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linphone-ios/coreapi/linphonecall.c
Ghislain MARY 7b5bddf359 Deactivate symmetric RTP when ICE is in use.
2012-08-14 12:37:21 +02:00

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/*
linphone
Copyright (C) 2010 Belledonne Communications SARL
(simon.morlat@linphone.org)
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
#ifdef WIN32
#include <time.h>
#endif
#include "linphonecore.h"
#include "sipsetup.h"
#include "lpconfig.h"
#include "private.h"
#include <ortp/event.h>
#include <ortp/b64.h>
#include "mediastreamer2/mediastream.h"
#include "mediastreamer2/msvolume.h"
#include "mediastreamer2/msequalizer.h"
#include "mediastreamer2/msfileplayer.h"
#include "mediastreamer2/msjpegwriter.h"
#include "mediastreamer2/mseventqueue.h"
#include "mediastreamer2/mssndcard.h"
#ifdef VIDEO_ENABLED
static MSWebCam *get_nowebcam_device(){
return ms_web_cam_manager_get_cam(ms_web_cam_manager_get(),"StaticImage: Static picture");
}
#endif
static bool_t generate_b64_crypto_key(int key_length, char* key_out) {
int b64_size;
uint8_t* tmp = (uint8_t*) malloc(key_length);
if (ortp_crypto_get_random(tmp, key_length)!=0) {
ms_error("Failed to generate random key");
free(tmp);
return FALSE;
}
b64_size = b64_encode((const char*)tmp, key_length, NULL, 0);
if (b64_size == 0) {
ms_error("Failed to b64 encode key");
free(tmp);
return FALSE;
}
key_out[b64_size] = '\0';
b64_encode((const char*)tmp, key_length, key_out, 40);
free(tmp);
return TRUE;
}
LinphoneCore *linphone_call_get_core(const LinphoneCall *call){
return call->core;
}
const char* linphone_call_get_authentication_token(LinphoneCall *call){
return call->auth_token;
}
bool_t linphone_call_get_authentication_token_verified(LinphoneCall *call){
return call->auth_token_verified;
}
static bool_t linphone_call_are_all_streams_encrypted(LinphoneCall *call) {
// Check ZRTP encryption in audiostream
if (!call->audiostream_encrypted) {
return FALSE;
}
#ifdef VIDEO_ENABLED
// If video enabled, check ZRTP encryption in videostream
const LinphoneCallParams *params=linphone_call_get_current_params(call);
if (params->has_video && !call->videostream_encrypted) {
return FALSE;
}
#endif
return TRUE;
}
void propagate_encryption_changed(LinphoneCall *call){
LinphoneCore *lc=call->core;
if (!linphone_call_are_all_streams_encrypted(call)) {
ms_message("Some streams are not encrypted");
call->current_params.media_encryption=LinphoneMediaEncryptionNone;
if (lc->vtable.call_encryption_changed)
lc->vtable.call_encryption_changed(call->core, call, FALSE, call->auth_token);
} else {
ms_message("All streams are encrypted");
call->current_params.media_encryption=LinphoneMediaEncryptionZRTP;
if (lc->vtable.call_encryption_changed)
lc->vtable.call_encryption_changed(call->core, call, TRUE, call->auth_token);
}
}
#ifdef VIDEO_ENABLED
static void linphone_call_videostream_encryption_changed(void *data, bool_t encrypted){
ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
LinphoneCall *call = (LinphoneCall *)data;
call->videostream_encrypted=encrypted;
propagate_encryption_changed(call);
}
#endif
static void linphone_call_audiostream_encryption_changed(void *data, bool_t encrypted) {
char status[255]={0};
ms_message("Audio stream is %s ", encrypted ? "encrypted" : "not encrypted");
