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"
40 static MSWebCam *get_nowebcam_device(){
41 return ms_web_cam_manager_get_cam(ms_web_cam_manager_get(),"StaticImage: Static picture");
45 static bool_t generate_b64_crypto_key(int key_length, char* key_out) {
47 uint8_t* tmp = (uint8_t*) malloc(key_length);
48 if (ortp_crypto_get_random(tmp, key_length)!=0) {
49 ms_error("Failed to generate random key");
54 b64_size = b64_encode((const char*)tmp, key_length, NULL, 0);
56 ms_error("Failed to b64 encode key");
60 key_out[b64_size] = '\0';
61 b64_encode((const char*)tmp, key_length, key_out, 40);
66 LinphoneCore *linphone_call_get_core(const LinphoneCall *call){
70 const char* linphone_call_get_authentication_token(LinphoneCall *call){
71 return call->auth_token;
74 bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
75 return call->auth_token_verified;
78 static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
79 // Check ZRTP encryption in audiostream
80 if (!call->audiostream_encrypted) {
85 // If video enabled, check ZRTP encryption in videostream
86 const LinphoneCallParams *params=linphone_call_get_current_params(call);
87 if (params->has_video && !call->videostream_encrypted) {
95 void propagate_encryption_changed(LinphoneCall *call){
96 LinphoneCore *lc=call->core;
97 if (!linphone_call_are_all_streams_encrypted(call)) {
98 ms_message("Some streams are not encrypted");
99 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
100 if (lc->vtable.call_encryption_changed)
101 lc->vtable.call_encryption_changed(call->core, call, FALSE, call->auth_token);
103 ms_message("All streams are encrypted");
104 call->current_params.media_encryption=LinphoneMediaEncryptionZRTP;
105 if (lc->vtable.call_encryption_changed)
106 lc->vtable.call_encryption_changed(call->core, call, TRUE, call->auth_token);
111 static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
112 ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
114 LinphoneCall *call = (LinphoneCall *)data;
115 call->videostream_encrypted=encrypted;
116 propagate_encryption_changed(call);
120 static void linphone_call_audiostream_encryption_changed(void *data, bool_t encrypted) {
121 char status[255]={0};
122 ms_message("Audio stream is %s ", encrypted ? "encrypted" : "not encrypted");
124 LinphoneCall *call = (LinphoneCall *)data;
125 call->audiostream_encrypted=encrypted;
127 if (encrypted && call->core->vtable.display_status != NULL) {
128 snprintf(status,sizeof(status)-1,_("Authentication token is %s"),call->auth_token);
129 call->core->vtable.display_status(call->core, status);
132 propagate_encryption_changed(call);
136 // Enable video encryption
137 const LinphoneCallParams *params=linphone_call_get_current_params(call);
138 if (params->has_video) {
139 ms_message("Trying to enable encryption on video stream");
140 OrtpZrtpParams params;
141 params.zid_file=NULL; //unused
142 video_stream_enable_zrtp(call->videostream,call->audiostream,¶ms);
148 static void linphone_call_audiostream_auth_token_ready(void *data, const char* auth_token, bool_t verified) {
149 LinphoneCall *call=(LinphoneCall *)data;
150 if (call->auth_token != NULL)
151 ms_free(call->auth_token);
153 call->auth_token=ms_strdup(auth_token);
154 call->auth_token_verified=verified;
156 ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
159 void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t verified){
160 if (call->audiostream==NULL){
161 ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
163 if (call->audiostream->ortpZrtpContext==NULL){
164 ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
166 if (!call->auth_token_verified && verified){
167 ortp_zrtp_sas_verified(call->audiostream->ortpZrtpContext);
168 }else if (call->auth_token_verified && !verified){
169 ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
171 call->auth_token_verified=verified;
172 propagate_encryption_changed(call);
175 static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit){
178 for(it=codecs;it!=NULL;it=it->next){
179 PayloadType *pt=(PayloadType*)it->data;
180 if (pt->flags & PAYLOAD_TYPE_ENABLED){
181 if (bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,bandwidth_limit)){
182 ms_message("Codec %s/%i eliminated because of audio bandwidth constraint.",pt->mime_type,pt->clock_rate);
185 if (linphone_core_check_payload_type_usability(lc,pt)){
186 l=ms_list_append(l,payload_type_clone(pt));
193 static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
197 const char *me=linphone_core_get_identity(lc);
198 LinphoneAddress *addr=linphone_address_new(me);
199 const char *username=linphone_address_get_username (addr);
200 SalMediaDescription *md=sal_media_description_new();
202 md->session_id=session_id;
203 md->session_ver=session_ver;
205 strncpy(md->addr,call->localip,sizeof(md->addr));
206 strncpy(md->username,username,sizeof(md->username));
