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();
204 IceSession *ice_session=sal_op_get_ice_session(call->op);
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].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
215 strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
216 md->streams[0].rtp_port=call->audio_port;
217 md->streams[0].rtcp_port=call->audio_port+1;
218 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
219 SalProtoRtpSavp : SalProtoRtpAvp;
220 md->streams[0].type=SalAudio;
221 md->streams[0].ptime=lc->net_conf.down_ptime;
222 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
223 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
224 l=ms_list_append(l,pt);
225 md->streams[0].payloads=l;
229 if (call->params.has_video){
231 md->streams[1].rtp_port=call->video_port;
232 md->streams[1].rtcp_port=call->video_port+1;
233 md->streams[1].proto=md->streams[0].proto;
234 md->streams[1].type=SalVideo;
235 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
236 md->streams[1].payloads=l;
239 for(i=0; i<md->nstreams; i++) {
240 if (md->streams[i].proto == SalProtoRtpSavp) {
241 md->streams[i].crypto[0].tag = 1;
242 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
243 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
244 md->streams[i].crypto[0].algo = 0;
245 md->streams[i].crypto[1].tag = 2;
246 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
247 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
248 md->streams[i].crypto[1].algo = 0;
249 md->streams[i].crypto[2].algo = 0;
251 if ((call->dir == LinphoneCallOutgoing) && (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (ice_session != NULL)){
252 ice_session_add_check_list(ice_session, ice_check_list_new());
256 linphone_address_destroy(addr);
260 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
261 SalMediaDescription *md=call->localdesc;
263 call->localdesc = create_local_media_description(lc,call);
265 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
266 sal_media_description_unref(md);
270 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
271 unsigned int id=rand() & 0xfff;
272 return _create_local_media_description(lc,call,id,id);
275 static int find_port_offset(LinphoneCore *lc){
279 bool_t already_used=FALSE;
280 for(offset=0;offset<100;offset+=2){
281 audio_port=linphone_core_get_audio_port (lc)+offset;
283 for(elem=lc->calls;elem!=NULL;elem=elem->next){
284 LinphoneCall *call=(LinphoneCall*)elem->data;
285 if (call->audio_port==audio_port) {
290 if (!already_used) break;
293 ms_error("Could not find any free port !");
299 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
301 call->magic=linphone_call_magic;
303 call->state=LinphoneCallIdle;
304 call->transfer_state = LinphoneCallIdle;
305 call->start_time=time(NULL);
306 call->media_start_time=0;
307 call->log=linphone_call_log_new(call, from, to);
308 call->owns_call_log=TRUE;
309 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
310 port_offset=find_port_offset (call->core);
311 if (port_offset==-1) return;
312 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
313 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
314 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
315 linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
318 void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
320 stats->received_rtcp = NULL;
321 stats->sent_rtcp = NULL;
324 static void discover_mtu(LinphoneCore *lc, const char *remote){
326 if (lc->net_conf.mtu==0 ){
327 /*attempt to discover mtu*/
328 mtu=ms_discover_mtu(remote);
331 ms_message("Discovered mtu is %i, RTP payload max size is %i",
332 mtu, ms_get_payload_max_size());
337 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
339 LinphoneCall *call=ms_new0(LinphoneCall,1);
340 call->dir=LinphoneCallOutgoing;
341 call->op=sal_op_new(lc->sal);
342 sal_op_set_user_pointer(call->op,call);
344 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
345 linphone_call_init_common(call,from,to);
346 call->params=*params;
347 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
348 sal_op_set_ice_session(call->op, ice_session_new());
349 ice_session_set_role(sal_op_get_ice_session(call->op), IR_Controlling);
351 call->localdesc=create_local_media_description (lc,call);
352 call->camera_active=params->has_video;
353 if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
354 linphone_core_run_stun_tests(call->core,call);
356 discover_mtu(lc,linphone_address_get_domain (to));
357 if (params->referer){
358 sal_call_set_referer(call->op,params->referer->op);
359 call->referer=linphone_call_ref(params->referer);
364 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
365 LinphoneCall *call=ms_new0(LinphoneCall,1);
368 call->dir=LinphoneCallIncoming;
369 sal_op_set_user_pointer(op,call);
373 if (lc->sip_conf.ping_with_options){
374 /*the following sends an option request back to the caller so that
375 we get a chance to discover our nat'd address before answering.*/
376 call->ping_op=sal_op_new(lc->sal);
377 from_str=linphone_address_as_string_uri_only(from);
378 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
379 sal_op_set_user_pointer(call->ping_op,call);
380 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
384 linphone_address_clean(from);
385 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
386 linphone_call_init_common(call, from, to);
387 linphone_core_init_default_params(lc, &call->params);
388 call->params.has_video &= !!lc->video_policy.automatically_accept;
389 call->localdesc=create_local_media_description (lc,call);
390 call->camera_active=call->params.has_video;
391 switch (linphone_core_get_firewall_policy(call->core)) {
392 case LinphonePolicyUseIce:
393 linphone_core_gather_ice_candidates(call->core, call);
395 case LinphonePolicyUseStun:
396 linphone_core_run_stun_tests(call->core,call);
397 /* No break to also destroy ice session in this case. */
399 if (sal_op_get_ice_session(call->op) != NULL) {
400 ice_session_destroy(sal_op_get_ice_session(call->op));
401 sal_op_set_ice_session(call->op, NULL);
405 discover_mtu(lc,linphone_address_get_domain(from));
409 /* this function is called internally to get rid of a call.