LinphoneCall *call = (LinphoneCall *)data;
call->audiostream_encrypted=encrypted;
if (encrypted && call->core->vtable.display_status != NULL) {
snprintf(status,sizeof(status)-1,_("Authentication token is %s"),call->auth_token);
call->core->vtable.display_status(call->core, status);
}
propagate_encryption_changed(call);
#ifdef VIDEO_ENABLED
// Enable video encryption
const LinphoneCallParams *params=linphone_call_get_current_params(call);
if (params->has_video) {
ms_message("Trying to enable encryption on video stream");
OrtpZrtpParams params;
params.zid_file=NULL; //unused
video_stream_enable_zrtp(call->videostream,call->audiostream,&params);
}
#endif
}
static void linphone_call_audiostream_auth_token_ready(void *data, const char* auth_token, bool_t verified) {
LinphoneCall *call=(LinphoneCall *)data;
if (call->auth_token != NULL)
ms_free(call->auth_token);
call->auth_token=ms_strdup(auth_token);
call->auth_token_verified=verified;
ms_message("Authentication token is %s (%s)", auth_token, verified?"verified":"unverified");
}
void linphone_call_set_authentication_token_verified(LinphoneCall *call, bool_t verified){
if (call->audiostream==NULL){
ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
}
if (call->audiostream->ortpZrtpContext==NULL){
ms_error("linphone_call_set_authentication_token_verified(): No zrtp context.");
}
if (!call->auth_token_verified && verified){
ortp_zrtp_sas_verified(call->audiostream->ortpZrtpContext);
}else if (call->auth_token_verified && !verified){
ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
}
call->auth_token_verified=verified;
propagate_encryption_changed(call);
}
static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate){
MSList *l=NULL;
const MSList *it;
if (max_sample_rate) *max_sample_rate=0;
for(it=codecs;it!=NULL;it=it->next){
PayloadType *pt=(PayloadType*)it->data;
if (pt->flags & PAYLOAD_TYPE_ENABLED){
if (bandwidth_limit>0 && !linphone_core_is_payload_type_usable_for_bandwidth(lc,pt,bandwidth_limit)){
ms_message("Codec %s/%i eliminated because of audio bandwidth constraint.",pt->mime_type,pt->clock_rate);
continue;
}
if (linphone_core_check_payload_type_usability(lc,pt)){
l=ms_list_append(l,payload_type_clone(pt));
if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
}
}
}
return l;
}
static SalMediaDescription *_create_local_media_description(LinphoneCore *lc, LinphoneCall *call, unsigned int session_id, unsigned int session_ver){
MSList *l;
PayloadType *pt;
int i;
const char *me=linphone_core_get_identity(lc);
LinphoneAddress *addr=linphone_address_new(me);
const char *username=linphone_address_get_username (addr);
SalMediaDescription *md=sal_media_description_new();
md->session_id=session_id;
md->session_ver=session_ver;
md->nstreams=1;
strncpy(md->addr,call->localip,sizeof(md->addr));
strncpy(md->username,username,sizeof(md->username));
md->bandwidth=linphone_core_get_download_bandwidth(lc);
/*set audio capabilities */
strncpy(md->streams[0].rtp_addr,call->localip,sizeof(md->streams[0].rtp_addr));
strncpy(md->streams[0].rtcp_addr,call->localip,sizeof(md->streams[0].rtcp_addr));
md->streams[0].rtp_port=call->audio_port;
md->streams[0].rtcp_port=call->audio_port+1;
md->streams[0].proto=(call->params.media_encryption == LinphoneMediaEncryptionSRTP) ?