207 md->bandwidth=linphone_core_get_download_bandwidth(lc);
209 /*set audio capabilities */
210 strncpy(md->streams[0].addr,call->localip,sizeof(md->streams[0].addr));
211 md->streams[0].port=call->audio_port;
212 md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
213 SalProtoRtpSavp : SalProtoRtpAvp;
214 md->streams[0].type=SalAudio;
215 md->streams[0].ptime=lc->net_conf.down_ptime;
216 l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw);
217 pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
218 l=ms_list_append(l,pt);
219 md->streams[0].payloads=l;
222 if (call->params.has_video){
224 md->streams[1].port=call->video_port;
225 md->streams[1].proto=md->streams[0].proto;
226 md->streams[1].type=SalVideo;
227 l=make_codec_list(lc,lc->codecs_conf.video_codecs,0);
228 md->streams[1].payloads=l;
231 for(i=0; i<md->nstreams; i++) {
232 if (md->streams[i].proto == SalProtoRtpSavp) {
233 md->streams[i].crypto[0].tag = 1;
234 md->streams[i].crypto[0].algo = AES_128_SHA1_80;
235 if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
236 md->streams[i].crypto[0].algo = 0;
237 md->streams[i].crypto[1].tag = 2;
238 md->streams[i].crypto[1].algo = AES_128_SHA1_32;
239 if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
240 md->streams[i].crypto[1].algo = 0;
241 md->streams[i].crypto[2].algo = 0;
245 linphone_address_destroy(addr);
249 void update_local_media_description(LinphoneCore *lc, LinphoneCall *call, SalMediaDescription **md){
251 *md = _create_local_media_description(lc,call,0,0);
253 unsigned int id = (*md)->session_id;
254 unsigned int ver = (*md)->session_ver+1;
255 sal_media_description_unref(*md);
256 *md = _create_local_media_description(lc,call,id,ver);
260 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
261 unsigned int id=rand() & 0xfff;
262 return _create_local_media_description(lc,call,id,id);
265 static int find_port_offset(LinphoneCore *lc){
269 bool_t already_used=FALSE;
270 for(offset=0;offset<100;offset+=2){
271 audio_port=linphone_core_get_audio_port (lc)+offset;
273 for(elem=lc->calls;elem!=NULL;elem=elem->next){
274 LinphoneCall *call=(LinphoneCall*)elem->data;
275 if (call->audio_port==audio_port) {
280 if (!already_used) break;
283 ms_error("Could not find any free port !");
289 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
292 call->state=LinphoneCallIdle;
293 call->start_time=time(NULL);
294 call->media_start_time=0;
295 call->log=linphone_call_log_new(call, from, to);
296 call->owns_call_log=TRUE;
297 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
298 port_offset=find_port_offset (call->core);
299 if (port_offset==-1) return;
300 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
301 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
305 static void discover_mtu(LinphoneCore *lc, const char *remote){
307 if (lc->net_conf.mtu==0 ){
308 /*attempt to discover mtu*/
309 mtu=ms_discover_mtu(remote);
312 ms_message("Discovered mtu is %i, RTP payload max size is %i",
313 mtu, ms_get_payload_max_size());
318 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
320 LinphoneCall *call=ms_new0(LinphoneCall,1);
321 call->dir=LinphoneCallOutgoing;
322 call->op=sal_op_new(lc->sal);
323 sal_op_set_user_pointer(call->op,call);
325 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
326 linphone_call_init_common(call,from,to);
327 call->params=*params;
328 call->localdesc=create_local_media_description (lc,call);
329 call->camera_active=params->has_video;
330 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
331 linphone_core_run_stun_tests(call->core,call);
332 discover_mtu(lc,linphone_address_get_domain (to));
333 if (params->referer){
334 sal_call_set_referer(call->op,params->referer->op);
339 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
340 LinphoneCall *call=ms_new0(LinphoneCall,1);
343 call->dir=LinphoneCallIncoming;
344 sal_op_set_user_pointer(op,call);
348 if (lc->sip_conf.ping_with_options){
349 /*the following sends an option request back to the caller so that
350 we get a chance to discover our nat'd address before answering.*/
351 call->ping_op=sal_op_new(lc->sal);
352 from_str=linphone_address_as_string_uri_only(from);
353 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
354 sal_op_set_user_pointer(call->ping_op,call);
355 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
359 linphone_address_clean(from);
360 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
361 linphone_call_init_common(call, from, to);
362 linphone_core_init_default_params(lc, &call->params);
363 call->localdesc=create_local_media_description (lc,call);
364 call->camera_active=call->params.has_video;
365 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
366 linphone_core_run_stun_tests(call->core,call);
367 discover_mtu(lc,linphone_address_get_domain(from));
371 /* this function is called internally to get rid of a call.
