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){
250 SalMediaDescription *md=call->localdesc;
252 call->localdesc = create_local_media_description(lc,call);
254 call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
255 sal_media_description_unref(md);
259 SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
260 unsigned int id=rand() & 0xfff;
261 return _create_local_media_description(lc,call,id,id);
264 static int find_port_offset(LinphoneCore *lc){
268 bool_t already_used=FALSE;
269 for(offset=0;offset<100;offset+=2){
270 audio_port=linphone_core_get_audio_port (lc)+offset;
272 for(elem=lc->calls;elem!=NULL;elem=elem->next){
273 LinphoneCall *call=(LinphoneCall*)elem->data;
274 if (call->audio_port==audio_port) {
279 if (!already_used) break;
282 ms_error("Could not find any free port !");
288 static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
291 call->state=LinphoneCallIdle;
292 call->start_time=time(NULL);
293 call->media_start_time=0;
294 call->log=linphone_call_log_new(call, from, to);
295 call->owns_call_log=TRUE;
296 linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
297 port_offset=find_port_offset (call->core);
298 if (port_offset==-1) return;
299 call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
300 call->video_port=linphone_core_get_video_port(call->core)+port_offset;
304 static void discover_mtu(LinphoneCore *lc, const char *remote){
306 if (lc->net_conf.mtu==0 ){
307 /*attempt to discover mtu*/
308 mtu=ms_discover_mtu(remote);
311 ms_message("Discovered mtu is %i, RTP payload max size is %i",
312 mtu, ms_get_payload_max_size());
317 LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
319 LinphoneCall *call=ms_new0(LinphoneCall,1);
320 call->dir=LinphoneCallOutgoing;
321 call->op=sal_op_new(lc->sal);
322 sal_op_set_user_pointer(call->op,call);
324 linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
325 linphone_call_init_common(call,from,to);
326 call->params=*params;
327 call->localdesc=create_local_media_description (lc,call);
328 call->camera_active=params->has_video;
329 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
330 linphone_core_run_stun_tests(call->core,call);
331 discover_mtu(lc,linphone_address_get_domain (to));
332 if (params->referer){
333 sal_call_set_referer(call->op,params->referer->op);
338 LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
339 LinphoneCall *call=ms_new0(LinphoneCall,1);
342 call->dir=LinphoneCallIncoming;
343 sal_op_set_user_pointer(op,call);
347 if (lc->sip_conf.ping_with_options){
348 /*the following sends an option request back to the caller so that
349 we get a chance to discover our nat'd address before answering.*/
350 call->ping_op=sal_op_new(lc->sal);
351 from_str=linphone_address_as_string_uri_only(from);
352 sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
353 sal_op_set_user_pointer(call->ping_op,call);
354 sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
358 linphone_address_clean(from);
359 linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
360 linphone_call_init_common(call, from, to);
361 linphone_core_init_default_params(lc, &call->params);
362 call->localdesc=create_local_media_description (lc,call);
363 call->camera_active=call->params.has_video;
364 if (linphone_core_get_firewall_policy(call->core)==LinphonePolicyUseStun)
365 linphone_core_run_stun_tests(call->core,call);
366 discover_mtu(lc,linphone_address_get_domain(from));
370 /* this function is called internally to get rid of a call.