410 It performs the following tasks:
411 - remove the call from the internal list of calls
412 - update the call logs accordingly
415 static void linphone_call_set_terminated(LinphoneCall *call){
416 LinphoneCore *lc=call->core;
418 linphone_core_update_allocated_audio_bandwidth(lc);
420 call->owns_call_log=FALSE;
421 linphone_call_log_completed(call);
424 if (call == lc->current_call){
425 ms_message("Resetting the current call");
426 lc->current_call=NULL;
429 if (linphone_core_del_call(lc,call) != 0){
430 ms_error("Could not remove the call from the list !!!");
433 if (ms_list_size(lc->calls)==0)
434 linphone_core_notify_all_friends(lc,lc->presence_mode);
436 linphone_core_conference_check_uninit(lc);
437 if (call->ringing_beep){
438 linphone_core_stop_dtmf(lc);
439 call->ringing_beep=FALSE;
442 linphone_call_unref(call->referer);
447 void linphone_call_fix_call_parameters(LinphoneCall *call){
448 call->params.has_video=call->current_params.has_video;
449 call->params.media_encryption=call->current_params.media_encryption;
452 const char *linphone_call_state_to_string(LinphoneCallState cs){
454 case LinphoneCallIdle:
455 return "LinphoneCallIdle";
456 case LinphoneCallIncomingReceived:
457 return "LinphoneCallIncomingReceived";
458 case LinphoneCallOutgoingInit:
459 return "LinphoneCallOutgoingInit";
460 case LinphoneCallOutgoingProgress:
461 return "LinphoneCallOutgoingProgress";
462 case LinphoneCallOutgoingRinging:
463 return "LinphoneCallOutgoingRinging";
464 case LinphoneCallOutgoingEarlyMedia:
465 return "LinphoneCallOutgoingEarlyMedia";
466 case LinphoneCallConnected:
467 return "LinphoneCallConnected";
468 case LinphoneCallStreamsRunning:
469 return "LinphoneCallStreamsRunning";
470 case LinphoneCallPausing:
471 return "LinphoneCallPausing";
472 case LinphoneCallPaused:
473 return "LinphoneCallPaused";
474 case LinphoneCallResuming:
475 return "LinphoneCallResuming";
476 case LinphoneCallRefered:
477 return "LinphoneCallRefered";
478 case LinphoneCallError:
479 return "LinphoneCallError";
480 case LinphoneCallEnd:
481 return "LinphoneCallEnd";
482 case LinphoneCallPausedByRemote:
483 return "LinphoneCallPausedByRemote";
484 case LinphoneCallUpdatedByRemote:
485 return "LinphoneCallUpdatedByRemote";
486 case LinphoneCallIncomingEarlyMedia:
487 return "LinphoneCallIncomingEarlyMedia";
488 case LinphoneCallUpdated:
489 return "LinphoneCallUpdated";
490 case LinphoneCallReleased:
491 return "LinphoneCallReleased";
493 return "undefined state";
496 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
497 LinphoneCore *lc=call->core;
499 if (call->state!=cstate){
500 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
501 if (cstate!=LinphoneCallReleased){
502 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
503 linphone_call_state_to_string(cstate));
507 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
508 linphone_call_state_to_string(cstate));
509 if (cstate!=LinphoneCallRefered){
510 /*LinphoneCallRefered is rather an event, not a state.
