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();
205 md->session_id=session_id;
206 md->session_ver=session_ver;
208 strncpy(md->addr,call->localip,sizeof(md->addr));
209 strncpy(md->username,username,sizeof(md->username));
210 md->bandwidth=linphone_core_get_download_bandwidth(lc);
212 /*set audio capabilities */
213 strncpy(md->streams[0].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
214 strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
215 md->streams[0].rtp_port=call->audio_port;
216 md->streams[0].rtcp_port=call->audio_port+1;
217 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
218 SalProtoRtpSavp : SalProtoRtpAvp;
219 md->streams[0].type=SalAudio;
220 md->streams[0].ptime=lc->net_conf.down_ptime;
221 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
222 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
223 l=ms_list_append(l,pt);
224 md->streams[0].payloads=l;
228 if (call->params.has_video){
230 md->streams[1].rtp_port=call->video_port;
231 md->streams[1].rtcp_port=call->video_port+1;
232 md->streams[1].proto=md->streams[0].proto;
233 md->streams[1].type=SalVideo;
234 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
235 md->streams[1].payloads=l;
238 for(i=0; i<md->nstreams; i++) {
239 if (md->streams[i].proto == SalProtoRtpSavp) {
240 md->streams[i].crypto[0].tag = 1;
241 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
242 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
243 md->streams[i].crypto[0].algo = 0;
244 md->streams[i].crypto[1].tag = 2;
245 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
246 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
247 md->streams[i].crypto[1].algo = 0;
248 md->streams[i].crypto[2].algo = 0;
250 if ((linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, i) == NULL)) {
251 ice_session_add_check_list(call->ice_session, ice_check_list_new());
255 linphone_address_destroy(addr);
259 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
260 SalMediaDescription *md=call->localdesc;
262 call->localdesc = create_local_media_description(lc,call);
264 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
265 sal_media_description_unref(md);
269 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
270 unsigned int id=rand() & 0xfff;
271 return _create_local_media_description(lc,call,id,id);
274 static int find_port_offset(LinphoneCore *lc){
278 bool_t already_used=FALSE;
279 for(offset=0;offset<100;offset+=2){
280 audio_port=linphone_core_get_audio_port (lc)+offset;
282 for(elem=lc->calls;elem!=NULL;elem=elem->next){
283 LinphoneCall *call=(LinphoneCall*)elem->data;
284 if (call->audio_port==audio_port) {
289 if (!already_used) break;
292 ms_error("Could not find any free port !");
298 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
300 call->magic=linphone_call_magic;
302 call->state=LinphoneCallIdle;
303 call->transfer_state = LinphoneCallIdle;
304 call->start_time=time(NULL);
305 call->media_start_time=0;
306 call->log=linphone_call_log_new(call, from, to);
307 call->owns_call_log=TRUE;
308 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
309 port_offset=find_port_offset (call->core);
310 if (port_offset==-1) return;
311 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
312 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
313 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
314 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
317 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
319 stats->received_rtcp = NULL;
320 stats->sent_rtcp = NULL;
323 static void discover_mtu(LinphoneCore *lc, const char *remote){
325 if (lc->net_conf.mtu==0 ){
326 /*attempt to discover mtu*/
327 mtu=ms_discover_mtu(remote);
330 ms_message("Discovered mtu is %i, RTP payload max size is %i",
331 mtu, ms_get_payload_max_size());
336 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
338 LinphoneCall *call=ms_new0(LinphoneCall,1);
339 call->dir=LinphoneCallOutgoing;
340 call->op=sal_op_new(lc->sal);
341 sal_op_set_user_pointer(call->op,call);
343 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
344 linphone_call_init_common(call,from,to);
345 call->params=*params;
346 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
347 call->ice_session = ice_session_new();
348 ice_session_set_role(call->ice_session, IR_Controlling);
350 call->localdesc=create_local_media_description (lc,call);
351 call->camera_active=params->has_video;
352 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
353 linphone_core_run_stun_tests(call->core,call);
355 discover_mtu(lc,linphone_address_get_domain (to));
356 if (params->referer){
357 sal_call_set_referer(call->op,params->referer->op);
358 call->referer=linphone_call_ref(params->referer);
363 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
364 LinphoneCall *call=ms_new0(LinphoneCall,1);
367 call->dir=LinphoneCallIncoming;
368 sal_op_set_user_pointer(op,call);
372 if (lc->sip_conf.ping_with_options){
373 /*the following sends an option request back to the caller so that
374 we get a chance to discover our nat'd address before answering.*/
375 call->ping_op=sal_op_new(lc->sal);
376 from_str=linphone_address_as_string_uri_only(from);
377 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
378 sal_op_set_user_pointer(call->ping_op,call);
379 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
383 linphone_address_clean(from);
384 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
385 linphone_call_init_common(call, from, to);
386 linphone_core_init_default_params(lc, &call->params);
387 call->params.has_video &= !!lc->video_policy.automatically_accept;
388 call->localdesc=create_local_media_description (lc,call);
389 call->camera_active=call->params.has_video;
390 switch (linphone_core_get_firewall_policy(call->core)) {
391 case LinphonePolicyUseIce:
392 call->ice_session = ice_session_new();
393 ice_session_set_role(call->ice_session, IR_Controlled);
394 linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
395 if (call->ice_session != NULL) {
396 linphone_call_init_media_streams(call);
397 linphone_call_start_media_streams_for_ice_gathering(call);
398 if (linphone_core_gather_ice_candidates(call->core,call)<0) {
399 /* Ice candidates gathering failed, proceed with the call anyway. */
400 linphone_call_delete_ice_session(call);
401 linphone_call_stop_media_streams(call);
405 case LinphonePolicyUseStun:
406 linphone_core_run_stun_tests(call->core,call);
407 /* No break to also destroy ice session in this case. */
411 discover_mtu(lc,linphone_address_get_domain(from));
415 /* this function is called internally to get rid of a call.