372 It performs the following tasks:
373 - remove the call from the internal list of calls
374 - update the call logs accordingly
377 static void linphone_call_set_terminated(LinphoneCall *call){
378 LinphoneCore *lc=call->core;
380 linphone_core_update_allocated_audio_bandwidth(lc);
382 call->owns_call_log=FALSE;
383 linphone_call_log_completed(call);
386 if (call == lc->current_call){
387 ms_message("Resetting the current call");
388 lc->current_call=NULL;
391 if (linphone_core_del_call(lc,call) != 0){
392 ms_error("Could not remove the call from the list !!!");
395 if (ms_list_size(lc->calls)==0)
396 linphone_core_notify_all_friends(lc,lc->presence_mode);
398 linphone_core_conference_check_uninit(lc);
399 if (call->ringing_beep){
400 linphone_core_stop_dtmf(lc);
401 call->ringing_beep=FALSE;
405 const char *linphone_call_state_to_string(LinphoneCallState cs){
407 case LinphoneCallIdle:
408 return "LinphoneCallIdle";
409 case LinphoneCallIncomingReceived:
410 return "LinphoneCallIncomingReceived";
411 case LinphoneCallOutgoingInit:
412 return "LinphoneCallOutgoingInit";
413 case LinphoneCallOutgoingProgress:
414 return "LinphoneCallOutgoingProgress";
415 case LinphoneCallOutgoingRinging:
416 return "LinphoneCallOutgoingRinging";
417 case LinphoneCallOutgoingEarlyMedia:
418 return "LinphoneCallOutgoingEarlyMedia";
419 case LinphoneCallConnected:
420 return "LinphoneCallConnected";
421 case LinphoneCallStreamsRunning:
422 return "LinphoneCallStreamsRunning";
423 case LinphoneCallPausing:
424 return "LinphoneCallPausing";
425 case LinphoneCallPaused:
426 return "LinphoneCallPaused";
427 case LinphoneCallResuming:
428 return "LinphoneCallResuming";
429 case LinphoneCallRefered:
430 return "LinphoneCallRefered";
431 case LinphoneCallError:
432 return "LinphoneCallError";
433 case LinphoneCallEnd:
434 return "LinphoneCallEnd";
435 case LinphoneCallPausedByRemote:
436 return "LinphoneCallPausedByRemote";
437 case LinphoneCallUpdatedByRemote:
438 return "LinphoneCallUpdatedByRemote";
439 case LinphoneCallIncomingEarlyMedia:
440 return "LinphoneCallIncomingEarlyMedia";
441 case LinphoneCallUpdated:
442 return "LinphoneCallUpdated";
443 case LinphoneCallReleased:
444 return "LinphoneCallReleased";
446 return "undefined state";
449 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
450 LinphoneCore *lc=call->core;
452 if (call->state!=cstate){
453 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
454 if (cstate!=LinphoneCallReleased){
455 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
456 linphone_call_state_to_string(cstate));
460 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
461 linphone_call_state_to_string(cstate));
462 if (cstate!=LinphoneCallRefered){
463 /*LinphoneCallRefered is rather an event, not a state.