511 Indeed it does not change the state of the call (still paused or running)*/
514 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
515 switch(call->reason){
516 case LinphoneReasonDeclined:
517 call->log->status=LinphoneCallDeclined;
518 case LinphoneReasonNotAnswered:
519 call->log->status=LinphoneCallMissed;
524 linphone_call_set_terminated (call);
526 if (cstate == LinphoneCallConnected) {
527 call->log->status=LinphoneCallSuccess;
528 call->media_start_time=time(NULL);
531 if (lc->vtable.call_state_changed)
532 lc->vtable.call_state_changed(lc,call,cstate,message);
533 if (cstate==LinphoneCallReleased){
534 if (call->op!=NULL) {
535 /* so that we cannot have anymore upcalls for SAL
536 concerning this call*/
537 sal_op_release(call->op);
540 linphone_call_unref(call);
545 static void linphone_call_destroy(LinphoneCall *obj)
548 sal_op_release(obj->op);
551 if (obj->resultdesc!=NULL) {
552 sal_media_description_unref(obj->resultdesc);
553 obj->resultdesc=NULL;
555 if (obj->localdesc!=NULL) {
556 sal_media_description_unref(obj->localdesc);
560 sal_op_release(obj->ping_op);
563 ms_free(obj->refer_to);
565 if (obj->owns_call_log)
566 linphone_call_log_destroy(obj->log);
567 if (obj->auth_token) {
568 ms_free(obj->auth_token);
575 * @addtogroup call_control
580 * Increments the call 's reference count.
581 * An application that wishes to retain a pointer to call object
582 * must use this function to unsure the pointer remains
583 * valid. Once the application no more needs this pointer,
584 * it must call linphone_call_unref().
586 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
592 * Decrements the call object reference count.
593 * See linphone_call_ref().
595 void linphone_call_unref(LinphoneCall *obj){
598 linphone_call_destroy(obj);
603 * Returns current parameters associated to the call.
605 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
606 return &call->current_params;
609 static bool_t is_video_active(const SalStreamDescription *sd){
610 return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
614 * Returns call parameters proposed by remote.
616 * This is useful when receiving an incoming call, to know whether the remote party
617 * supports video, encryption or whatever.
619 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
620 LinphoneCallParams *cp=&call->remote_params;
621 memset(cp,0,sizeof(*cp));
623 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
625 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
627 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
628 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
629 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
630 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
632 cp->has_video=is_video_active(secure_vsd);
633 if (secure_asd || asd==NULL)
634 cp->media_encryption=LinphoneMediaEncryptionSRTP;
636 cp->has_video=is_video_active(vsd);
645 * Returns the remote address associated to this call
648 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
649 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
653 * Returns the remote address associated to this call as a string.
655 * The result string must be freed by user using ms_free().
657 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
658 return linphone_address_as_string(linphone_call_get_remote_address(call));
662 * Retrieves the call's current state.
664 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
669 * Returns the reason for a call termination (either error or normal termination)
671 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
676 * Get the user_pointer in the LinphoneCall
678 * @ingroup call_control
680 * return user_pointer an opaque user pointer that can be retrieved at any time
682 void *linphone_call_get_user_pointer(LinphoneCall *call)
684 return call->user_pointer;
688 * Set the user_pointer in the LinphoneCall
690 * @ingroup call_control
692 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
694 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
696 call->user_pointer = user_pointer;
700 * Returns the call log associated to this call.
702 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
707 * Returns the refer-to uri (if the call was transfered).
709 const char *linphone_call_get_refer_to(const LinphoneCall *call){
710 return call->refer_to;
714 * Returns direction of the call (incoming or outgoing).
716 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
717 return call->log->dir;
721 * Returns the far end's user agent description string, if available.
723 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
725 return sal_op_get_remote_ua (call->op);
731 * Returns true if this calls has received a transfer that has not been
733 * Pending transfers are executed when this call is being paused or closed,
734 * locally or by remote endpoint.
735 * If the call is already paused while receiving the transfer request, the
736 * transfer immediately occurs.
738 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
739 return call->refer_pending;
743 * Returns call's duration in seconds.
745 int linphone_call_get_duration(const LinphoneCall *call){
746 if (call->media_start_time==0) return 0;
747 return time(NULL)-call->media_start_time;
751 * Returns the call object this call is replacing, if any.
752 * Call replacement can occur during call transfers.
753 * By default, the core automatically terminates the replaced call and accept the new one.
754 * This function allows the application to know whether a new incoming call is a one that replaces another one.
756 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
757 SalOp *op=sal_call_get_replaces(call->op);
759 return (LinphoneCall*)sal_op_get_user_pointer(op);
765 * Indicate whether camera input should be sent to remote end.
767 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
769 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
770 LinphoneCore *lc=call->core;
771 MSWebCam *nowebcam=get_nowebcam_device();
772 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
773 video_stream_change_camera(call->videostream,
774 enable ? lc->video_conf.device : nowebcam);
777 call->camera_active=enable;
782 * Take a photo of currently received video and write it into a jpeg file.