416 It performs the following tasks:
417 - remove the call from the internal list of calls
418 - update the call logs accordingly
421 static void linphone_call_set_terminated(LinphoneCall *call){
422 LinphoneCore *lc=call->core;
424 linphone_core_update_allocated_audio_bandwidth(lc);
426 call->owns_call_log=FALSE;
427 linphone_call_log_completed(call);
430 if (call == lc->current_call){
431 ms_message("Resetting the current call");
432 lc->current_call=NULL;
435 if (linphone_core_del_call(lc,call) != 0){
436 ms_error("Could not remove the call from the list !!!");
439 if (ms_list_size(lc->calls)==0)
440 linphone_core_notify_all_friends(lc,lc->presence_mode);
442 linphone_core_conference_check_uninit(lc);
443 if (call->ringing_beep){
444 linphone_core_stop_dtmf(lc);
445 call->ringing_beep=FALSE;
448 linphone_call_unref(call->referer);
453 void linphone_call_fix_call_parameters(LinphoneCall *call){
454 call->params.has_video=call->current_params.has_video;
455 call->params.media_encryption=call->current_params.media_encryption;
458 const char *linphone_call_state_to_string(LinphoneCallState cs){
460 case LinphoneCallIdle:
461 return "LinphoneCallIdle";
462 case LinphoneCallIncomingReceived:
463 return "LinphoneCallIncomingReceived";
464 case LinphoneCallOutgoingInit:
465 return "LinphoneCallOutgoingInit";
466 case LinphoneCallOutgoingProgress:
467 return "LinphoneCallOutgoingProgress";
468 case LinphoneCallOutgoingRinging:
469 return "LinphoneCallOutgoingRinging";
470 case LinphoneCallOutgoingEarlyMedia:
471 return "LinphoneCallOutgoingEarlyMedia";
472 case LinphoneCallConnected:
473 return "LinphoneCallConnected";
474 case LinphoneCallStreamsRunning:
475 return "LinphoneCallStreamsRunning";
476 case LinphoneCallPausing:
477 return "LinphoneCallPausing";
478 case LinphoneCallPaused:
479 return "LinphoneCallPaused";
480 case LinphoneCallResuming:
481 return "LinphoneCallResuming";
482 case LinphoneCallRefered:
483 return "LinphoneCallRefered";
484 case LinphoneCallError:
485 return "LinphoneCallError";
486 case LinphoneCallEnd:
487 return "LinphoneCallEnd";
488 case LinphoneCallPausedByRemote:
489 return "LinphoneCallPausedByRemote";
490 case LinphoneCallUpdatedByRemote:
491 return "LinphoneCallUpdatedByRemote";
492 case LinphoneCallIncomingEarlyMedia:
493 return "LinphoneCallIncomingEarlyMedia";
494 case LinphoneCallUpdated:
495 return "LinphoneCallUpdated";
496 case LinphoneCallReleased:
497 return "LinphoneCallReleased";
499 return "undefined state";
502 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
503 LinphoneCore *lc=call->core;
505 if (call->state!=cstate){
506 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
507 if (cstate!=LinphoneCallReleased){
508 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
509 linphone_call_state_to_string(cstate));
513 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
514 linphone_call_state_to_string(cstate));
515 if (cstate!=LinphoneCallRefered){
516 /*LinphoneCallRefered is rather an event, not a state.