464 Indeed it does not change the state of the call (still paused or running)*/
467 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
468 if (call->reason==LinphoneReasonDeclined){
469 call->log->status=LinphoneCallDeclined;
471 linphone_call_set_terminated (call);
473 if (cstate == LinphoneCallConnected) {
474 call->log->status=LinphoneCallSuccess;
475 call->media_start_time=time(NULL);
478 if (lc->vtable.call_state_changed)
479 lc->vtable.call_state_changed(lc,call,cstate,message);
480 if (cstate==LinphoneCallReleased){
481 if (call->op!=NULL) {
482 /* so that we cannot have anymore upcalls for SAL
483 concerning this call*/
484 sal_op_release(call->op);
487 linphone_call_unref(call);
492 static void linphone_call_destroy(LinphoneCall *obj)
495 sal_op_release(obj->op);
498 if (obj->resultdesc!=NULL) {
499 sal_media_description_unref(obj->resultdesc);
500 obj->resultdesc=NULL;
502 if (obj->localdesc!=NULL) {
503 sal_media_description_unref(obj->localdesc);
507 sal_op_release(obj->ping_op);
510 ms_free(obj->refer_to);
512 if (obj->owns_call_log)
513 linphone_call_log_destroy(obj->log);
514 if (obj->auth_token) {
515 ms_free(obj->auth_token);
522 * @addtogroup call_control
527 * Increments the call 's reference count.
528 * An application that wishes to retain a pointer to call object
529 * must use this function to unsure the pointer remains
530 * valid. Once the application no more needs this pointer,
531 * it must call linphone_call_unref().
533 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
539 * Decrements the call object reference count.
540 * See linphone_call_ref().
542 void linphone_call_unref(LinphoneCall *obj){
545 linphone_call_destroy(obj);
550 * Returns current parameters associated to the call.
552 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
553 return &call->current_params;
557 * Returns the remote address associated to this call
560 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
561 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
565 * Returns the remote address associated to this call as a string.
567 * The result string must be freed by user using ms_free().
569 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
570 return linphone_address_as_string(linphone_call_get_remote_address(call));
574 * Retrieves the call's current state.
576 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
581 * Returns the reason for a call termination (either error or normal termination)
583 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
588 * Get the user_pointer in the LinphoneCall
590 * @ingroup call_control
592 * return user_pointer an opaque user pointer that can be retrieved at any time
594 void *linphone_call_get_user_pointer(LinphoneCall *call)
596 return call->user_pointer;
600 * Set the user_pointer in the LinphoneCall
602 * @ingroup call_control
604 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
606 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
608 call->user_pointer = user_pointer;
612 * Returns the call log associated to this call.
614 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
619 * Returns the refer-to uri (if the call was transfered).
621 const char *linphone_call_get_refer_to(const LinphoneCall *call){
622 return call->refer_to;
626 * Returns direction of the call (incoming or outgoing).
628 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
629 return call->log->dir;
633 * Returns the far end's user agent description string, if available.
635 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
637 return sal_op_get_remote_ua (call->op);
643 * Returns true if this calls has received a transfer that has not been
645 * Pending transfers are executed when this call is being paused or closed,
646 * locally or by remote endpoint.
647 * If the call is already paused while receiving the transfer request, the
648 * transfer immediately occurs.
650 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
651 return call->refer_pending;
655 * Returns call's duration in seconds.
657 int linphone_call_get_duration(const LinphoneCall *call){
658 if (call->media_start_time==0) return 0;
659 return time(NULL)-call->media_start_time;
663 * Returns the call object this call is replacing, if any.
664 * Call replacement can occur during call transfers.
665 * By default, the core automatically terminates the replaced call and accept the new one.
666 * This function allows the application to know whether a new incoming call is a one that replaces another one.
668 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
669 SalOp *op=sal_call_get_replaces(call->op);
671 return (LinphoneCall*)sal_op_get_user_pointer(op);
677 * Indicate whether camera input should be sent to remote end.
679 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
681 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
682 LinphoneCore *lc=call->core;
683 MSWebCam *nowebcam=get_nowebcam_device();
684 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
685 video_stream_change_camera(call->videostream,
686 enable ? lc->video_conf.device : nowebcam);
689 call->camera_active=enable;
694 * Take a photo of currently received video and write it into a jpeg file.
696 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
698 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
699 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
701 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
708 * Returns TRUE if camera pictures are sent to the remote party.
710 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
711 return call->camera_active;
715 * Enable video stream.
717 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
718 cp->has_video=enabled;
722 * Returns whether video is enabled.
724 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
725 return cp->has_video;
728 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
729 return cp->media_encryption;
732 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
733 cp->media_encryption = e;
738 * Enable sending of real early media (during outgoing calls).
740 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
741 cp->real_early_media=enabled;
744 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
745 return cp->real_early_media;
749 * Returns true if the call is part of the locally managed conference.
751 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
752 return cp->in_conference;
756 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
757 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
759 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
760 cp->audio_bw=bandwidth;
765 * Request remote side to send us a Video Fast Update.