784 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
786 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
787 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
789 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
796 * Returns TRUE if camera pictures are sent to the remote party.
798 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
799 return call->camera_active;
803 * Enable video stream.
805 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
806 cp->has_video=enabled;
809 const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
810 return cp->audio_codec;
813 const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
814 return cp->video_codec;
818 * Returns whether video is enabled.
820 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
821 return cp->has_video;
824 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
825 return cp->media_encryption;
828 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
829 cp->media_encryption = e;
834 * Enable sending of real early media (during outgoing calls).
836 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
837 cp->real_early_media=enabled;
840 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
841 return cp->real_early_media;
845 * Returns true if the call is part of the locally managed conference.
847 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
848 return cp->in_conference;
852 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
853 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
855 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
856 cp->audio_bw=bandwidth;
861 * Request remote side to send us a Video Fast Update.
863 void linphone_call_send_vfu_request(LinphoneCall *call)
865 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
866 sal_call_send_vfu_request(call->op);
873 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
874 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
875 memcpy(ncp,cp,sizeof(LinphoneCallParams));
882 void linphone_call_params_destroy(LinphoneCallParams *p){
891 #ifdef TEST_EXT_RENDERER
892 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
893 ms_message("rendercb, local buffer=%p, remote buffer=%p",
894 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
899 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
900 LinphoneCall* call = (LinphoneCall*) user_pointer;
901 ms_warning("In linphonecall.c: video_stream_event_cb");
903 case MS_VIDEO_DECODER_DECODING_ERRORS:
904 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
905 linphone_call_send_vfu_request(call);
907 case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
908 ms_message("First video frame decoded successfully");
909 if (call->nextVideoFrameDecoded._func != NULL)
910 call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
913 ms_warning("Unhandled event %i", event_id);
919 void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
920 call->nextVideoFrameDecoded._func = cb;
921 call->nextVideoFrameDecoded._user_data = user_data;
923 ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
927 void linphone_call_init_media_streams(LinphoneCall *call){
928 LinphoneCore *lc=call->core;
929 SalMediaDescription *md=call->localdesc;
930 AudioStream *audiostream;
931 IceSession *ice_session = sal_op_get_ice_session(call->op);
933 call->audiostream=audiostream=audio_stream_new(md->streams[0].rtp_port,md->streams[0].rtcp_port,linphone_core_ipv6_enabled(lc));
934 if (linphone_core_echo_limiter_enabled(lc)){
935 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
936 if (strcasecmp(type,"mic")==0)
937 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
938 else if (strcasecmp(type,"full")==0)
939 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
941 audio_stream_enable_gain_control(audiostream,TRUE);
942 if (linphone_core_echo_cancellation_enabled(lc)){
943 int len,delay,framesize;
944 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
945 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
946 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
947 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
948 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
949 if (statestr && audiostream->ec){
950 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
953 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
955 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
956 audio_stream_enable_noise_gate(audiostream,enabled);
960 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
961 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
962 rtp_session_set_transports(audiostream->session,artp,artcp);
964 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (ice_session != NULL)){
965 rtp_session_set_pktinfo(audiostream->session, TRUE);
966 audiostream->ice_check_list = ice_session_check_list(ice_session, 0);
967 ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
970 call->audiostream_app_evq = ortp_ev_queue_new();
971 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
975 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].rtp_port>0){
976 int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
977 call->videostream=video_stream_new(md->streams[1].rtp_port,md->streams[1].rtcp_port,linphone_core_ipv6_enabled(lc));
978 video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
979 if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
981 if( lc->video_conf.displaytype != NULL)
982 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
983 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
985 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
986 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
987 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
989 if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (ice_session != NULL)){
990 rtp_session_set_pktinfo(call->videostream->session, TRUE);
991 call->videostream->ice_check_list = ice_session_check_list(ice_session, 1);
992 ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
994 call->videostream_app_evq = ortp_ev_queue_new();
995 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
996 #ifdef TEST_EXT_RENDERER
997 video_stream_set_render_callback(call->videostream,rendercb,NULL);