517 Indeed it does not change the state of the call (still paused or running)*/
520 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
521 switch(call->reason){
522 case LinphoneReasonDeclined:
523 call->log->status=LinphoneCallDeclined;
528 >>>>>>> add device identifier api
529 case LinphoneReasonNotAnswered:
530 call->log->status=LinphoneCallMissed;
535 linphone_call_set_terminated (call);
537 if (cstate == LinphoneCallConnected) {
538 call->log->status=LinphoneCallSuccess;
539 call->media_start_time=time(NULL);
542 if (lc->vtable.call_state_changed)
543 lc->vtable.call_state_changed(lc,call,cstate,message);
544 if (cstate==LinphoneCallReleased){
545 if (call->op!=NULL) {
546 /* so that we cannot have anymore upcalls for SAL
547 concerning this call*/
548 sal_op_release(call->op);
551 linphone_call_unref(call);
556 static void linphone_call_destroy(LinphoneCall *obj)
559 sal_op_release(obj->op);
562 if (obj->resultdesc!=NULL) {
563 sal_media_description_unref(obj->resultdesc);
564 obj->resultdesc=NULL;
566 if (obj->localdesc!=NULL) {
567 sal_media_description_unref(obj->localdesc);
571 sal_op_release(obj->ping_op);
574 ms_free(obj->refer_to);
576 if (obj->owns_call_log)
577 linphone_call_log_destroy(obj->log);
578 if (obj->auth_token) {
579 ms_free(obj->auth_token);
581 if (obj->ice_session) {
582 ice_session_destroy(obj->ice_session);
589 * @addtogroup call_control
594 * Increments the call 's reference count.
595 * An application that wishes to retain a pointer to call object
596 * must use this function to unsure the pointer remains
597 * valid. Once the application no more needs this pointer,
598 * it must call linphone_call_unref().
600 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
606 * Decrements the call object reference count.
607 * See linphone_call_ref().
609 void linphone_call_unref(LinphoneCall *obj){
612 linphone_call_destroy(obj);
617 * Returns current parameters associated to the call.
619 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
620 return &call->current_params;
623 static bool_t is_video_active(const SalStreamDescription *sd){
624 return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
628 * Returns call parameters proposed by remote.
630 * This is useful when receiving an incoming call, to know whether the remote party
631 * supports video, encryption or whatever.
633 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
634 LinphoneCallParams *cp=&call->remote_params;
635 memset(cp,0,sizeof(*cp));
637 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
639 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
641 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
642 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
643 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
644 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
646 cp->has_video=is_video_active(secure_vsd);
647 if (secure_asd || asd==NULL)
648 cp->media_encryption=LinphoneMediaEncryptionSRTP;
650 cp->has_video=is_video_active(vsd);
659 * Returns the remote address associated to this call
662 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
663 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
667 * Returns the remote address associated to this call as a string.
669 * The result string must be freed by user using ms_free().
671 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
672 return linphone_address_as_string(linphone_call_get_remote_address(call));
676 * Retrieves the call's current state.
678 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
683 * Returns the reason for a call termination (either error or normal termination)
685 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
690 * Get the user_pointer in the LinphoneCall
692 * @ingroup call_control
694 * return user_pointer an opaque user pointer that can be retrieved at any time
696 void *linphone_call_get_user_pointer(LinphoneCall *call)
698 return call->user_pointer;
702 * Set the user_pointer in the LinphoneCall
704 * @ingroup call_control
706 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
708 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
710 call->user_pointer = user_pointer;
714 * Returns the call log associated to this call.
716 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
721 * Returns the refer-to uri (if the call was transfered).
723 const char *linphone_call_get_refer_to(const LinphoneCall *call){
724 return call->refer_to;
728 * Returns direction of the call (incoming or outgoing).
730 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
731 return call->log->dir;
735 * Returns the far end's user agent description string, if available.
737 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
739 return sal_op_get_remote_ua (call->op);
745 * Returns true if this calls has received a transfer that has not been
747 * Pending transfers are executed when this call is being paused or closed,
748 * locally or by remote endpoint.
749 * If the call is already paused while receiving the transfer request, the
750 * transfer immediately occurs.
752 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
753 return call->refer_pending;
757 * Returns call's duration in seconds.
759 int linphone_call_get_duration(const LinphoneCall *call){
760 if (call->media_start_time==0) return 0;
761 return time(NULL)-call->media_start_time;
765 * Returns the call object this call is replacing, if any.
766 * Call replacement can occur during call transfers.
767 * By default, the core automatically terminates the replaced call and accept the new one.
768 * This function allows the application to know whether a new incoming call is a one that replaces another one.
770 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
771 SalOp *op=sal_call_get_replaces(call->op);
773 return (LinphoneCall*)sal_op_get_user_pointer(op);
779 * Indicate whether camera input should be sent to remote end.
781 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
783 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
784 LinphoneCore *lc=call->core;
785 MSWebCam *nowebcam=get_nowebcam_device();
786 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
787 video_stream_change_camera(call->videostream,
788 enable ? lc->video_conf.device : nowebcam);
791 call->camera_active=enable;
796 * Take a photo of currently received video and write it into a jpeg file.
798 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
800 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
801 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
803 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
810 * Returns TRUE if camera pictures are sent to the remote party.
812 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
813 return call->camera_active;
817 * Enable video stream.
819 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
820 cp->has_video=enabled;
823 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
824 return cp->audio_codec;
827 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
828 return cp->video_codec;
832 * Returns whether video is enabled.