767 void linphone_call_send_vfu_request(LinphoneCall *call)
769 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
770 sal_call_send_vfu_request(call->op);
777 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
778 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
779 memcpy(ncp,cp,sizeof(LinphoneCallParams));
786 void linphone_call_params_destroy(LinphoneCallParams *p){
795 #ifdef TEST_EXT_RENDERER
796 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
797 ms_message("rendercb, local buffer=%p, remote buffer=%p",
798 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
803 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
804 ms_warning("In linphonecall.c: video_stream_event_cb");
806 case MS_VIDEO_DECODER_DECODING_ERRORS:
807 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
808 linphone_call_send_vfu_request((LinphoneCall*) user_pointer);
811 ms_warning("Unhandled event %i", event_id);
817 void linphone_call_init_media_streams(LinphoneCall *call){
818 LinphoneCore *lc=call->core;
819 SalMediaDescription *md=call->localdesc;
820 AudioStream *audiostream;
822 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
823 if (linphone_core_echo_limiter_enabled(lc)){
824 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
825 if (strcasecmp(type,"mic")==0)
826 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
827 else if (strcasecmp(type,"full")==0)
828 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
830 audio_stream_enable_gain_control(audiostream,TRUE);
831 if (linphone_core_echo_cancellation_enabled(lc)){
832 int len,delay,framesize;
833 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
834 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
835 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
836 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
837 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
838 if (statestr && audiostream->ec){
839 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
842 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
844 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
845 audio_stream_enable_noise_gate(audiostream,enabled);
849 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
850 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
851 rtp_session_set_transports(audiostream->session,artp,artcp);
854 call->audiostream_app_evq = ortp_ev_queue_new();
855 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
859 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
860 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
861 if( lc->video_conf.displaytype != NULL)
862 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
863 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
865 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
866 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
867 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
869 call->videostream_app_evq = ortp_ev_queue_new();
870 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
871 #ifdef TEST_EXT_RENDERER
872 video_stream_set_render_callback(call->videostream,rendercb,NULL);
876 call->videostream=NULL;
881 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
883 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
884 LinphoneCore* lc = (LinphoneCore*)user_data;
885 if (dtmf<0 || dtmf>15){
886 ms_warning("Bad dtmf value %i",dtmf);
889 if (lc->vtable.dtmf_received != NULL)
890 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
893 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
895 MSFilter *f=st->equalizer;
896 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
897 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
898 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
904 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
905 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
906 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
915 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
916 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
919 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
920 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
921 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
924 audio_stream_set_mic_gain(st,mic_gain);
926 audio_stream_set_mic_gain(st,0);
928 recv_gain = lc->sound_conf.soft_play_lev;
929 if (recv_gain != 0) {
930 linphone_core_set_playback_gain_db (lc,recv_gain);
934 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
935 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
936 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
937 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
938 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
939 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
942 if (speed==-1) speed=0.03;
943 if (force==-1) force=25;
944 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
945 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
947 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
949 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
950 if (transmit_thres!=-1)
951 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
953 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
954 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
957 /* parameters for a limited noise-gate effect, using echo limiter threshold */
958 float floorgain = 1/mic_gain;
959 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
960 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
961 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
962 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
964 parametrize_equalizer(lc,st);
967 static void post_configure_audio_streams(LinphoneCall*call){
968 AudioStream *st=call->audiostream;
969 LinphoneCore *lc=call->core;
970 _post_configure_audio_stream(st,lc,call->audio_muted);
971 if (lc->vtable.dtmf_received!=NULL){
972 /* replace by our default action*/
973 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
974 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
978 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
981 RtpProfile *prof=rtp_profile_new("Call profile");
984 LinphoneCore *lc=call->core;