1001 call->videostream=NULL;
1006 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
1008 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
1009 LinphoneCore* lc = (LinphoneCore*)user_data;
1010 if (dtmf<0 || dtmf>15){
1011 ms_warning("Bad dtmf value %i",dtmf);
1014 if (lc->vtable.dtmf_received != NULL)
1015 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
1018 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
1020 MSFilter *f=st->equalizer;
1021 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
1022 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
1023 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
1029 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
1030 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
1031 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
1040 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
1041 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
1044 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
1045 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
1046 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
1049 audio_stream_set_mic_gain(st,mic_gain);
1051 audio_stream_set_mic_gain(st,0);
1053 recv_gain = lc->sound_conf.soft_play_lev;
1054 if (recv_gain != 0) {
1055 linphone_core_set_playback_gain_db (lc,recv_gain);
1059 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
1060 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
1061 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
1062 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
1063 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
1064 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
1067 if (speed==-1) speed=0.03;
1068 if (force==-1) force=25;
1069 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
1070 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
1072 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
1074 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
1075 if (transmit_thres!=-1)
1076 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
1078 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1079 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
1082 /* parameters for a limited noise-gate effect, using echo limiter threshold */
1083 float floorgain = 1/mic_gain;
1084 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
1085 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1086 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1087 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1089 parametrize_equalizer(lc,st);
1092 static void post_configure_audio_streams(LinphoneCall*call){
1093 AudioStream *st=call->audiostream;
1094 LinphoneCore *lc=call->core;
1095 _post_configure_audio_stream(st,lc,call->audio_muted);
1096 if (lc->vtable.dtmf_received!=NULL){
1097 /* replace by our default action*/
1098 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1099 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1103 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1106 RtpProfile *prof=rtp_profile_new("Call profile");
1109 LinphoneCore *lc=call->core;
1113 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1114 PayloadType *pt=(PayloadType*)elem->data;
1117 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1118 if (desc->type==SalAudio){
1119 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1120 up_ptime=linphone_core_get_upload_ptime(lc);
1122 *used_pt=payload_type_get_number(pt);
1125 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1126 else if (md->bandwidth>0) {
1127 /*case where b=AS is given globally, not per stream*/
1128 remote_bw=md->bandwidth;
1129 if (desc->type==SalVideo){
1130 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1134 if (desc->type==SalAudio){
1135 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1136 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1137 if (bw>0) pt->normal_bitrate=bw*1000;
1138 else if (desc->type==SalAudio){
1139 pt->normal_bitrate=-1;
1142 up_ptime=desc->ptime;
1146 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1147 payload_type_append_send_fmtp(pt,tmp);
1149 number=payload_type_get_number(pt);
1150 if (rtp_profile_get_payload(prof,number)!=NULL){
1151 ms_warning("A payload type with number %i already exists in profile !",number);
1153 rtp_profile_set_payload(prof,number,pt);
1159 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1160 int pause_time=3000;
1161 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1162 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1165 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1167 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1168 LinphoneCore *lc=call->core;
1169 LinphoneCall *current=linphone_core_get_current_call(lc);
1170 return !linphone_core_is_in_conference(lc) &&
1171 (current==NULL || current==call);
1173 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1175 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1176 if (crypto[i].tag == tag) {
1182 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1183 LinphoneCore *lc=call->core;
1184 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1186 /* look for savp stream first */
1187 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1188 SalProtoRtpSavp,SalAudio);
1189 /* no savp audio stream, use avp */
1191 stream=sal_media_description_find_stream(call->resultdesc,
1192 SalProtoRtpAvp,SalAudio);
1194 if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
1195 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1196 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1197 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1198 const char *playfile=lc->play_file;
1199 const char *recfile=lc->rec_file;
1200 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1204 call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
1205 if (playcard==NULL) {
1206 ms_warning("No card defined for playback !");
1208 if (captcard==NULL) {
1209 ms_warning("No card defined for capture !");
1211 /*Replace soundcard filters by inactive file players or recorders
1212 when placed in recvonly or sendonly mode*/
1213 if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
