395 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1396 ortp_ev_queue_flush(call->audiostream_app_evq);
1397 ortp_ev_queue_destroy(call->audiostream_app_evq);
1399 if (call->audiostream->ec){
1400 const char *state_str=NULL;
1401 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1403 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1404 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1407 linphone_call_log_fill_stats (call->log,call->audiostream);
1408 if (call->endpoint){
1409 linphone_call_remove_from_conf(call);
1411 audio_stream_stop(call->audiostream);
1412 call->audiostream=NULL;
1416 #ifdef VIDEO_ENABLED
1417 if (call->videostream!=NULL){
1418 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1419 ortp_ev_queue_flush(call->videostream_app_evq);
1420 ortp_ev_queue_destroy(call->videostream_app_evq);
1421 video_stream_stop(call->videostream);
1422 call->videostream=NULL;
1425 ms_event_queue_skip(call->core->msevq);
1427 if (call->audio_profile){
1428 rtp_profile_clear_all(call->audio_profile);
1429 rtp_profile_destroy(call->audio_profile);
1430 call->audio_profile=NULL;
1432 if (call->video_profile){
1433 rtp_profile_clear_all(call->video_profile);
1434 rtp_profile_destroy(call->video_profile);
1435 call->video_profile=NULL;
1441 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1442 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1443 bool_t bypass_mode = !enable;
1444 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1447 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1448 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1450 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1453 return linphone_core_echo_cancellation_enabled(call->core);
1457 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1458 if (call!=NULL && call->audiostream!=NULL ) {
1460 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1461 if (strcasecmp(type,"mic")==0)
1462 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1463 else if (strcasecmp(type,"full")==0)
1464 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1466 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1471 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1472 if (call!=NULL && call->audiostream!=NULL ){
1473 return call->audiostream->el_type !=ELInactive ;
1475 return linphone_core_echo_limiter_enabled(call->core);
1480 * @addtogroup call_misc
1485 * Returns the measured sound volume played locally (received from remote)
1486 * It is expressed in dbm0.
1488 float linphone_call_get_play_volume(LinphoneCall *call){
1489 AudioStream *st=call->audiostream;
1490 if (st && st->volrecv){
1492 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1496 return LINPHONE_VOLUME_DB_LOWEST;
1500 * Returns the measured sound volume recorded locally (sent to remote)
1501 * It is expressed in dbm0.
1503 float linphone_call_get_record_volume(LinphoneCall *call){
1504 AudioStream *st=call->audiostream;
1505 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1507 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1511 return LINPHONE_VOLUME_DB_LOWEST;
1515 * Obtain real-time quality rating of the call
1517 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1518 * during all the duration of the call. This function returns its value at the time of the function call.
1519 * It is expected that the rating is updated at least every 5 seconds or so.
1520 * The rating is a floating point number comprised between 0 and 5.
1522 * 4-5 = good quality <br>
1523 * 3-4 = average quality <br>
1524 * 2-3 = poor quality <br>
1525 * 1-2 = very poor quality <br>
1526 * 0-1 = can't be worse, mostly unusable <br>
1528 * @returns The function returns -1 if no quality measurement is available, for example if no
1529 * active audio stream exist. Otherwise it returns the quality rating.
1531 float linphone_call_get_current_quality(LinphoneCall *call){
1532 if (call->audiostream){
1533 return audio_stream_get_quality_rating(call->audiostream);
1539 * Returns call quality averaged over all the duration of the call.
1541 * See linphone_call_get_current_quality() for more details about quality measurement.
1543 float linphone_call_get_average_quality(LinphoneCall *call){
1544 if (call->audiostream){
1545 return audio_stream_get_average_quality_rating(call->audiostream);
1554 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1555 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1556 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1557 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1558 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1559 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1562 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1566 from = linphone_call_get_remote_address_as_string(call);
1569 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1574 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1576 if (lc->vtable.display_warning!=NULL)
1577 lc->vtable.display_warning(lc,temp);
1578 linphone_core_terminate_call(lc,call);
1581 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1582 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1583 bool_t disconnected=FALSE;
1585 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1586 RtpSession *as=NULL,*vs=NULL;
1587 float audio_load=0, video_load=0;
1588 if (call->audiostream!=NULL){
1589 as=call->audiostream->session;
1590 if (call->audiostream->ticker)
1591 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1593 if (call->videostream!=NULL){
1594 if (call->videostream->ticker)
1595 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1596 vs=call->videostream->session;
1598 display_bandwidth(as,vs);
1599 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1601 #ifdef VIDEO_ENABLED
1602 if (call->videostream!=NULL) {
1603 // Beware that the application queue should not depend on treatments fron the
1604 // mediastreamer queue.
1605 video_stream_iterate(call->videostream);
1607 if (call->videostream_app_evq){
1609 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1610 OrtpEventType evt=ortp_event_get_type(ev);
1611 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1612 OrtpEventData *evd=ortp_event_get_data(ev);
1613 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1615 ortp_event_destroy(ev);
1620 if (call->audiostream!=NULL) {
1621 // Beware that the application queue should not depend on treatments fron the
1622 // mediastreamer queue.
1623 audio_stream_iterate(call->audiostream);
1625 if (call->audiostream_app_evq){
1627 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1628 OrtpEventType evt=ortp_event_get_type(ev);
1629 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1630 OrtpEventData *evd=ortp_event_get_data(ev);
1631 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1632 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1633 OrtpEventData *evd=ortp_event_get_data(ev);
1634 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1636 ortp_event_destroy(ev);
1640 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1641 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1643 linphone_core_disconnected(call->core,call);
1646 void linphone_call_log_completed(LinphoneCall *call){
1647 LinphoneCore *lc=call->core;
1649 call->log->duration=time(NULL)-call->start_time;
1651 if (call->log->status==LinphoneCallMissed){
1654 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1655 "You have missed %i calls.", lc->missed_calls),
1657 if (lc->vtable.display_status!=NULL)
1658 lc->vtable.display_status(lc,info);
1661 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1662 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1663 MSList *elem,*prevelem=NULL;
1664 /*find the last element*/
1665 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1669 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1670 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1672 if (lc->vtable.call_log_updated!=NULL){
1673 lc->vtable.call_log_updated(lc,call->log);
1675 call_logs_write_to_config_file(lc);