834 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
835 return cp->has_video;
838 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
839 return cp->media_encryption;
842 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
843 cp->media_encryption = e;
848 * Enable sending of real early media (during outgoing calls).
850 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
851 cp->real_early_media=enabled;
854 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
855 return cp->real_early_media;
859 * Returns true if the call is part of the locally managed conference.
861 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
862 return cp->in_conference;
866 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
867 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
869 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
870 cp->audio_bw=bandwidth;
875 * Request remote side to send us a Video Fast Update.
877 void linphone_call_send_vfu_request(LinphoneCall *call)
879 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
880 sal_call_send_vfu_request(call->op);
887 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
888 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
889 memcpy(ncp,cp,sizeof(LinphoneCallParams));
896 void linphone_call_params_destroy(LinphoneCallParams *p){
905 #ifdef TEST_EXT_RENDERER
906 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
907 ms_message("rendercb, local buffer=%p, remote buffer=%p",
908 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
913 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
914 LinphoneCall* call = (LinphoneCall*) user_pointer;
915 ms_warning("In linphonecall.c: video_stream_event_cb");
917 case MS_VIDEO_DECODER_DECODING_ERRORS:
918 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
919 linphone_call_send_vfu_request(call);
921 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
922 ms_message("First video frame decoded successfully");
923 if (call->nextVideoFrameDecoded._func != NULL)
924 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
927 ms_warning("Unhandled event %i", event_id);
933 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
934 call->nextVideoFrameDecoded._func = cb;
935 call->nextVideoFrameDecoded._user_data = user_data;
937 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
941 void linphone_call_init_audio_stream(LinphoneCall *call){
942 LinphoneCore *lc=call->core;
943 SalMediaDescription *md=call->localdesc;
944 AudioStream *audiostream;
945 int dscp=lp_config_get_int(lc->config,"rtp","audio_dscp",-1);
947 call->audiostream=audiostream=audio_stream_new(md->streams[0].rtp_port,md->streams[0].rtcp_port,linphone_core_ipv6_enabled(lc));
949 audio_stream_set_dscp(audiostream,dscp);
950 if (linphone_core_echo_limiter_enabled(lc)){
951 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
952 if (strcasecmp(type,"mic")==0)
953 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
954 else if (strcasecmp(type,"full")==0)
955 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
957 audio_stream_enable_gain_control(audiostream,TRUE);
958 if (linphone_core_echo_cancellation_enabled(lc)){
959 int len,delay,framesize;
960 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
961 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
962 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
963 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
964 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
965 if (statestr && audiostream->ec){
966 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
969 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
971 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
972 audio_stream_enable_noise_gate(audiostream,enabled);
975 audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
978 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
979 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
980 rtp_session_set_transports(audiostream->session,artp,artcp);
982 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
983 rtp_session_set_pktinfo(audiostream->session, TRUE);
984 rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
985 audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
986 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
989 call->audiostream_app_evq = ortp_ev_queue_new();
990 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
993 void linphone_call_init_video_stream(LinphoneCall *call){
995 LinphoneCore *lc=call->core;
996 SalMediaDescription *md=call->localdesc;
998 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].rtp_port>0){
999 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
1000 int dscp=lp_config_get_int(lc->config,"rtp","video_dscp",-1);
1002 call->videostream=video_stream_new(md->streams[1].rtp_port,md->streams[1].rtcp_port,linphone_core_ipv6_enabled(lc));
1004 video_stream_set_dscp(call->videostream,dscp);
1005 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
1006 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
1008 if( lc->video_conf.displaytype != NULL)
1009 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
1010 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
1012 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
1013 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
1014 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
1016 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, 1))){
1017 rtp_session_set_pktinfo(call->videostream->session, TRUE);
1018 rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
1019 call->videostream->ice_check_list = ice_session_check_list(call->ice_session, 1);
1020 ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
1022 call->videostream_app_evq = ortp_ev_queue_new();
1023 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
1024 #ifdef TEST_EXT_RENDERER
1025 video_stream_set_render_callback(call->videostream,rendercb,NULL);
1029 call->videostream=NULL;
1033 void linphone_call_init_media_streams(LinphoneCall *call){
1034 linphone_call_init_audio_stream(call);
1035 linphone_call_init_video_stream(call);
1039 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1041 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
1042 LinphoneCore* lc = (LinphoneCore*)user_data;
1043 if (dtmf<0 || dtmf>15){