1216 }else if (stream->dir==SalStreamSendOnly){
1220 /*And we will eventually play "playfile" if set by the user*/
1223 if (send_ringbacktone){
1225 playfile=NULL;/* it is setup later*/
1227 /*if playfile are supplied don't use soundcards*/
1228 if (lc->use_files) {
1232 if (call->params.in_conference){
1233 /* first create the graph without soundcard resources*/
1234 captcard=playcard=NULL;
1236 if (!linphone_call_sound_resources_available(call)){
1237 ms_message("Sound resources are used by another call, not using soundcard.");
1238 captcard=playcard=NULL;
1240 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1241 if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
1242 if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
1243 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1244 audio_stream_start_full(
1246 call->audio_profile,
1247 stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
1249 stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
1250 linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
1259 post_configure_audio_streams(call);
1260 if (muted && !send_ringbacktone){
1261 audio_stream_set_mic_gain(call->audiostream,0);
1263 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1265 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1267 if (send_ringbacktone){
1268 setup_ring_player(lc,call);
1270 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1272 /* valid local tags are > 0 */
1273 if (stream->proto == SalProtoRtpSavp) {
1274 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1275 SalProtoRtpSavp,SalAudio);
1276 int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
1278 if (crypto_idx >= 0) {
1279 audio_stream_enable_strp(
1281 stream->crypto[0].algo,
1282 local_st_desc->crypto[crypto_idx].master_key,
1283 stream->crypto[0].master_key);
1284 call->audiostream_encrypted=TRUE;
1286 ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
1287 call->audiostream_encrypted=FALSE;
1289 }else call->audiostream_encrypted=FALSE;
1290 if (call->params.in_conference){
1291 /*transform the graph to connect it to the conference filter */
1292 bool_t mute=stream->dir==SalStreamRecvOnly;
1293 linphone_call_add_to_conf(call, mute);
1295 call->current_params.in_conference=call->params.in_conference;
1296 }else ms_warning("No audio stream accepted ?");
1300 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1301 #ifdef VIDEO_ENABLED
1302 LinphoneCore *lc=call->core;
1304 /* look for savp stream first */
1305 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1306 SalProtoRtpSavp,SalVideo);
1307 /* no savp audio stream, use avp */
1309 vstream=sal_media_description_find_stream(call->resultdesc,
1310 SalProtoRtpAvp,SalVideo);
1312 /* shutdown preview */
1313 if (lc->previewstream!=NULL) {
1314 video_preview_stop(lc->previewstream);
1315 lc->previewstream=NULL;
1318 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
1319 const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
1320 const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
1321 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1323 call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
1324 VideoStreamDir dir=VideoStreamSendRecv;
1325 MSWebCam *cam=lc->video_conf.device;
1326 bool_t is_inactive=FALSE;
1328 call->current_params.has_video=TRUE;
1330 video_stream_enable_adaptive_bitrate_control(call->videostream,
1331 linphone_core_adaptive_rate_control_enabled(lc));
1332 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1333 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1334 if (lc->video_window_id!=0)
1335 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1336 if (lc->preview_window_id!=0)
1337 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1338 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1340 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1341 cam=get_nowebcam_device();
1342 dir=VideoStreamSendOnly;
1343 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1344 dir=VideoStreamRecvOnly;
1345 }else if (vstream->dir==SalStreamSendRecv){
1346 if (lc->video_conf.display && lc->video_conf.capture)
1347 dir=VideoStreamSendRecv;
1348 else if (lc->video_conf.display)
1349 dir=VideoStreamRecvOnly;
1351 dir=VideoStreamSendOnly;
1353 ms_warning("video stream is inactive.");
1354 /*either inactive or incompatible with local capabilities*/
1357 if (call->camera_active==FALSE || all_inputs_muted){
1358 cam=get_nowebcam_device();
1361 call->log->video_enabled = TRUE;
1362 video_stream_set_direction (call->videostream, dir);
1363 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1364 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1365 video_stream_start(call->videostream,
1366 call->video_profile, rtp_addr, vstream->rtp_port,
1367 rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
1368 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1369 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1372 if (vstream->proto == SalProtoRtpSavp) {
1373 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1374 SalProtoRtpSavp,SalVideo);
1376 video_stream_enable_strp(
1378 vstream->crypto[0].algo,
1379 local_st_desc->crypto[0].master_key,
1380 vstream->crypto[0].master_key
1382 call->videostream_encrypted=TRUE;
1384 call->videostream_encrypted=FALSE;
1386 }else ms_warning("No video stream accepted.");
1388 ms_warning("No valid video stream defined.");
1393 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1394 LinphoneCore *lc=call->core;
1396 call->current_params.audio_codec = NULL;
1397 call->current_params.video_codec = NULL;
1399 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1401 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1402 #ifdef VIDEO_ENABLED
1403 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1404 SalProtoRtpAvp,SalVideo);
1407 if(call->audiostream == NULL)
1409 ms_fatal("start_media_stream() called without prior init !");
1412 cname=linphone_address_as_string_uri_only(me);
1414 #if defined(VIDEO_ENABLED)
1415 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1416 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1420 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1421 call->current_params.has_video=FALSE;