988 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
989 PayloadType *pt=(PayloadType*)elem->data;
992 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
993 if (desc->type==SalAudio){
994 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
995 up_ptime=linphone_core_get_upload_ptime(lc);
997 *used_pt=payload_type_get_number(pt);
1000 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1001 else if (md->bandwidth>0) {
1002 /*case where b=AS is given globally, not per stream*/
1003 remote_bw=md->bandwidth;
1004 if (desc->type==SalVideo){
1005 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1009 if (desc->type==SalAudio){
1010 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1011 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1012 if (bw>0) pt->normal_bitrate=bw*1000;
1013 else if (desc->type==SalAudio){
1014 pt->normal_bitrate=-1;
1017 up_ptime=desc->ptime;
1021 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1022 payload_type_append_send_fmtp(pt,tmp);
1024 number=payload_type_get_number(pt);
1025 if (rtp_profile_get_payload(prof,number)!=NULL){
1026 ms_warning("A payload type with number %i already exists in profile !",number);
1028 rtp_profile_set_payload(prof,number,pt);
1034 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1035 int pause_time=3000;
1036 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1037 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1040 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1042 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1043 LinphoneCore *lc=call->core;
1044 LinphoneCall *current=linphone_core_get_current_call(lc);
1045 return !linphone_core_is_in_conference(lc) &&
1046 (current==NULL || current==call);
1048 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1050 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1051 if (crypto[i].tag == tag) {
1057 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1058 LinphoneCore *lc=call->core;
1059 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1061 /* look for savp stream first */
1062 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1063 SalProtoRtpSavp,SalAudio);
1064 /* no savp audio stream, use avp */
1066 stream=sal_media_description_find_stream(call->resultdesc,
1067 SalProtoRtpAvp,SalAudio);
1069 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1070 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1071 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1072 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1073 const char *playfile=lc->play_file;
1074 const char *recfile=lc->rec_file;
1075 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1079 if (playcard==NULL) {
1080 ms_warning("No card defined for playback !");
1082 if (captcard==NULL) {
1083 ms_warning("No card defined for capture !");
1085 /*Replace soundcard filters by inactive file players or recorders
1086 when placed in recvonly or sendonly mode*/
1087 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1090 }else if (stream->dir==SalStreamSendOnly){
1094 /*And we will eventually play "playfile" if set by the user*/
1097 if (send_ringbacktone){
1099 playfile=NULL;/* it is setup later*/
1101 /*if playfile are supplied don't use soundcards*/
1102 if (lc->use_files) {
1106 if (call->params.in_conference){
1107 /* first create the graph without soundcard resources*/
1108 captcard=playcard=NULL;
1110 if (!linphone_call_sound_resources_available(call)){
1111 ms_message("Sound resources are used by another call, not using soundcard.");
1112 captcard=playcard=NULL;
1114 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1116 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1117 audio_stream_start_full(
1119 call->audio_profile,
1120 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1122 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1131 post_configure_audio_streams(call);
1132 if (muted && !send_ringbacktone){
1133 audio_stream_set_mic_gain(call->audiostream,0);
1135 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1137 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1139 if (send_ringbacktone){
1140 setup_ring_player(lc,call);
1142 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1144 if (stream->proto == SalProtoRtpSavp) {
1145 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1146 SalProtoRtpSavp,SalAudio);
1147 audio_stream_enable_strp(
1149 stream->crypto[0].algo,
1150 local_st_desc->crypto[find_crypto_index_from_tag(local_st_desc->crypto,stream->crypto[0].tag)].master_key,
1151 stream->crypto[0].master_key);
1152 call->audiostream_encrypted=TRUE;
1153 }else call->audiostream_encrypted=FALSE;
1154 if (call->params.in_conference){
1155 /*transform the graph to connect it to the conference filter */
1156 bool_t mute=stream->dir==SalStreamRecvOnly;
1157 linphone_call_add_to_conf(call, mute);
1159 call->current_params.in_conference=call->params.in_conference;
1160 }else ms_warning("No audio stream accepted ?");
1164 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1165 #ifdef VIDEO_ENABLED
1166 LinphoneCore *lc=call->core;
1168 /* look for savp stream first */
1169 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1170 SalProtoRtpSavp,SalVideo);
1171 /* no savp audio stream, use avp */
1173 vstream=sal_media_description_find_stream(call->resultdesc,
1174 SalProtoRtpAvp,SalVideo);
1176 /* shutdown preview */
1177 if (lc->previewstream!=NULL) {
1178 video_preview_stop(lc->previewstream);
1179 lc->previewstream=NULL;
1181 call->current_params.has_video=FALSE;
1182 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1183 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1184 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1186 VideoStreamDir dir=VideoStreamSendRecv;
1187 MSWebCam *cam=lc->video_conf.device;
1188 bool_t is_inactive=FALSE;
1190 call->current_params.has_video=TRUE;
1192 video_stream_enable_adaptive_bitrate_control(call->videostream,
1193 linphone_core_adaptive_rate_control_enabled(lc));
1194 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1195 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1196 if (lc->video_window_id!=0)