1044 ms_warning("Bad dtmf value %i",dtmf);
1047 if (lc->vtable.dtmf_received != NULL)
1048 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1051 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1053 MSFilter *f=st->equalizer;
1054 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1055 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1056 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1062 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1063 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1064 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1073 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1074 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1077 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1078 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1079 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1082 audio_stream_set_mic_gain(st,mic_gain);
1084 audio_stream_set_mic_gain(st,0);
1086 recv_gain = lc->sound_conf.soft_play_lev;
1087 if (recv_gain != 0) {
1088 linphone_core_set_playback_gain_db (lc,recv_gain);
1092 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1093 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1094 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1095 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1096 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1097 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1100 if (speed==-1) speed=0.03;
1101 if (force==-1) force=25;
1102 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1103 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1105 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1107 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1108 if (transmit_thres!=-1)
1109 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1111 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1112 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1115 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1116 float floorgain = 1/mic_gain;
1117 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1118 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1119 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1120 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1122 parametrize_equalizer(lc,st);
1125 static void post_configure_audio_streams(LinphoneCall*call){
1126 AudioStream *st=call->audiostream;
1127 LinphoneCore *lc=call->core;
1128 _post_configure_audio_stream(st,lc,call->audio_muted);
1129 if (lc->vtable.dtmf_received!=NULL){
1130 /* replace by our default action*/
1131 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1132 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1136 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1139 RtpProfile *prof=rtp_profile_new("Call profile");
1142 LinphoneCore *lc=call->core;
1146 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1147 PayloadType *pt=(PayloadType*)elem->data;
1150 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1151 if (desc->type==SalAudio){
1152 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1153 up_ptime=linphone_core_get_upload_ptime(lc);
1155 *used_pt=payload_type_get_number(pt);
1158 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1159 else if (md->bandwidth>0) {
1160 /*case where b=AS is given globally, not per stream*/
1161 remote_bw=md->bandwidth;
1162 if (desc->type==SalVideo){
1163 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1167 if (desc->type==SalAudio){
1168 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1169 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1170 if (bw>0) pt->normal_bitrate=bw*1000;
1171 else if (desc->type==SalAudio){
1172 pt->normal_bitrate=-1;
1175 up_ptime=desc->ptime;
1179 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1180 payload_type_append_send_fmtp(pt,tmp);
1182 number=payload_type_get_number(pt);
1183 if (rtp_profile_get_payload(prof,number)!=NULL){
1184 ms_warning("A payload type with number %i already exists in profile !",number);
1186 rtp_profile_set_payload(prof,number,pt);
1192 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1193 int pause_time=3000;
1194 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1195 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1198 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1200 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1201 LinphoneCore *lc=call->core;
1202 LinphoneCall *current=linphone_core_get_current_call(lc);
1203 return !linphone_core_is_in_conference(lc) &&
1204 (current==NULL || current==call);
1206 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1208 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1209 if (crypto[i].tag == tag) {
1215 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1216 LinphoneCore *lc=call->core;
1218 /* look for savp stream first */
1219 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1220 SalProtoRtpSavp,SalAudio);
1221 /* no savp audio stream, use avp */
1223 stream=sal_media_description_find_stream(call->resultdesc,
1224 SalProtoRtpAvp,SalAudio);
1226 if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1227 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1228 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1229 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1230 const char *playfile=lc->play_file;
1231 const char *recfile=lc->rec_file;
1232 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1236 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1237 if (playcard==NULL) {
1238 ms_warning("No card defined for playback !");
1240 if (captcard==NULL) {
1241 ms_warning("No card defined for capture !");
1243 /*Replace soundcard filters by inactive file players or recorders
1244 when placed in recvonly or sendonly mode*/
1245 if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1248 }else if (stream->dir==SalStreamSendOnly){
1252 /*And we will eventually play "playfile" if set by the user*/
1255 if (send_ringbacktone){
1257 playfile=NULL;/* it is setup later*/
1259 /*if playfile are supplied don't use soundcards*/
1260 if (lc->use_files) {
1264 if (call->params.in_conference){
1265 /* first create the graph without soundcard resources*/
1266 captcard=playcard=NULL;
1268 if (!linphone_call_sound_resources_available(call)){