SalProtoRtpSavp : SalProtoRtpAvp;
md->streams[0].type=SalAudio;
md->streams[0].ptime=lc->net_conf.down_ptime;
l=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate);
pt=payload_type_clone(rtp_profile_get_payload_from_mime(&av_profile,"telephone-event"));
l=ms_list_append(l,pt);
md->streams[0].payloads=l;
if (call->params.has_video){
md->nstreams++;
md->streams[1].rtp_port=call->video_port;
md->streams[1].rtcp_port=call->video_port+1;
md->streams[1].proto=md->streams[0].proto;
md->streams[1].type=SalVideo;
l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL);
md->streams[1].payloads=l;
}
for(i=0; i<md->nstreams; i++) {
if (md->streams[i].proto == SalProtoRtpSavp) {
md->streams[i].crypto[0].tag = 1;
md->streams[i].crypto[0].algo = AES_128_SHA1_80;
if (!generate_b64_crypto_key(30, md->streams[i].crypto[0].master_key))
md->streams[i].crypto[0].algo = 0;
md->streams[i].crypto[1].tag = 2;
md->streams[i].crypto[1].algo = AES_128_SHA1_32;
if (!generate_b64_crypto_key(30, md->streams[i].crypto[1].master_key))
md->streams[i].crypto[1].algo = 0;
md->streams[i].crypto[2].algo = 0;
}
if ((linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, i) == NULL)) {
ice_session_add_check_list(call->ice_session, ice_check_list_new());
}
}
linphone_address_destroy(addr);
return md;
}
void update_local_media_description(LinphoneCore *lc, LinphoneCall *call){
SalMediaDescription *md=call->localdesc;
if (md== NULL) {
call->localdesc = create_local_media_description(lc,call);
} else {
call->localdesc = _create_local_media_description(lc,call,md->session_id,md->session_ver+1);
sal_media_description_unref(md);
}
}
SalMediaDescription *create_local_media_description(LinphoneCore *lc, LinphoneCall *call){
unsigned int id=rand() & 0xfff;
return _create_local_media_description(lc,call,id,id);
}
static int find_port_offset(LinphoneCore *lc){
int offset;
MSList *elem;
int audio_port;
bool_t already_used=FALSE;
for(offset=0;offset<100;offset+=2){
audio_port=linphone_core_get_audio_port (lc)+offset;
already_used=FALSE;
for(elem=lc->calls;elem!=NULL;elem=elem->next){
LinphoneCall *call=(LinphoneCall*)elem->data;
if (call->audio_port==audio_port) {
already_used=TRUE;
break;
}
}
if (!already_used) break;
}
if (offset==100){
ms_error("Could not find any free port !");
return -1;
}
return offset;
}
static void linphone_call_init_common(LinphoneCall *call, LinphoneAddress *from, LinphoneAddress *to){
int port_offset;
call->magic=linphone_call_magic;
call->refcnt=1;
call->state=LinphoneCallIdle;
call->transfer_state = LinphoneCallIdle;
call->start_time=time(NULL);
call->media_start_time=0;
call->log=linphone_call_log_new(call, from, to);
call->owns_call_log=TRUE;
linphone_core_notify_all_friends(call->core,LinphoneStatusOnThePhone);
port_offset=find_port_offset (call->core);
if (port_offset==-1) return;
call->audio_port=linphone_core_get_audio_port(call->core)+port_offset;
call->video_port=linphone_core_get_video_port(call->core)+port_offset;
linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO], LINPHONE_CALL_STATS_AUDIO);
linphone_call_init_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO], LINPHONE_CALL_STATS_VIDEO);
}
void linphone_call_init_stats(LinphoneCallStats *stats, int type) {
stats->type = type;
stats->received_rtcp = NULL;
stats->sent_rtcp = NULL;
}
static void discover_mtu(LinphoneCore *lc, const char *remote){
int mtu;
if (lc->net_conf.mtu==0 ){
/*attempt to discover mtu*/
mtu=ms_discover_mtu(remote);
if (mtu>0){
ms_set_mtu(mtu);
ms_message("Discovered mtu is %i, RTP payload max size is %i",
mtu, ms_get_payload_max_size());
}
}
}
LinphoneCall * linphone_call_new_outgoing(struct _LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, const LinphoneCallParams *params)
{
LinphoneCall *call=ms_new0(LinphoneCall,1);
call->dir=LinphoneCallOutgoing;
call->op=sal_op_new(lc->sal);
sal_op_set_user_pointer(call->op,call);
call->core=lc;