371 It performs the following tasks:
372 - remove the call from the internal list of calls
373 - update the call logs accordingly
376 static void linphone_call_set_terminated(LinphoneCall *call){
377 LinphoneCore *lc=call->core;
379 linphone_core_update_allocated_audio_bandwidth(lc);
381 call->owns_call_log=FALSE;
382 linphone_call_log_completed(call);
385 if (call == lc->current_call){
386 ms_message("Resetting the current call");
387 lc->current_call=NULL;
390 if (linphone_core_del_call(lc,call) != 0){
391 ms_error("Could not remove the call from the list !!!");
394 if (ms_list_size(lc->calls)==0)
395 linphone_core_notify_all_friends(lc,lc->presence_mode);
397 linphone_core_conference_check_uninit(lc);
398 if (call->ringing_beep){
399 linphone_core_stop_dtmf(lc);
400 call->ringing_beep=FALSE;
404 void linphone_call_fix_call_parameters(LinphoneCall *call){
405 call->params.has_video=call->current_params.has_video;
406 call->params.media_encryption=call->current_params.media_encryption;
409 const char *linphone_call_state_to_string(LinphoneCallState cs){
411 case LinphoneCallIdle:
412 return "LinphoneCallIdle";
413 case LinphoneCallIncomingReceived:
414 return "LinphoneCallIncomingReceived";
415 case LinphoneCallOutgoingInit:
416 return "LinphoneCallOutgoingInit";
417 case LinphoneCallOutgoingProgress:
418 return "LinphoneCallOutgoingProgress";
419 case LinphoneCallOutgoingRinging:
420 return "LinphoneCallOutgoingRinging";
421 case LinphoneCallOutgoingEarlyMedia:
422 return "LinphoneCallOutgoingEarlyMedia";
423 case LinphoneCallConnected:
424 return "LinphoneCallConnected";
425 case LinphoneCallStreamsRunning:
426 return "LinphoneCallStreamsRunning";
427 case LinphoneCallPausing:
428 return "LinphoneCallPausing";
429 case LinphoneCallPaused:
430 return "LinphoneCallPaused";
431 case LinphoneCallResuming:
432 return "LinphoneCallResuming";
433 case LinphoneCallRefered:
434 return "LinphoneCallRefered";
435 case LinphoneCallError:
436 return "LinphoneCallError";
437 case LinphoneCallEnd:
438 return "LinphoneCallEnd";
439 case LinphoneCallPausedByRemote:
440 return "LinphoneCallPausedByRemote";
441 case LinphoneCallUpdatedByRemote:
442 return "LinphoneCallUpdatedByRemote";
443 case LinphoneCallIncomingEarlyMedia:
444 return "LinphoneCallIncomingEarlyMedia";
445 case LinphoneCallUpdated:
446 return "LinphoneCallUpdated";
447 case LinphoneCallReleased:
448 return "LinphoneCallReleased";
450 return "undefined state";
453 void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
454 LinphoneCore *lc=call->core;
456 if (call->state!=cstate){
457 if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
458 if (cstate!=LinphoneCallReleased){
459 ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
460 linphone_call_state_to_string(cstate));
464 ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
465 linphone_call_state_to_string(cstate));
466 if (cstate!=LinphoneCallRefered){
467 /*LinphoneCallRefered is rather an event, not a state.
468 Indeed it does not change the state of the call (still paused or running)*/
471 if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
472 if (call->reason==LinphoneReasonDeclined){
473 call->log->status=LinphoneCallDeclined;
475 linphone_call_set_terminated (call);
477 if (cstate == LinphoneCallConnected) {
478 call->log->status=LinphoneCallSuccess;
479 call->media_start_time=time(NULL);
482 if (lc->vtable.call_state_changed)
483 lc->vtable.call_state_changed(lc,call,cstate,message);
484 if (cstate==LinphoneCallReleased){
485 if (call->op!=NULL) {
486 /* so that we cannot have anymore upcalls for SAL
487 concerning this call*/
488 sal_op_release(call->op);
491 linphone_call_unref(call);
496 static void linphone_call_destroy(LinphoneCall *obj)
499 sal_op_release(obj->op);
502 if (obj->resultdesc!=NULL) {
503 sal_media_description_unref(obj->resultdesc);
504 obj->resultdesc=NULL;
506 if (obj->localdesc!=NULL) {
507 sal_media_description_unref(obj->localdesc);
511 sal_op_release(obj->ping_op);
514 ms_free(obj->refer_to);
516 if (obj->owns_call_log)
517 linphone_call_log_destroy(obj->log);
518 if (obj->auth_token) {
519 ms_free(obj->auth_token);
526 * @addtogroup call_control
531 * Increments the call 's reference count.
532 * An application that wishes to retain a pointer to call object
533 * must use this function to unsure the pointer remains
534 * valid. Once the application no more needs this pointer,
535 * it must call linphone_call_unref().
537 LinphoneCall * linphone_call_ref(LinphoneCall *obj){
543 * Decrements the call object reference count.
544 * See linphone_call_ref().
546 void linphone_call_unref(LinphoneCall *obj){
549 linphone_call_destroy(obj);
554 * Returns current parameters associated to the call.