1422 if (call->videostream!=NULL) {
1423 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1426 call->all_muted=all_inputs_muted;
1427 call->playing_ringbacktone=send_ringbacktone;
1428 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1430 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1431 OrtpZrtpParams params;
1432 /*will be set later when zrtp is activated*/
1433 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1435 params.zid_file=lc->zrtp_secrets_cache;
1436 audio_stream_enable_zrtp(call->audiostream,¶ms);
1437 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1438 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1439 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1442 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1443 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1444 linphone_call_fix_call_parameters(call);
1445 if ((sal_op_get_ice_session(call->op) != NULL) && (ice_session_state(sal_op_get_ice_session(call->op)) != IS_Completed)) {
1446 ice_session_start_connectivity_checks(sal_op_get_ice_session(call->op));
1452 linphone_address_destroy(me);
1455 void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
1456 audio_stream_start_ice_gathering(call->audiostream);
1457 if (call->videostream) {
1458 video_stream_start_ice_gathering(call->videostream);
1462 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1463 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1464 log->quality=audio_stream_get_average_quality_rating(st);
1467 void linphone_call_stop_media_streams(LinphoneCall *call){
1468 if (call->audiostream!=NULL) {
1469 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1470 ortp_ev_queue_flush(call->audiostream_app_evq);
1471 ortp_ev_queue_destroy(call->audiostream_app_evq);
1473 if (call->audiostream->ec){
1474 const char *state_str=NULL;
1475 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1477 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1478 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1481 linphone_call_log_fill_stats (call->log,call->audiostream);
1482 if (call->endpoint){
1483 linphone_call_remove_from_conf(call);
1485 audio_stream_stop(call->audiostream);
1486 call->audiostream=NULL;
1490 #ifdef VIDEO_ENABLED
1491 if (call->videostream!=NULL){
1492 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1493 ortp_ev_queue_flush(call->videostream_app_evq);
1494 ortp_ev_queue_destroy(call->videostream_app_evq);
1495 video_stream_stop(call->videostream);
1496 call->videostream=NULL;
1499 ms_event_queue_skip(call->core->msevq);
1501 if (call->audio_profile){
1502 rtp_profile_clear_all(call->audio_profile);
1503 rtp_profile_destroy(call->audio_profile);
1504 call->audio_profile=NULL;
1506 if (call->video_profile){
1507 rtp_profile_clear_all(call->video_profile);
1508 rtp_profile_destroy(call->video_profile);
1509 call->video_profile=NULL;
1515 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1516 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1517 bool_t bypass_mode = !enable;
1518 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1521 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1522 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1524 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1527 return linphone_core_echo_cancellation_enabled(call->core);
1531 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1532 if (call!=NULL && call->audiostream!=NULL ) {
1534 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1535 if (strcasecmp(type,"mic")==0)
1536 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1537 else if (strcasecmp(type,"full")==0)
1538 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1540 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1545 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1546 if (call!=NULL && call->audiostream!=NULL ){
1547 return call->audiostream->el_type !=ELInactive ;
1549 return linphone_core_echo_limiter_enabled(call->core);
1554 * @addtogroup call_misc
1559 * Returns the measured sound volume played locally (received from remote).
1560 * It is expressed in dbm0.
1562 float linphone_call_get_play_volume(LinphoneCall *call){
1563 AudioStream *st=call->audiostream;
1564 if (st && st->volrecv){
1566 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1570 return LINPHONE_VOLUME_DB_LOWEST;
1574 * Returns the measured sound volume recorded locally (sent to remote).
1575 * It is expressed in dbm0.
1577 float linphone_call_get_record_volume(LinphoneCall *call){
1578 AudioStream *st=call->audiostream;
1579 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1581 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1585 return LINPHONE_VOLUME_DB_LOWEST;
1589 * Obtain real-time quality rating of the call
1591 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1592 * during all the duration of the call. This function returns its value at the time of the function call.
1593 * It is expected that the rating is updated at least every 5 seconds or so.
1594 * The rating is a floating point number comprised between 0 and 5.
1596 * 4-5 = good quality <br>
1597 * 3-4 = average quality <br>
1598 * 2-3 = poor quality <br>
1599 * 1-2 = very poor quality <br>
1600 * 0-1 = can't be worse, mostly unusable <br>
1602 * @returns The function returns -1 if no quality measurement is available, for example if no
1603 * active audio stream exist. Otherwise it returns the quality rating.
1605 float linphone_call_get_current_quality(LinphoneCall *call){
1606 if (call->audiostream){
1607 return audio_stream_get_quality_rating(call->audiostream);
1613 * Returns call quality averaged over all the duration of the call.
1615 * See linphone_call_get_current_quality() for more details about quality measurement.
1617 float linphone_call_get_average_quality(LinphoneCall *call){
1618 if (call->audiostream){
1619 return audio_stream_get_average_quality_rating(call->audiostream);
1625 * Access last known statistics for audio stream, for a given call.
1627 const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
1628 return &call->stats[LINPHONE_CALL_STATS_AUDIO];
1632 * Access last known statistics for video stream, for a given call.