linphone_core_get_local_ip(lc,linphone_address_get_domain(to),call->localip);
linphone_call_init_common(call,from,to);
call->params=*params;
if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
call->ice_session = ice_session_new();
ice_session_set_role(call->ice_session, IR_Controlling);
}
call->localdesc=create_local_media_description (lc,call);
call->camera_active=params->has_video;
if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
linphone_core_run_stun_tests(call->core,call);
}
discover_mtu(lc,linphone_address_get_domain (to));
if (params->referer){
sal_call_set_referer(call->op,params->referer->op);
call->referer=linphone_call_ref(params->referer);
}
return call;
}
LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
LinphoneCall *call=ms_new0(LinphoneCall,1);
char *from_str;
call->dir=LinphoneCallIncoming;
sal_op_set_user_pointer(op,call);
call->op=op;
call->core=lc;
if (lc->sip_conf.ping_with_options){
/*the following sends an option request back to the caller so that
we get a chance to discover our nat'd address before answering.*/
call->ping_op=sal_op_new(lc->sal);
from_str=linphone_address_as_string_uri_only(from);
sal_op_set_route(call->ping_op,sal_op_get_network_origin(op));
sal_op_set_user_pointer(call->ping_op,call);
sal_ping(call->ping_op,linphone_core_find_best_identity(lc,from,NULL),from_str);
ms_free(from_str);
}
linphone_address_clean(from);
linphone_core_get_local_ip(lc,linphone_address_get_domain(from),call->localip);
linphone_call_init_common(call, from, to);
linphone_core_init_default_params(lc, &call->params);
call->params.has_video &= !!lc->video_policy.automatically_accept;
call->localdesc=create_local_media_description (lc,call);
call->camera_active=call->params.has_video;
switch (linphone_core_get_firewall_policy(call->core)) {
case LinphonePolicyUseIce:
call->ice_session = ice_session_new();
ice_session_set_role(call->ice_session, IR_Controlled);
linphone_core_update_ice_from_remote_media_description(call, sal_call_get_remote_media_description(op));
if (call->ice_session != NULL) {
linphone_call_init_media_streams(call);
linphone_call_start_media_streams_for_ice_gathering(call);
if (linphone_core_gather_ice_candidates(call->core,call)<0) {
/* Ice candidates gathering failed, proceed with the call anyway. */
linphone_call_delete_ice_session(call);
linphone_call_stop_media_streams(call);
}
}
break;
case LinphonePolicyUseStun:
linphone_core_run_stun_tests(call->core,call);
/* No break to also destroy ice session in this case. */
default:
break;
}
discover_mtu(lc,linphone_address_get_domain(from));
return call;
}
/* this function is called internally to get rid of a call.
It performs the following tasks:
- remove the call from the internal list of calls
- update the call logs accordingly
*/
static void linphone_call_set_terminated(LinphoneCall *call){
LinphoneCore *lc=call->core;
linphone_core_update_allocated_audio_bandwidth(lc);
call->owns_call_log=FALSE;
linphone_call_log_completed(call);
if (call == lc->current_call){
ms_message("Resetting the current call");
lc->current_call=NULL;
}
if (linphone_core_del_call(lc,call) != 0){
ms_error("Could not remove the call from the list !!!");
}
if (ms_list_size(lc->calls)==0)
linphone_core_notify_all_friends(lc,lc->presence_mode);
linphone_core_conference_check_uninit(lc);
if (call->ringing_beep){
linphone_core_stop_dtmf(lc);
call->ringing_beep=FALSE;
}
if (call->referer){
linphone_call_unref(call->referer);
call->referer=NULL;
}
}
void linphone_call_fix_call_parameters(LinphoneCall *call){
call->params.has_video=call->current_params.has_video;
call->params.media_encryption=call->current_params.media_encryption;
}
const char *linphone_call_state_to_string(LinphoneCallState cs){
switch (cs){
case LinphoneCallIdle:
return "LinphoneCallIdle";
case LinphoneCallIncomingReceived:
return "LinphoneCallIncomingReceived";
case LinphoneCallOutgoingInit:
return "LinphoneCallOutgoingInit";
case LinphoneCallOutgoingProgress:
return "LinphoneCallOutgoingProgress";
case LinphoneCallOutgoingRinging:
return "LinphoneCallOutgoingRinging";
case LinphoneCallOutgoingEarlyMedia:
return "LinphoneCallOutgoingEarlyMedia";
case LinphoneCallConnected:
return "LinphoneCallConnected";
case LinphoneCallStreamsRunning:
return "LinphoneCallStreamsRunning";
case LinphoneCallPausing:
return "LinphoneCallPausing";
case LinphoneCallPaused:
return "LinphoneCallPaused";
case LinphoneCallResuming:
return "LinphoneCallResuming";
case LinphoneCallRefered:
return "LinphoneCallRefered";
case LinphoneCallError:
return "LinphoneCallError";
case LinphoneCallEnd:
return "LinphoneCallEnd";
case LinphoneCallPausedByRemote:
return "LinphoneCallPausedByRemote";
case LinphoneCallUpdatedByRemote:
return "LinphoneCallUpdatedByRemote";
case LinphoneCallIncomingEarlyMedia:
return "LinphoneCallIncomingEarlyMedia";
case LinphoneCallUpdated:
return "LinphoneCallUpdated";
case LinphoneCallReleased:
return "LinphoneCallReleased";
}
return "undefined state";
}
void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const char *message){
LinphoneCore *lc=call->core;
if (call->state!=cstate){
if (call->state==LinphoneCallEnd || call->state==LinphoneCallError){
if (cstate!=LinphoneCallReleased){
ms_warning("Spurious call state change from %s to %s, ignored.",linphone_call_state_to_string(call->state),
linphone_call_state_to_string(cstate));
return;
}
}
ms_message("Call %p: moving from state %s to %s",call,linphone_call_state_to_string(call->state),
linphone_call_state_to_string(cstate));
if (cstate!=LinphoneCallRefered){
/*LinphoneCallRefered is rather an event, not a state.
Indeed it does not change the state of the call (still paused or running)*/
call->state=cstate;
}
if (cstate==LinphoneCallEnd || cstate==LinphoneCallError){
switch(call->reason){
case LinphoneReasonDeclined:
call->log->status=LinphoneCallDeclined;
break;
case LinphoneReasonNotAnswered:
call->log->status=LinphoneCallMissed;
break;
default:
break;
}
linphone_call_set_terminated (call);
}
if (cstate == LinphoneCallConnected) {
call->log->status=LinphoneCallSuccess;
call->media_start_time=time(NULL);
}
if (lc->vtable.call_state_changed)
lc->vtable.call_state_changed(lc,call,cstate,message);
if (cstate==LinphoneCallReleased){
if (call->op!=NULL) {
/* so that we cannot have anymore upcalls for SAL
concerning this call*/
sal_op_release(call->op);
call->op=NULL;
}
linphone_call_unref(call);
}
}
}
static void linphone_call_destroy(LinphoneCall *obj)
{
if (obj->op!=NULL) {
sal_op_release(obj->op);
obj->op=NULL;
}
if (obj->resultdesc!=NULL) {
sal_media_description_unref(obj->resultdesc);
obj->resultdesc=NULL;
}
if (obj->localdesc!=NULL) {
sal_media_description_unref(obj->localdesc);
obj->localdesc=NULL;
}
if (obj->ping_op) {
sal_op_release(obj->ping_op);
}
if (obj->refer_to){
ms_free(obj->refer_to);
}
if (obj->owns_call_log)
linphone_call_log_destroy(obj->log);
if (obj->auth_token) {
ms_free(obj->auth_token);
}
if (obj->ice_session) {
ice_session_destroy(obj->ice_session);
}
ms_free(obj);
}
/**
* @addtogroup call_control
* @{
**/
/**
* Increments the call 's reference count.
* An application that wishes to retain a pointer to call object
* must use this function to unsure the pointer remains
* valid. Once the application no more needs this pointer,
* it must call linphone_call_unref().
**/
LinphoneCall * linphone_call_ref(LinphoneCall *obj){
obj->refcnt++;
return obj;
}
/**
* Decrements the call object reference count.
* See linphone_call_ref().
**/
void linphone_call_unref(LinphoneCall *obj){
obj->refcnt--;
if (obj->refcnt==0){
linphone_call_destroy(obj);
}
}
/**
* Returns current parameters associated to the call.
**/
const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
return &call->current_params;
}
static bool_t is_video_active(const SalStreamDescription *sd){
return sd->rtp_port!=0 && sd->dir!=SalStreamInactive;
}
/**
* Returns call parameters proposed by remote.