556 const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
557 return &call->current_params;
560 static bool_t is_video_active(const SalStreamDescription *sd){
561 return sd->port!=0 && sd->dir!=SalStreamInactive;
565 * Returns call parameters proposed by remote.
567 * This is useful when receiving an incoming call, to know whether the remote party
568 * supports video, encryption or whatever.
570 const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
571 LinphoneCallParams *cp=&call->remote_params;
572 memset(cp,0,sizeof(*cp));
574 SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
576 SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
578 asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
579 vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
580 secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
581 secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
583 cp->has_video=is_video_active(secure_vsd);
584 if (secure_asd || asd==NULL)
585 cp->media_encryption=LinphoneMediaEncryptionSRTP;
587 cp->has_video=is_video_active(vsd);
596 * Returns the remote address associated to this call
599 const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
600 return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
604 * Returns the remote address associated to this call as a string.
606 * The result string must be freed by user using ms_free().
608 char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
609 return linphone_address_as_string(linphone_call_get_remote_address(call));
613 * Retrieves the call's current state.
615 LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
620 * Returns the reason for a call termination (either error or normal termination)
622 LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
627 * Get the user_pointer in the LinphoneCall
629 * @ingroup call_control
631 * return user_pointer an opaque user pointer that can be retrieved at any time
633 void *linphone_call_get_user_pointer(LinphoneCall *call)
635 return call->user_pointer;
639 * Set the user_pointer in the LinphoneCall
641 * @ingroup call_control
643 * the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
645 void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
647 call->user_pointer = user_pointer;
651 * Returns the call log associated to this call.
653 LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
658 * Returns the refer-to uri (if the call was transfered).
660 const char *linphone_call_get_refer_to(const LinphoneCall *call){
661 return call->refer_to;
665 * Returns direction of the call (incoming or outgoing).
667 LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
668 return call->log->dir;
672 * Returns the far end's user agent description string, if available.
674 const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
676 return sal_op_get_remote_ua (call->op);
682 * Returns true if this calls has received a transfer that has not been
684 * Pending transfers are executed when this call is being paused or closed,
685 * locally or by remote endpoint.
686 * If the call is already paused while receiving the transfer request, the
687 * transfer immediately occurs.
689 bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
690 return call->refer_pending;
694 * Returns call's duration in seconds.
696 int linphone_call_get_duration(const LinphoneCall *call){
697 if (call->media_start_time==0) return 0;
698 return time(NULL)-call->media_start_time;
702 * Returns the call object this call is replacing, if any.
703 * Call replacement can occur during call transfers.
704 * By default, the core automatically terminates the replaced call and accept the new one.
705 * This function allows the application to know whether a new incoming call is a one that replaces another one.
707 LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
708 SalOp *op=sal_call_get_replaces(call->op);
710 return (LinphoneCall*)sal_op_get_user_pointer(op);
716 * Indicate whether camera input should be sent to remote end.
718 void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
720 if (call->videostream!=NULL && call->videostream->ticker!=NULL){
721 LinphoneCore *lc=call->core;
722 MSWebCam *nowebcam=get_nowebcam_device();
723 if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
724 video_stream_change_camera(call->videostream,
725 enable ? lc->video_conf.device : nowebcam);
728 call->camera_active=enable;
733 * Take a photo of currently received video and write it into a jpeg file.
735 int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
737 if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
738 return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
740 ms_warning("Cannot take snapshot: no currently running video stream on this call.");
747 * Returns TRUE if camera pictures are sent to the remote party.
749 bool_t linphone_call_camera_enabled (const LinphoneCall *call){
750 return call->camera_active;
754 * Enable video stream.
756 void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
757 cp->has_video=enabled;
761 * Returns whether video is enabled.
763 bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
764 return cp->has_video;
767 enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
768 return cp->media_encryption;
771 void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
772 cp->media_encryption = e;
777 * Enable sending of real early media (during outgoing calls).
779 void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
780 cp->real_early_media=enabled;
783 bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
784 return cp->real_early_media;
788 * Returns true if the call is part of the locally managed conference.
790 bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
791 return cp->in_conference;
795 * Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
796 * As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
798 void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
799 cp->audio_bw=bandwidth;
804 * Request remote side to send us a Video Fast Update.