1197 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1198 if (lc->preview_window_id!=0)
1199 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1200 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1202 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1203 cam=get_nowebcam_device();
1204 dir=VideoStreamSendOnly;
1205 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1206 dir=VideoStreamRecvOnly;
1207 }else if (vstream->dir==SalStreamSendRecv){
1208 if (lc->video_conf.display && lc->video_conf.capture)
1209 dir=VideoStreamSendRecv;
1210 else if (lc->video_conf.display)
1211 dir=VideoStreamRecvOnly;
1213 dir=VideoStreamSendOnly;
1215 ms_warning("video stream is inactive.");
1216 /*either inactive or incompatible with local capabilities*/
1219 if (call->camera_active==FALSE || all_inputs_muted){
1220 cam=get_nowebcam_device();
1223 video_stream_set_direction (call->videostream, dir);
1224 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1225 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1226 video_stream_start(call->videostream,
1227 call->video_profile, addr, vstream->port,
1228 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1229 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1230 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1233 if (vstream->proto == SalProtoRtpSavp) {
1234 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1235 SalProtoRtpSavp,SalVideo);
1237 video_stream_enable_strp(
1239 vstream->crypto[0].algo,
1240 local_st_desc->crypto[0].master_key,
1241 vstream->crypto[0].master_key
1243 call->videostream_encrypted=TRUE;
1245 call->videostream_encrypted=FALSE;
1247 }else ms_warning("No video stream accepted.");
1249 ms_warning("No valid video stream defined.");
1254 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1255 LinphoneCore *lc=call->core;
1256 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1258 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1259 #ifdef VIDEO_ENABLED
1260 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1261 SalProtoRtpAvp,SalVideo);
1264 if(call->audiostream == NULL)
1266 ms_fatal("start_media_stream() called without prior init !");
1269 cname=linphone_address_as_string_uri_only(me);
1271 #if defined(VIDEO_ENABLED)
1272 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1273 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1277 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1278 if (call->videostream!=NULL) {
1279 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1282 call->all_muted=all_inputs_muted;
1283 call->playing_ringbacktone=send_ringbacktone;
1284 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1286 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1287 OrtpZrtpParams params;
1288 /*will be set later when zrtp is activated*/
1289 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1291 params.zid_file=lc->zrtp_secrets_cache;
1292 audio_stream_enable_zrtp(call->audiostream,¶ms);
1293 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1294 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1295 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1296 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP again
1297 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1298 call->params.media_encryption=call->current_params.media_encryption;
1304 linphone_address_destroy(me);
1307 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1308 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1309 log->quality=audio_stream_get_average_quality_rating(st);
1312 void linphone_call_stop_media_streams(LinphoneCall *call){
1313 if (call->audiostream!=NULL) {
1314 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1315 ortp_ev_queue_flush(call->audiostream_app_evq);
1316 ortp_ev_queue_destroy(call->audiostream_app_evq);
1318 if (call->audiostream->ec){
1319 const char *state_str=NULL;
1320 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1322 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1323 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1326 linphone_call_log_fill_stats (call->log,call->audiostream);
1327 if (call->endpoint){
1328 linphone_call_remove_from_conf(call);
1330 audio_stream_stop(call->audiostream);
1331 call->audiostream=NULL;
1335 #ifdef VIDEO_ENABLED
1336 if (call->videostream!=NULL){
1337 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1338 ortp_ev_queue_flush(call->videostream_app_evq);
1339 ortp_ev_queue_destroy(call->videostream_app_evq);
1340 video_stream_stop(call->videostream);
1341 call->videostream=NULL;
1344 ms_event_queue_skip(call->core->msevq);
1346 if (call->audio_profile){
1347 rtp_profile_clear_all(call->audio_profile);
1348 rtp_profile_destroy(call->audio_profile);
1349 call->audio_profile=NULL;
1351 if (call->video_profile){
1352 rtp_profile_clear_all(call->video_profile);
1353 rtp_profile_destroy(call->video_profile);
1354 call->video_profile=NULL;
1360 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1361 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1362 bool_t bypass_mode = !enable;
1363 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1366 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1367 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1369 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1372 return linphone_core_echo_cancellation_enabled(call->core);
1376 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1377 if (call!=NULL && call->audiostream!=NULL ) {
1379 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1380 if (strcasecmp(type,"mic")==0)
1381 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1382 else if (strcasecmp(type,"full")==0)
1383 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1385 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1390 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1391 if (call!=NULL && call->audiostream!=NULL ){
1392 return call->audiostream->el_type !=ELInactive ;
1394 return linphone_core_echo_limiter_enabled(call->core);
1399 * @addtogroup call_misc
1404 * Returns the measured sound volume played locally (received from remote)
1405 * It is expressed in dbm0.