*
* This is useful when receiving an incoming call, to know whether the remote party
* supports video, encryption or whatever.
**/
const LinphoneCallParams * linphone_call_get_remote_params(LinphoneCall *call){
LinphoneCallParams *cp=&call->remote_params;
memset(cp,0,sizeof(*cp));
if (call->op){
SalMediaDescription *md=sal_call_get_remote_media_description(call->op);
if (md){
SalStreamDescription *asd,*vsd,*secure_asd,*secure_vsd;
asd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalAudio);
vsd=sal_media_description_find_stream(md,SalProtoRtpAvp,SalVideo);
secure_asd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalAudio);
secure_vsd=sal_media_description_find_stream(md,SalProtoRtpSavp,SalVideo);
if (secure_vsd){
cp->has_video=is_video_active(secure_vsd);
if (secure_asd || asd==NULL)
cp->media_encryption=LinphoneMediaEncryptionSRTP;
}else if (vsd){
cp->has_video=is_video_active(vsd);
}
return cp;
}
}
return NULL;
}
/**
* Returns the remote address associated to this call
*
**/
const LinphoneAddress * linphone_call_get_remote_address(const LinphoneCall *call){
return call->dir==LinphoneCallIncoming ? call->log->from : call->log->to;
}
/**
* Returns the remote address associated to this call as a string.
*
* The result string must be freed by user using ms_free().
**/
char *linphone_call_get_remote_address_as_string(const LinphoneCall *call){
return linphone_address_as_string(linphone_call_get_remote_address(call));
}
/**
* Retrieves the call's current state.
**/
LinphoneCallState linphone_call_get_state(const LinphoneCall *call){
return call->state;
}
/**
* Returns the reason for a call termination (either error or normal termination)
**/
LinphoneReason linphone_call_get_reason(const LinphoneCall *call){
return call->reason;
}
/**
* Get the user_pointer in the LinphoneCall
*
* @ingroup call_control
*
* return user_pointer an opaque user pointer that can be retrieved at any time
**/
void *linphone_call_get_user_pointer(LinphoneCall *call)
{
return call->user_pointer;
}
/**
* Set the user_pointer in the LinphoneCall
*
* @ingroup call_control
*
* the user_pointer is an opaque user pointer that can be retrieved at any time in the LinphoneCall
**/
void linphone_call_set_user_pointer(LinphoneCall *call, void *user_pointer)
{
call->user_pointer = user_pointer;
}
/**
* Returns the call log associated to this call.
**/
LinphoneCallLog *linphone_call_get_call_log(const LinphoneCall *call){
return call->log;
}
/**
* Returns the refer-to uri (if the call was transfered).
**/
const char *linphone_call_get_refer_to(const LinphoneCall *call){
return call->refer_to;
}
/**
* Returns direction of the call (incoming or outgoing).
**/
LinphoneCallDir linphone_call_get_dir(const LinphoneCall *call){
return call->log->dir;
}
/**
* Returns the far end's user agent description string, if available.
**/
const char *linphone_call_get_remote_user_agent(LinphoneCall *call){
if (call->op){
return sal_op_get_remote_ua (call->op);
}
return NULL;
}
/**
* Returns true if this calls has received a transfer that has not been
* executed yet.
* Pending transfers are executed when this call is being paused or closed,
* locally or by remote endpoint.
* If the call is already paused while receiving the transfer request, the
* transfer immediately occurs.
**/
bool_t linphone_call_has_transfer_pending(const LinphoneCall *call){
return call->refer_pending;
}
/**
* Returns call's duration in seconds.
**/
int linphone_call_get_duration(const LinphoneCall *call){
if (call->media_start_time==0) return 0;
return time(NULL)-call->media_start_time;
}
/**
* Returns the call object this call is replacing, if any.
* Call replacement can occur during call transfers.
* By default, the core automatically terminates the replaced call and accept the new one.
* This function allows the application to know whether a new incoming call is a one that replaces another one.
**/
LinphoneCall *linphone_call_get_replaced_call(LinphoneCall *call){
SalOp *op=sal_call_get_replaces(call->op);
if (op){
return (LinphoneCall*)sal_op_get_user_pointer(op);
}
return NULL;
}
/**
* Indicate whether camera input should be sent to remote end.