806 void linphone_call_send_vfu_request(LinphoneCall *call)
808 if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
809 sal_call_send_vfu_request(call->op);
816 LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
817 LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
818 memcpy(ncp,cp,sizeof(LinphoneCallParams));
825 void linphone_call_params_destroy(LinphoneCallParams *p){
834 #ifdef TEST_EXT_RENDERER
835 static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
836 ms_message("rendercb, local buffer=%p, remote buffer=%p",
837 local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
842 static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
843 ms_warning("In linphonecall.c: video_stream_event_cb");
845 case MS_VIDEO_DECODER_DECODING_ERRORS:
846 ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
847 linphone_call_send_vfu_request((LinphoneCall*) user_pointer);
850 ms_warning("Unhandled event %i", event_id);
856 void linphone_call_init_media_streams(LinphoneCall *call){
857 LinphoneCore *lc=call->core;
858 SalMediaDescription *md=call->localdesc;
859 AudioStream *audiostream;
861 call->audiostream=audiostream=audio_stream_new(md->streams[0].port,linphone_core_ipv6_enabled(lc));
862 if (linphone_core_echo_limiter_enabled(lc)){
863 const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
864 if (strcasecmp(type,"mic")==0)
865 audio_stream_enable_echo_limiter(audiostream,ELControlMic);
866 else if (strcasecmp(type,"full")==0)
867 audio_stream_enable_echo_limiter(audiostream,ELControlFull);
869 audio_stream_enable_gain_control(audiostream,TRUE);
870 if (linphone_core_echo_cancellation_enabled(lc)){
871 int len,delay,framesize;
872 const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
873 len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
874 delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
875 framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
876 audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
877 if (statestr && audiostream->ec){
878 ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
881 audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
883 int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
884 audio_stream_enable_noise_gate(audiostream,enabled);
888 RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
889 RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
890 rtp_session_set_transports(audiostream->session,artp,artcp);
893 call->audiostream_app_evq = ortp_ev_queue_new();
894 rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
898 if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].port>0){
899 call->videostream=video_stream_new(md->streams[1].port,linphone_core_ipv6_enabled(lc));
900 if( lc->video_conf.displaytype != NULL)
901 video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
902 video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
904 RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
905 RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
906 rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
908 call->videostream_app_evq = ortp_ev_queue_new();
909 rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
910 #ifdef TEST_EXT_RENDERER
911 video_stream_set_render_callback(call->videostream,rendercb,NULL);
915 call->videostream=NULL;
920 static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
922 static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
923 LinphoneCore* lc = (LinphoneCore*)user_data;
924 if (dtmf<0 || dtmf>15){
925 ms_warning("Bad dtmf value %i",dtmf);
928 if (lc->vtable.dtmf_received != NULL)
929 lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
932 static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
934 MSFilter *f=st->equalizer;
935 int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
936 const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
937 ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
943 if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
944 ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
945 ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
954 void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
955 float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
958 float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
959 float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
960 int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
963 audio_stream_set_mic_gain(st,mic_gain);
965 audio_stream_set_mic_gain(st,0);
967 recv_gain = lc->sound_conf.soft_play_lev;
968 if (recv_gain != 0) {
969 linphone_core_set_playback_gain_db (lc,recv_gain);
973 ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
974 float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
975 thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
976 float force=lp_config_get_float(lc->config,"sound","el_force",-1);
977 int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
978 float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