1407 float linphone_call_get_play_volume(LinphoneCall *call){
1408 AudioStream *st=call->audiostream;
1409 if (st && st->volrecv){
1411 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1415 return LINPHONE_VOLUME_DB_LOWEST;
1419 * Returns the measured sound volume recorded locally (sent to remote)
1420 * It is expressed in dbm0.
1422 float linphone_call_get_record_volume(LinphoneCall *call){
1423 AudioStream *st=call->audiostream;
1424 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1426 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1430 return LINPHONE_VOLUME_DB_LOWEST;
1434 * Obtain real-time quality rating of the call
1436 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1437 * during all the duration of the call. This function returns its value at the time of the function call.
1438 * It is expected that the rating is updated at least every 5 seconds or so.
1439 * The rating is a floating point number comprised between 0 and 5.
1441 * 4-5 = good quality <br>
1442 * 3-4 = average quality <br>
1443 * 2-3 = poor quality <br>
1444 * 1-2 = very poor quality <br>
1445 * 0-1 = can't be worse, mostly unusable <br>
1447 * @returns The function returns -1 if no quality measurement is available, for example if no
1448 * active audio stream exist. Otherwise it returns the quality rating.
1450 float linphone_call_get_current_quality(LinphoneCall *call){
1451 if (call->audiostream){
1452 return audio_stream_get_quality_rating(call->audiostream);
1458 * Returns call quality averaged over all the duration of the call.
1460 * See linphone_call_get_current_quality() for more details about quality measurement.
1462 float linphone_call_get_average_quality(LinphoneCall *call){
1463 if (call->audiostream){
1464 return audio_stream_get_average_quality_rating(call->audiostream);
1473 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1474 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1475 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1476 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1477 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1478 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1481 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1485 from = linphone_call_get_remote_address_as_string(call);
1488 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1493 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1495 if (lc->vtable.display_warning!=NULL)
1496 lc->vtable.display_warning(lc,temp);
1497 linphone_core_terminate_call(lc,call);
1500 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1501 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1502 bool_t disconnected=FALSE;
1504 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1505 RtpSession *as=NULL,*vs=NULL;
1506 float audio_load=0, video_load=0;
1507 if (call->audiostream!=NULL){
1508 as=call->audiostream->session;
1509 if (call->audiostream->ticker)
1510 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1512 if (call->videostream!=NULL){
1513 if (call->videostream->ticker)
1514 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1515 vs=call->videostream->session;
1517 display_bandwidth(as,vs);
1518 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1520 #ifdef VIDEO_ENABLED
1521 if (call->videostream!=NULL) {
1522 // Beware that the application queue should not depend on treatments fron the
1523 // mediastreamer queue.
1524 video_stream_iterate(call->videostream);
1526 if (call->videostream_app_evq){
1528 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1529 OrtpEventType evt=ortp_event_get_type(ev);
1530 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1531 OrtpEventData *evd=ortp_event_get_data(ev);
1532 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1534 ortp_event_destroy(ev);
1539 if (call->audiostream!=NULL) {
1540 // Beware that the application queue should not depend on treatments fron the
1541 // mediastreamer queue.
1542 audio_stream_iterate(call->audiostream);
1544 if (call->audiostream_app_evq){
1546 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1547 OrtpEventType evt=ortp_event_get_type(ev);
1548 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1549 OrtpEventData *evd=ortp_event_get_data(ev);
1550 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1551 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1552 OrtpEventData *evd=ortp_event_get_data(ev);
1553 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1555 ortp_event_destroy(ev);
1559 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1560 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1562 linphone_core_disconnected(call->core,call);
1565 void linphone_call_log_completed(LinphoneCall *call){
1566 LinphoneCore *lc=call->core;
1568 call->log->duration=time(NULL)-call->start_time;
1570 if (call->log->status==LinphoneCallMissed){
1573 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1574 "You have missed %i calls.", lc->missed_calls),
1576 if (lc->vtable.display_status!=NULL)
1577 lc->vtable.display_status(lc,info);
1580 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1581 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1582 MSList *elem,*prevelem=NULL;
1583 /*find the last element*/
1584 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1588 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1589 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1591 if (lc->vtable.call_log_updated!=NULL){
1592 lc->vtable.call_log_updated(lc,call->log);
1594 call_logs_write_to_config_file(lc);