**/
void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL && call->videostream->ticker!=NULL){
LinphoneCore *lc=call->core;
MSWebCam *nowebcam=get_nowebcam_device();
if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
video_stream_change_camera(call->videostream,
enable ? lc->video_conf.device : nowebcam);
}
}
call->camera_active=enable;
#endif
}
/**
* Take a photo of currently received video and write it into a jpeg file.
**/
int linphone_call_take_video_snapshot(LinphoneCall *call, const char *file){
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL && call->videostream->jpegwriter!=NULL){
return ms_filter_call_method(call->videostream->jpegwriter,MS_JPEG_WRITER_TAKE_SNAPSHOT,(void*)file);
}
ms_warning("Cannot take snapshot: no currently running video stream on this call.");
return -1;
#endif
return -1;
}
/**
* Returns TRUE if camera pictures are sent to the remote party.
**/
bool_t linphone_call_camera_enabled (const LinphoneCall *call){
return call->camera_active;
}
/**
* Enable video stream.
**/
void linphone_call_params_enable_video(LinphoneCallParams *cp, bool_t enabled){
cp->has_video=enabled;
}
const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
return cp->audio_codec;
}
const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
return cp->video_codec;
}
/**
* Returns whether video is enabled.
**/
bool_t linphone_call_params_video_enabled(const LinphoneCallParams *cp){
return cp->has_video;
}
enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
return cp->media_encryption;
}
void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
cp->media_encryption = e;
}
/**
* Enable sending of real early media (during outgoing calls).
**/
void linphone_call_params_enable_early_media_sending(LinphoneCallParams *cp, bool_t enabled){
cp->real_early_media=enabled;
}
bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
return cp->real_early_media;
}
/**
* Returns true if the call is part of the locally managed conference.
**/
bool_t linphone_call_params_local_conference_mode(const LinphoneCallParams *cp){
return cp->in_conference;
}
/**
* Refine bandwidth settings for this call by setting a bandwidth limit for audio streams.
* As a consequence, codecs whose bitrates are not compatible with this limit won't be used.
**/
void linphone_call_params_set_audio_bandwidth_limit(LinphoneCallParams *cp, int bandwidth){
cp->audio_bw=bandwidth;
}
#ifdef VIDEO_ENABLED
/**
* Request remote side to send us a Video Fast Update.
**/
void linphone_call_send_vfu_request(LinphoneCall *call)
{
if (LinphoneCallStreamsRunning == linphone_call_get_state(call))
sal_call_send_vfu_request(call->op);
}
#endif
/**
*
**/
LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
memcpy(ncp,cp,sizeof(LinphoneCallParams));
return ncp;
}
/**
*
**/
void linphone_call_params_destroy(LinphoneCallParams *p){
ms_free(p);
}
/**
* @}
**/
#ifdef TEST_EXT_RENDERER
static void rendercb(void *data, const MSPicture *local, const MSPicture *remote){
ms_message("rendercb, local buffer=%p, remote buffer=%p",
local ? local->planes[0] : NULL, remote? remote->planes[0] : NULL);
}
#endif
#ifdef VIDEO_ENABLED
static void video_stream_event_cb(void *user_pointer, const MSFilter *f, const unsigned int event_id, const void *args){
LinphoneCall* call = (LinphoneCall*) user_pointer;
ms_warning("In linphonecall.c: video_stream_event_cb");
switch (event_id) {
case MS_VIDEO_DECODER_DECODING_ERRORS:
ms_warning("Case is MS_VIDEO_DECODER_DECODING_ERRORS");
linphone_call_send_vfu_request(call);
break;
case MS_VIDEO_DECODER_FIRST_IMAGE_DECODED:
ms_message("First video frame decoded successfully");
if (call->nextVideoFrameDecoded._func != NULL)
call->nextVideoFrameDecoded._func(call, call->nextVideoFrameDecoded._user_data);