981 if (speed==-1) speed=0.03;
982 if (force==-1) force=25;
983 ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
984 ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
986 ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
988 ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
989 if (transmit_thres!=-1)
990 ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
992 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
993 ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
996 /* parameters for a limited noise-gate effect, using echo limiter threshold */
997 float floorgain = 1/mic_gain;
998 int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
999 ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
1000 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
1001 ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
1003 parametrize_equalizer(lc,st);
1006 static void post_configure_audio_streams(LinphoneCall*call){
1007 AudioStream *st=call->audiostream;
1008 LinphoneCore *lc=call->core;
1009 _post_configure_audio_stream(st,lc,call->audio_muted);
1010 if (lc->vtable.dtmf_received!=NULL){
1011 /* replace by our default action*/
1012 audio_stream_play_received_dtmfs(call->audiostream,FALSE);
1013 rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
1017 static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
1020 RtpProfile *prof=rtp_profile_new("Call profile");
1023 LinphoneCore *lc=call->core;
1027 for(elem=desc->payloads;elem!=NULL;elem=elem->next){
1028 PayloadType *pt=(PayloadType*)elem->data;
1031 if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
1032 if (desc->type==SalAudio){
1033 linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
1034 up_ptime=linphone_core_get_upload_ptime(lc);
1036 *used_pt=payload_type_get_number(pt);
1039 if (desc->bandwidth>0) remote_bw=desc->bandwidth;
1040 else if (md->bandwidth>0) {
1041 /*case where b=AS is given globally, not per stream*/
1042 remote_bw=md->bandwidth;
1043 if (desc->type==SalVideo){
1044 remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
1048 if (desc->type==SalAudio){
1049 bw=get_min_bandwidth(call->audio_bw,remote_bw);
1050 }else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
1051 if (bw>0) pt->normal_bitrate=bw*1000;
1052 else if (desc->type==SalAudio){
1053 pt->normal_bitrate=-1;
1056 up_ptime=desc->ptime;
1060 snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
1061 payload_type_append_send_fmtp(pt,tmp);
1063 number=payload_type_get_number(pt);
1064 if (rtp_profile_get_payload(prof,number)!=NULL){
1065 ms_warning("A payload type with number %i already exists in profile !",number);
1067 rtp_profile_set_payload(prof,number,pt);
1073 static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
1074 int pause_time=3000;
1075 audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
1076 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1079 #define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
1081 static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
1082 LinphoneCore *lc=call->core;
1083 LinphoneCall *current=linphone_core_get_current_call(lc);
1084 return !linphone_core_is_in_conference(lc) &&
1085 (current==NULL || current==call);
1087 static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
1089 for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
1090 if (crypto[i].tag == tag) {
1096 static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
1097 LinphoneCore *lc=call->core;
1098 int jitt_comp=lc->rtp_conf.audio_jitt_comp;
1100 /* look for savp stream first */
1101 const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
1102 SalProtoRtpSavp,SalAudio);
1103 /* no savp audio stream, use avp */
1105 stream=sal_media_description_find_stream(call->resultdesc,
1106 SalProtoRtpAvp,SalAudio);
1108 if (stream && stream->dir!=SalStreamInactive && stream->port!=0){
1109 MSSndCard *playcard=lc->sound_conf.lsd_card ?
1110 lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
1111 MSSndCard *captcard=lc->sound_conf.capt_sndcard;
1112 const char *playfile=lc->play_file;
1113 const char *recfile=lc->rec_file;
1114 call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
1118 if (playcard==NULL) {
1119 ms_warning("No card defined for playback !");
1121 if (captcard==NULL) {
1122 ms_warning("No card defined for capture !");
1124 /*Replace soundcard filters by inactive file players or recorders
1125 when placed in recvonly or sendonly mode*/
1126 if (stream->port==0 || stream->dir==SalStreamRecvOnly){
1129 }else if (stream->dir==SalStreamSendOnly){
1133 /*And we will eventually play "playfile" if set by the user*/
1136 if (send_ringbacktone){
1138 playfile=NULL;/* it is setup later*/
1140 /*if playfile are supplied don't use soundcards*/
1141 if (lc->use_files) {
1145 if (call->params.in_conference){
1146 /* first create the graph without soundcard resources*/
1147 captcard=playcard=NULL;
1149 if (!linphone_call_sound_resources_available(call)){