1269 ms_message("Sound resources are used by another call, not using soundcard.");
1270 captcard=playcard=NULL;
1272 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1273 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1274 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1275 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1276 audio_stream_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
1277 audio_stream_start_full(
1279 call->audio_profile,
1280 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1282 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1283 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1285 linphone_core_get_audio_jittcomp(lc),
1292 post_configure_audio_streams(call);
1293 if (muted && !send_ringbacktone){
1294 audio_stream_set_mic_gain(call->audiostream,0);
1296 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1298 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1300 if (send_ringbacktone){
1301 setup_ring_player(lc,call);
1303 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1305 /* valid local tags are > 0 */
1306 if (stream->proto == SalProtoRtpSavp) {
1307 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1308 SalProtoRtpSavp,SalAudio);
1309 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1311 if (crypto_idx >= 0) {
1312 audio_stream_enable_strp(
1314 stream->crypto[0].algo,
1315 local_st_desc->crypto[crypto_idx].master_key,
1316 stream->crypto[0].master_key);
1317 call->audiostream_encrypted=TRUE;
1319 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1320 call->audiostream_encrypted=FALSE;
1322 }else call->audiostream_encrypted=FALSE;
1323 if (call->params.in_conference){
1324 /*transform the graph to connect it to the conference filter */
1325 bool_t mute=stream->dir==SalStreamRecvOnly;
1326 linphone_call_add_to_conf(call, mute);
1328 call->current_params.in_conference=call->params.in_conference;
1329 }else ms_warning("No audio stream accepted ?");
1333 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1334 #ifdef VIDEO_ENABLED
1335 LinphoneCore *lc=call->core;
1337 /* look for savp stream first */
1338 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1339 SalProtoRtpSavp,SalVideo);
1340 /* no savp audio stream, use avp */
1342 vstream=sal_media_description_find_stream(call->resultdesc,
1343 SalProtoRtpAvp,SalVideo);
1345 /* shutdown preview */
1346 if (lc->previewstream!=NULL) {
1347 video_preview_stop(lc->previewstream);
1348 lc->previewstream=NULL;
1351 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1352 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1353 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1354 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1356 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1357 VideoStreamDir dir=VideoStreamSendRecv;
1358 MSWebCam *cam=lc->video_conf.device;
1359 bool_t is_inactive=FALSE;
1361 call->current_params.has_video=TRUE;
1363 video_stream_enable_adaptive_bitrate_control(call->videostream,
1364 linphone_core_adaptive_rate_control_enabled(lc));
1365 video_stream_enable_adaptive_jittcomp(call->videostream, linphone_core_video_adaptive_jittcomp_enabled(lc));
1366 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1367 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1368 if (lc->video_window_id!=0)
1369 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1370 if (lc->preview_window_id!=0)
1371 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1372 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1374 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1375 cam=get_nowebcam_device();
1376 dir=VideoStreamSendOnly;
1377 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1378 dir=VideoStreamRecvOnly;
1379 }else if (vstream->dir==SalStreamSendRecv){
1380 if (lc->video_conf.display && lc->video_conf.capture)
1381 dir=VideoStreamSendRecv;
1382 else if (lc->video_conf.display)
1383 dir=VideoStreamRecvOnly;
1385 dir=VideoStreamSendOnly;
1387 ms_warning("video stream is inactive.");
1388 /*either inactive or incompatible with local capabilities*/
1391 if (call->camera_active==FALSE || all_inputs_muted){
1392 cam=get_nowebcam_device();
1395 call->log->video_enabled = TRUE;
1396 video_stream_set_direction (call->videostream, dir);
1397 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1398 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1399 video_stream_start(call->videostream,
1400 call->video_profile, rtp_addr, vstream->rtp_port,
1401 rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1402 used_pt, linphone_core_get_video_jittcomp(lc), cam);
1403 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1406 if (vstream->proto == SalProtoRtpSavp) {
1407 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1408 SalProtoRtpSavp,SalVideo);
1410 video_stream_enable_strp(
1412 vstream->crypto[0].algo,
1413 local_st_desc->crypto[0].master_key,
1414 vstream->crypto[0].master_key
1416 call->videostream_encrypted=TRUE;
1418 call->videostream_encrypted=FALSE;
1420 }else ms_warning("No video stream accepted.");
1422 ms_warning("No valid video stream defined.");
1427 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1428 LinphoneCore *lc=call->core;
1430 call->current_params.audio_codec = NULL;
1431 call->current_params.video_codec = NULL;
1433 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1435 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1436 #ifdef VIDEO_ENABLED
1437 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1438 SalProtoRtpAvp,SalVideo);
1441 if ((call->audiostream == NULL) && (call->videostream == NULL)) {
1442 ms_fatal("start_media_stream() called without prior init !");
1445 cname=linphone_address_as_string_uri_only(me);
1447 #if defined(VIDEO_ENABLED)
1448 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1449 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1453 if (call->audiostream!=NULL) {
1454 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1456 call->current_params.has_video=FALSE;
1457 if (call->videostream!=NULL) {
1458 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1461 call->all_muted=all_inputs_muted;
1462 call->playing_ringbacktone=send_ringbacktone;
1463 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1465 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1466 OrtpZrtpParams params;