1634 const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
1635 return &call->stats[LINPHONE_CALL_STATS_VIDEO];
1643 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1644 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1645 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1646 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1647 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1648 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1651 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1655 from = linphone_call_get_remote_address_as_string(call);
1658 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1663 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1665 if (lc->vtable.display_warning!=NULL)
1666 lc->vtable.display_warning(lc,temp);
1667 linphone_core_terminate_call(lc,call);
1670 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1671 LinphoneCore* lc = call->core;
1672 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1673 bool_t disconnected=FALSE;
1675 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1676 RtpSession *as=NULL,*vs=NULL;
1677 float audio_load=0, video_load=0;
1678 if (call->audiostream!=NULL){
1679 as=call->audiostream->session;
1680 if (call->audiostream->ticker)
1681 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1683 if (call->videostream!=NULL){
1684 if (call->videostream->ticker)
1685 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1686 vs=call->videostream->session;
1688 display_bandwidth(as,vs);
1689 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1691 #ifdef VIDEO_ENABLED
1692 if (call->videostream!=NULL) {
1693 // Beware that the application queue should not depend on treatments fron the
1694 // mediastreamer queue.
1695 video_stream_iterate(call->videostream);
1697 if (call->videostream_app_evq){
1699 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1700 OrtpEventType evt=ortp_event_get_type(ev);
1701 OrtpEventData *evd=ortp_event_get_data(ev);
1702 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1703 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1704 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1705 call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
1706 if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
1707 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
1708 call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
1710 if (lc->vtable.call_stats_updated)
1711 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1712 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1713 memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
1714 if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
1715 freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
1716 call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
1718 if (lc->vtable.call_stats_updated)
1719 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
1720 } else if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1721 if (ice_session_role(sal_op_get_ice_session(call->op)) == IR_Controlling) {
1722 linphone_core_update_call(lc, call, &call->current_params);
1724 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1725 if (call->state==LinphoneCallOutgoingInit) {
1726 linphone_call_stop_media_streams(call);
1727 if (evd->info.ice_processing_successful==FALSE) {
1728 ice_session_destroy(sal_op_get_ice_session(call->op));
1729 sal_op_set_ice_session(call->op, NULL);
1731 linphone_core_start_invite(call->core,call,NULL);
1734 ortp_event_destroy(ev);
1739 if (call->audiostream!=NULL) {
1740 // Beware that the application queue should not depend on treatments fron the
1741 // mediastreamer queue.
1742 audio_stream_iterate(call->audiostream);
1744 if (call->audiostream_app_evq){
1746 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1747 OrtpEventType evt=ortp_event_get_type(ev);
1748 OrtpEventData *evd=ortp_event_get_data(ev);
1749 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1750 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1751 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1752 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1753 } else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
1754 call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
1755 if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
1756 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
1757 call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
1759 if (lc->vtable.call_stats_updated)
1760 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1761 } else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
1762 memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
1763 if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
1764 freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
1765 call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
1767 if (lc->vtable.call_stats_updated)
1768 lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
1769 } else if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
1770 if (ice_session_role(sal_op_get_ice_session(call->op)) == IR_Controlling) {
1771 linphone_core_update_call(lc, call, &call->current_params);
1773 } else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
1774 if (call->state==LinphoneCallOutgoingInit) {
1775 linphone_call_stop_media_streams(call);
1776 if (evd->info.ice_processing_successful==FALSE) {
1777 ice_session_destroy(sal_op_get_ice_session(call->op));
1778 sal_op_set_ice_session(call->op, NULL);
1780 linphone_core_start_invite(call->core,call,NULL);
1783 ortp_event_destroy(ev);
1787 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1788 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1790 linphone_core_disconnected(call->core,call);
1793 void linphone_call_log_completed(LinphoneCall *call){
1794 LinphoneCore *lc=call->core;
1796 call->log->duration=time(NULL)-call->start_time;
1798 if (call->log->status==LinphoneCallMissed){
1801 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1802 "You have missed %i calls.", lc->missed_calls),
1804 if (lc->vtable.display_status!=NULL)
1805 lc->vtable.display_status(lc,info);
1808 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1809 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1810 MSList *elem,*prevelem=NULL;
1811 /*find the last element*/
1812 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1816 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1817 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1819 if (lc->vtable.call_log_updated!=NULL){
1820 lc->vtable.call_log_updated(lc,call->log);
1822 call_logs_write_to_config_file(lc);
1825 LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
1826 return call->transfer_state;
1829 void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
1830 if (state != call->transfer_state) {
1831 LinphoneCore* lc = call->core;
1832 call->transfer_state = state;
1833 if (lc->vtable.transfer_state_changed)
1834 lc->vtable.transfer_state_changed(lc, call, state);