1150 ms_message("Sound resources are used by another call, not using soundcard.");
1151 captcard=playcard=NULL;
1153 use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
1155 audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
1156 audio_stream_start_full(
1158 call->audio_profile,
1159 stream->addr[0]!='\0' ? stream->addr : call->resultdesc->addr,
1161 linphone_core_rtcp_enabled(lc) ? (stream->port+1) : 0,
1170 post_configure_audio_streams(call);
1171 if (muted && !send_ringbacktone){
1172 audio_stream_set_mic_gain(call->audiostream,0);
1174 if (stream->dir==SalStreamSendOnly && playfile!=NULL){
1176 ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
1178 if (send_ringbacktone){
1179 setup_ring_player(lc,call);
1181 audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
1183 if (stream->proto == SalProtoRtpSavp) {
1184 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1185 SalProtoRtpSavp,SalAudio);
1186 audio_stream_enable_strp(
1188 stream->crypto[0].algo,
1189 local_st_desc->crypto[find_crypto_index_from_tag(local_st_desc->crypto,stream->crypto[0].tag)].master_key,
1190 stream->crypto[0].master_key);
1191 call->audiostream_encrypted=TRUE;
1192 }else call->audiostream_encrypted=FALSE;
1193 if (call->params.in_conference){
1194 /*transform the graph to connect it to the conference filter */
1195 bool_t mute=stream->dir==SalStreamRecvOnly;
1196 linphone_call_add_to_conf(call, mute);
1198 call->current_params.in_conference=call->params.in_conference;
1199 }else ms_warning("No audio stream accepted ?");
1203 static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
1204 #ifdef VIDEO_ENABLED
1205 LinphoneCore *lc=call->core;
1207 /* look for savp stream first */
1208 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1209 SalProtoRtpSavp,SalVideo);
1210 /* no savp audio stream, use avp */
1212 vstream=sal_media_description_find_stream(call->resultdesc,
1213 SalProtoRtpAvp,SalVideo);
1215 /* shutdown preview */
1216 if (lc->previewstream!=NULL) {
1217 video_preview_stop(lc->previewstream);
1218 lc->previewstream=NULL;
1220 call->current_params.has_video=FALSE;
1221 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->port!=0) {
1222 const char *addr=vstream->addr[0]!='\0' ? vstream->addr : call->resultdesc->addr;
1223 call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
1225 VideoStreamDir dir=VideoStreamSendRecv;
1226 MSWebCam *cam=lc->video_conf.device;
1227 bool_t is_inactive=FALSE;
1229 call->current_params.has_video=TRUE;
1231 video_stream_enable_adaptive_bitrate_control(call->videostream,
1232 linphone_core_adaptive_rate_control_enabled(lc));
1233 video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
1234 video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
1235 if (lc->video_window_id!=0)
1236 video_stream_set_native_window_id(call->videostream,lc->video_window_id);
1237 if (lc->preview_window_id!=0)
1238 video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
1239 video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
1241 if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
1242 cam=get_nowebcam_device();
1243 dir=VideoStreamSendOnly;
1244 }else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
1245 dir=VideoStreamRecvOnly;
1246 }else if (vstream->dir==SalStreamSendRecv){
1247 if (lc->video_conf.display && lc->video_conf.capture)
1248 dir=VideoStreamSendRecv;
1249 else if (lc->video_conf.display)
1250 dir=VideoStreamRecvOnly;
1252 dir=VideoStreamSendOnly;
1254 ms_warning("video stream is inactive.");
1255 /*either inactive or incompatible with local capabilities*/
1258 if (call->camera_active==FALSE || all_inputs_muted){
1259 cam=get_nowebcam_device();
1262 video_stream_set_direction (call->videostream, dir);
1263 ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
1264 video_stream_set_device_rotation(call->videostream, lc->device_rotation);
1265 video_stream_start(call->videostream,
1266 call->video_profile, addr, vstream->port,
1267 linphone_core_rtcp_enabled(lc) ? (vstream->port+1) : 0,
1268 used_pt, lc->rtp_conf.audio_jitt_comp, cam);
1269 video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
1272 if (vstream->proto == SalProtoRtpSavp) {
1273 const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
1274 SalProtoRtpSavp,SalVideo);
1276 video_stream_enable_strp(
1278 vstream->crypto[0].algo,
1279 local_st_desc->crypto[0].master_key,
1280 vstream->crypto[0].master_key
1282 call->videostream_encrypted=TRUE;
1284 call->videostream_encrypted=FALSE;
1286 }else ms_warning("No video stream accepted.");
1288 ms_warning("No valid video stream defined.");
1293 void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
1294 LinphoneCore *lc=call->core;
1295 LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
1297 bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
1298 #ifdef VIDEO_ENABLED
1299 const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
1300 SalProtoRtpAvp,SalVideo);
1303 if(call->audiostream == NULL)