1467 /*will be set later when zrtp is activated*/
1468 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1470 params.zid_file=lc->zrtp_secrets_cache;
1471 audio_stream_enable_zrtp(call->audiostream,¶ms);
1472 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1473 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1474 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1477 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1478 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1479 linphone_call_fix_call_parameters(call);
1480 if ((call->ice_session != NULL) && (ice_session_state(call->ice_session) != IS_Completed)) {
1481 ice_session_start_connectivity_checks(call->ice_session);
1487 linphone_address_destroy(me);
1490 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1491 audio_stream_prepare_sound(call->audiostream, NULL, NULL);
1492 #ifdef VIDEO_ENABLED
1493 if (call->videostream) {
1494 video_stream_prepare_video(call->videostream);
1499 void linphone_call_delete_ice_session(LinphoneCall *call){
1500 if (call->ice_session != NULL) {
1501 ice_session_destroy(call->ice_session);
1502 call->ice_session = NULL;
1503 if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
1504 if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
1508 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1509 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1510 log->quality=audio_stream_get_average_quality_rating(st);
1513 void linphone_call_stop_media_streams(LinphoneCall *call){
1514 if (call->audiostream!=NULL) {
1515 call->audiostream->ice_check_list = NULL;
1516 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1517 ortp_ev_queue_flush(call->audiostream_app_evq);
1518 ortp_ev_queue_destroy(call->audiostream_app_evq);
1519 call->audiostream_app_evq=NULL;
1521 if (call->audiostream->ec){
1522 const char *state_str=NULL;
1523 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1525 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1526 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1529 linphone_call_log_fill_stats (call->log,call->audiostream);
1530 if (call->endpoint){
1531 linphone_call_remove_from_conf(call);
1533 audio_stream_stop(call->audiostream);
1534 call->audiostream=NULL;
1538 #ifdef VIDEO_ENABLED
1539 if (call->videostream!=NULL){
1540 call->videostream->ice_check_list = NULL;
1541 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1542 ortp_ev_queue_flush(call->videostream_app_evq);
1543 ortp_ev_queue_destroy(call->videostream_app_evq);
1544 call->videostream_app_evq=NULL;
1545 video_stream_stop(call->videostream);
1546 call->videostream=NULL;
1549 ms_event_queue_skip(call->core->msevq);
1551 if (call->audio_profile){
1552 rtp_profile_clear_all(call->audio_profile);
1553 rtp_profile_destroy(call->audio_profile);
1554 call->audio_profile=NULL;
1556 if (call->video_profile){
1557 rtp_profile_clear_all(call->video_profile);
1558 rtp_profile_destroy(call->video_profile);
1559 call->video_profile=NULL;
1565 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1566 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1567 bool_t bypass_mode = !enable;
1568 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1571 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1572 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1574 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1577 return linphone_core_echo_cancellation_enabled(call->core);
1581 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1582 if (call!=NULL && call->audiostream!=NULL ) {
1584 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1585 if (strcasecmp(type,"mic")==0)
1586 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1587 else if (strcasecmp(type,"full")==0)
1588 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1590 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1595 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1596 if (call!=NULL && call->audiostream!=NULL ){
1597 return call->audiostream->el_type !=ELInactive ;
1599 return linphone_core_echo_limiter_enabled(call->core);
1604 * @addtogroup call_misc
1609 * Returns the measured sound volume played locally (received from remote).
1610 * It is expressed in dbm0.
1612 float linphone_call_get_play_volume(LinphoneCall *call){
1613 AudioStream *st=call->audiostream;
1614 if (st && st->volrecv){
1616 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1620 return LINPHONE_VOLUME_DB_LOWEST;
1624 * Returns the measured sound volume recorded locally (sent to remote).
1625 * It is expressed in dbm0.
1627 float linphone_call_get_record_volume(LinphoneCall *call){
1628 AudioStream *st=call->audiostream;
1629 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1631 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1635 return LINPHONE_VOLUME_DB_LOWEST;
1639 * Obtain real-time quality rating of the call
1641 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1642 * during all the duration of the call. This function returns its value at the time of the function call.
1643 * It is expected that the rating is updated at least every 5 seconds or so.
1644 * The rating is a floating point number comprised between 0 and 5.
1646 * 4-5 = good quality <br>
1647 * 3-4 = average quality <br>
1648 * 2-3 = poor quality <br>
1649 * 1-2 = very poor quality <br>
1650 * 0-1 = can't be worse, mostly unusable <br>
1652 * @returns The function returns -1 if no quality measurement is available, for example if no
1653 * active audio stream exist. Otherwise it returns the quality rating.
1655 float linphone_call_get_current_quality(LinphoneCall *call){
1656 if (call->audiostream){
1657 return audio_stream_get_quality_rating(call->audiostream);
1663 * Returns call quality averaged over all the duration of the call.
1665 * See linphone_call_get_current_quality() for more details about quality measurement.
1667 float linphone_call_get_average_quality(LinphoneCall *call){
1668 if (call->audiostream){
1669 return audio_stream_get_average_quality_rating(call->audiostream);
1675 * Access last known statistics for audio stream, for a given call.
1677 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1678 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1682 * Access last known statistics for video stream, for a given call.