1305 ms_fatal("start_media_stream() called without prior init !");
1308 cname=linphone_address_as_string_uri_only(me);
1310 #if defined(VIDEO_ENABLED)
1311 if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
1312 /*when video is used, do not make adaptive rate control on audio, it is stupid.*/
1316 linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
1317 if (call->videostream!=NULL) {
1318 linphone_call_start_video_stream(call,cname,all_inputs_muted);
1321 call->all_muted=all_inputs_muted;
1322 call->playing_ringbacktone=send_ringbacktone;
1323 call->up_bw=linphone_core_get_upload_bandwidth(lc);
1325 if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
1326 OrtpZrtpParams params;
1327 /*will be set later when zrtp is activated*/
1328 call->current_params.media_encryption=LinphoneMediaEncryptionNone;
1330 params.zid_file=lc->zrtp_secrets_cache;
1331 audio_stream_enable_zrtp(call->audiostream,¶ms);
1332 }else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
1333 call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
1334 LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
1337 /*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
1338 * further in the call, for example during pause,resume, conferencing reINVITEs*/
1339 linphone_call_fix_call_parameters(call);
1344 linphone_address_destroy(me);
1347 static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
1348 audio_stream_get_local_rtp_stats (st,&log->local_stats);
1349 log->quality=audio_stream_get_average_quality_rating(st);
1352 void linphone_call_stop_media_streams(LinphoneCall *call){
1353 if (call->audiostream!=NULL) {
1354 rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
1355 ortp_ev_queue_flush(call->audiostream_app_evq);
1356 ortp_ev_queue_destroy(call->audiostream_app_evq);
1358 if (call->audiostream->ec){
1359 const char *state_str=NULL;
1360 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
1362 ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
1363 lp_config_set_string(call->core->config,"sound","ec_state",state_str);
1366 linphone_call_log_fill_stats (call->log,call->audiostream);
1367 if (call->endpoint){
1368 linphone_call_remove_from_conf(call);
1370 audio_stream_stop(call->audiostream);
1371 call->audiostream=NULL;
1375 #ifdef VIDEO_ENABLED
1376 if (call->videostream!=NULL){
1377 rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
1378 ortp_ev_queue_flush(call->videostream_app_evq);
1379 ortp_ev_queue_destroy(call->videostream_app_evq);
1380 video_stream_stop(call->videostream);
1381 call->videostream=NULL;
1384 ms_event_queue_skip(call->core->msevq);
1386 if (call->audio_profile){
1387 rtp_profile_clear_all(call->audio_profile);
1388 rtp_profile_destroy(call->audio_profile);
1389 call->audio_profile=NULL;
1391 if (call->video_profile){
1392 rtp_profile_clear_all(call->video_profile);
1393 rtp_profile_destroy(call->video_profile);
1394 call->video_profile=NULL;
1400 void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
1401 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1402 bool_t bypass_mode = !enable;
1403 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
1406 bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
1407 if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
1409 ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
1412 return linphone_core_echo_cancellation_enabled(call->core);
1416 void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
1417 if (call!=NULL && call->audiostream!=NULL ) {
1419 const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
1420 if (strcasecmp(type,"mic")==0)
1421 audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
1422 else if (strcasecmp(type,"full")==0)
1423 audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
1425 audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
1430 bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
1431 if (call!=NULL && call->audiostream!=NULL ){
1432 return call->audiostream->el_type !=ELInactive ;
1434 return linphone_core_echo_limiter_enabled(call->core);
1439 * @addtogroup call_misc
1444 * Returns the measured sound volume played locally (received from remote)
1445 * It is expressed in dbm0.
1447 float linphone_call_get_play_volume(LinphoneCall *call){
1448 AudioStream *st=call->audiostream;
1449 if (st && st->volrecv){
1451 ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
1455 return LINPHONE_VOLUME_DB_LOWEST;
1459 * Returns the measured sound volume recorded locally (sent to remote)
1460 * It is expressed in dbm0.
1462 float linphone_call_get_record_volume(LinphoneCall *call){
1463 AudioStream *st=call->audiostream;
1464 if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
1466 ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
1470 return LINPHONE_VOLUME_DB_LOWEST;
1474 * Obtain real-time quality rating of the call
1476 * Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
1477 * during all the duration of the call. This function returns its value at the time of the function call.
1478 * It is expected that the rating is updated at least every 5 seconds or so.
1479 * The rating is a floating point number comprised between 0 and 5.
1481 * 4-5 = good quality <br>
1482 * 3-4 = average quality <br>
1483 * 2-3 = poor quality <br>
1484 * 1-2 = very poor quality <br>
1485 * 0-1 = can't be worse, mostly unusable <br>
1487 * @returns The function returns -1 if no quality measurement is available, for example if no
1488 * active audio stream exist. Otherwise it returns the quality rating.
1490 float linphone_call_get_current_quality(LinphoneCall *call){
1491 if (call->audiostream){
1492 return audio_stream_get_quality_rating(call->audiostream);
1498 * Returns call quality averaged over all the duration of the call.
1500 * See linphone_call_get_current_quality() for more details about quality measurement.
1502 float linphone_call_get_average_quality(LinphoneCall *call){
1503 if (call->audiostream){
1504 return audio_stream_get_average_quality_rating(call->audiostream);
1513 static void display_bandwidth(RtpSession *as, RtpSession *vs){
1514 ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
1515 (as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
1516 (as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
1517 (vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
1518 (vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
1521 static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
1525 from = linphone_call_get_remote_address_as_string(call);
1528 snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
1533 snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
1535 if (lc->vtable.display_warning!=NULL)
1536 lc->vtable.display_warning(lc,temp);
1537 linphone_core_terminate_call(lc,call);
1540 void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
1541 int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
1542 bool_t disconnected=FALSE;
1544 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
1545 RtpSession *as=NULL,*vs=NULL;
1546 float audio_load=0, video_load=0;
1547 if (call->audiostream!=NULL){
1548 as=call->audiostream->session;
1549 if (call->audiostream->ticker)
1550 audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
1552 if (call->videostream!=NULL){
1553 if (call->videostream->ticker)
1554 video_load=ms_ticker_get_average_load(call->videostream->ticker);
1555 vs=call->videostream->session;
1557 display_bandwidth(as,vs);
1558 ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
1560 #ifdef VIDEO_ENABLED
1561 if (call->videostream!=NULL) {
1562 // Beware that the application queue should not depend on treatments fron the
1563 // mediastreamer queue.
1564 video_stream_iterate(call->videostream);
1566 if (call->videostream_app_evq){
1568 while (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq))){
1569 OrtpEventType evt=ortp_event_get_type(ev);
1570 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1571 OrtpEventData *evd=ortp_event_get_data(ev);
1572 linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1574 ortp_event_destroy(ev);
1579 if (call->audiostream!=NULL) {
1580 // Beware that the application queue should not depend on treatments fron the
1581 // mediastreamer queue.
1582 audio_stream_iterate(call->audiostream);
1584 if (call->audiostream_app_evq){
1586 while (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq))){
1587 OrtpEventType evt=ortp_event_get_type(ev);
1588 if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
1589 OrtpEventData *evd=ortp_event_get_data(ev);
1590 linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
1591 } else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
1592 OrtpEventData *evd=ortp_event_get_data(ev);
1593 linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
1595 ortp_event_destroy(ev);
1599 if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
1600 disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
1602 linphone_core_disconnected(call->core,call);
1605 void linphone_call_log_completed(LinphoneCall *call){
1606 LinphoneCore *lc=call->core;
1608 call->log->duration=time(NULL)-call->start_time;
1610 if (call->log->status==LinphoneCallMissed){
1613 info=ortp_strdup_printf(ngettext("You have missed %i call.",
1614 "You have missed %i calls.", lc->missed_calls),
1616 if (lc->vtable.display_status!=NULL)
1617 lc->vtable.display_status(lc,info);
1620 lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
1621 if (ms_list_size(lc->call_logs)>lc->max_call_logs){
1622 MSList *elem,*prevelem=NULL;
1623 /*find the last element*/
1624 for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
1628 linphone_call_log_destroy((LinphoneCallLog*)elem->data);
1629 lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
1631 if (lc->vtable.call_log_updated!=NULL){
1632 lc->vtable.call_log_updated(lc,call->log);
1634 call_logs_write_to_config_file(lc);