1684 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1685 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1693 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1694 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1695 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1696 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1697 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1698 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1701 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1705 from = linphone_call_get_remote_address_as_string(call);
1708 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1713 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1715 if (lc->vtable.display_warning!=NULL)
1716 lc->vtable.display_warning(lc,temp);
1717 linphone_core_terminate_call(lc,call);
1720 static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
1721 OrtpEventType evt=ortp_event_get_type(ev);
1722 OrtpEventData *evd=ortp_event_get_data(ev);
1724 if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1725 switch (ice_session_state(call->ice_session)) {
1727 if (ice_session_role(call->ice_session) == IR_Controlling) {
1728 ice_session_select_candidates(call->ice_session);
1729 linphone_core_update_call(call->core, call, &call->current_params);
1733 if (ice_session_has_completed_check_list(call->ice_session) == TRUE) {
1734 if (ice_session_role(call->ice_session) == IR_Controlling) {
1735 /* At least one ICE session has succeeded, so perform a call update. */
1736 ice_session_select_candidates(call->ice_session);
1737 linphone_core_update_call(call->core, call, &call->current_params);
1744 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1745 if (evd->info.ice_processing_successful==TRUE) {
1746 ice_session_compute_candidates_foundations(call->ice_session);
1747 ice_session_eliminate_redundant_candidates(call->ice_session);
1748 ice_session_choose_default_candidates(call->ice_session);
1750 linphone_call_delete_ice_session(call);
1752 switch (call->state) {
1753 case LinphoneCallStreamsRunning:
1754 linphone_core_start_update_call(call->core, call);
1756 case LinphoneCallUpdatedByRemote:
1757 linphone_core_start_accept_call_update(call->core, call);
1759 case LinphoneCallOutgoingInit:
1760 linphone_call_stop_media_streams(call);
1761 linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
1764 linphone_call_stop_media_streams(call);
1765 linphone_core_notify_incoming_call(call->core, call);
1768 } else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
1769 linphone_core_start_accept_call_update(call->core, call);
1770 } else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
1771 ice_session_restart(call->ice_session);
1772 ice_session_set_role(call->ice_session, IR_Controlling);
1773 linphone_core_update_call(call->core, call, &call->current_params);
1777 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1778 LinphoneCore* lc = call->core;
1779 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1780 bool_t disconnected=FALSE;
1782 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1783 RtpSession *as=NULL,*vs=NULL;
1784 float audio_load=0, video_load=0;
1785 if (call->audiostream!=NULL){
1786 as=call->audiostream->session;
1787 if (call->audiostream->ticker)
1788 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1790 if (call->videostream!=NULL){
1791 if (call->videostream->ticker)
1792 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1793 vs=call->videostream->session;
1795 display_bandwidth(as,vs);
1796 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1798 #ifdef VIDEO_ENABLED
1799 if (call->videostream!=NULL) {
1802 /* Ensure there is no dangling ICE check list. */
1803 if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
1805 // Beware that the application queue should not depend on treatments fron the
1806 // mediastreamer queue.
1807 video_stream_iterate(call->videostream);
1809 while (call->videostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq)))){
1810 OrtpEventType evt=ortp_event_get_type(ev);
1811 OrtpEventData *evd=ortp_event_get_data(ev);
1812 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1813 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1814 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1815 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1816 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1817 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1818 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1820 if (lc->vtable.call_stats_updated)
1821 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1822 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1823 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1824 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1825 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1826 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1828 if (lc->vtable.call_stats_updated)
1829 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1830 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1831 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1832 handle_ice_events(call, ev);
1834 ortp_event_destroy(ev);
1838 if (call->audiostream!=NULL) {
1841 /* Ensure there is no dangling ICE check list. */
1842 if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
1844 // Beware that the application queue should not depend on treatments fron the
1845 // mediastreamer queue.
1846 audio_stream_iterate(call->audiostream);
1848 while (call->audiostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq)))){
1849 OrtpEventType evt=ortp_event_get_type(ev);
1850 OrtpEventData *evd=ortp_event_get_data(ev);
1851 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1852 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1853 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1854 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1855 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1856 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1857 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1858 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1859 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1861 if (lc->vtable.call_stats_updated)
1862 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1863 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1864 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1865 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1866 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1867 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1869 if (lc->vtable.call_stats_updated)
1870 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1871 } else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
1872 || (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
1873 handle_ice_events(call, ev);
1875 ortp_event_destroy(ev);
1878 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1879 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1881 linphone_core_disconnected(call->core,call);
1884 void linphone_call_log_completed(LinphoneCall *call){
1885 LinphoneCore *lc=call->core;
1887 call->log->duration=time(NULL)-call->start_time;
1889 if (call->log->status==LinphoneCallMissed){
1892 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1893 "You have missed %i calls.", lc->missed_calls),
1895 if (lc->vtable.display_status!=NULL)
1896 lc->vtable.display_status(lc,info);
1899 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1900 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1901 MSList *elem,*prevelem=NULL;
1902 /*find the last element*/
1903 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1907 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1908 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1910 if (lc->vtable.call_log_updated!=NULL){
1911 lc->vtable.call_log_updated(lc,call->log);
1913 call_logs_write_to_config_file(lc);
1916 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1917 return call->transfer_state;
1920 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1921 if (state != call->transfer_state) {
1922 LinphoneCore* lc = call->core;
1923 call->transfer_state = state;
1924 if (lc->vtable.transfer_state_changed)
1925 lc->vtable.transfer_state_changed(lc, call, state);
1929 bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
1930 return call->params.in_conference;