break;
default:
ms_warning("Unhandled event %i", event_id);
break;
}
}
#endif
void linphone_call_set_next_video_frame_decoded_callback(LinphoneCall *call, LinphoneCallCbFunc cb, void* user_data) {
call->nextVideoFrameDecoded._func = cb;
call->nextVideoFrameDecoded._user_data = user_data;
#ifdef VIDEO_ENABLED
ms_filter_call_method_noarg(call->videostream->decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
#endif
}
void linphone_call_init_audio_stream(LinphoneCall *call){
LinphoneCore *lc=call->core;
SalMediaDescription *md=call->localdesc;
AudioStream *audiostream;
call->audiostream=audiostream=audio_stream_new(md->streams[0].rtp_port,md->streams[0].rtcp_port,linphone_core_ipv6_enabled(lc));
if (linphone_core_echo_limiter_enabled(lc)){
const char *type=lp_config_get_string(lc->config,"sound","el_type","mic");
if (strcasecmp(type,"mic")==0)
audio_stream_enable_echo_limiter(audiostream,ELControlMic);
else if (strcasecmp(type,"full")==0)
audio_stream_enable_echo_limiter(audiostream,ELControlFull);
}
audio_stream_enable_gain_control(audiostream,TRUE);
if (linphone_core_echo_cancellation_enabled(lc)){
int len,delay,framesize;
const char *statestr=lp_config_get_string(lc->config,"sound","ec_state",NULL);
len=lp_config_get_int(lc->config,"sound","ec_tail_len",0);
delay=lp_config_get_int(lc->config,"sound","ec_delay",0);
framesize=lp_config_get_int(lc->config,"sound","ec_framesize",0);
audio_stream_set_echo_canceller_params(audiostream,len,delay,framesize);
if (statestr && audiostream->ec){
ms_filter_call_method(audiostream->ec,MS_ECHO_CANCELLER_SET_STATE_STRING,(void*)statestr);
}
}
audio_stream_enable_automatic_gain_control(audiostream,linphone_core_agc_enabled(lc));
{
int enabled=lp_config_get_int(lc->config,"sound","noisegate",0);
audio_stream_enable_noise_gate(audiostream,enabled);
}
audio_stream_set_features(audiostream,linphone_core_get_audio_features(lc));
if (lc->rtptf){
RtpTransport *artp=lc->rtptf->audio_rtp_func(lc->rtptf->audio_rtp_func_data, call->audio_port);
RtpTransport *artcp=lc->rtptf->audio_rtcp_func(lc->rtptf->audio_rtcp_func_data, call->audio_port+1);
rtp_session_set_transports(audiostream->session,artp,artcp);
}
if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
rtp_session_set_pktinfo(audiostream->session, TRUE);
rtp_session_set_symmetric_rtp(audiostream->session, FALSE);
audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
}
call->audiostream_app_evq = ortp_ev_queue_new();
rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
}
void linphone_call_init_video_stream(LinphoneCall *call){
#ifdef VIDEO_ENABLED
LinphoneCore *lc=call->core;
SalMediaDescription *md=call->localdesc;
if ((lc->video_conf.display || lc->video_conf.capture) && md->streams[1].rtp_port>0){
int video_recv_buf_size=lp_config_get_int(lc->config,"video","recv_buf_size",0);
call->videostream=video_stream_new(md->streams[1].rtp_port,md->streams[1].rtcp_port,linphone_core_ipv6_enabled(lc));
video_stream_enable_display_filter_auto_rotate(call->videostream, lp_config_get_int(lc->config,"video","display_filter_auto_rotate",0));
if (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->session,video_recv_buf_size);
if( lc->video_conf.displaytype != NULL)
video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
video_stream_set_event_callback(call->videostream,video_stream_event_cb, call);
if (lc->rtptf){
RtpTransport *vrtp=lc->rtptf->video_rtp_func(lc->rtptf->video_rtp_func_data, call->video_port);
RtpTransport *vrtcp=lc->rtptf->video_rtcp_func(lc->rtptf->video_rtcp_func_data, call->video_port+1);
rtp_session_set_transports(call->videostream->session,vrtp,vrtcp);
}
if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL) && (ice_session_check_list(call->ice_session, 1))){
rtp_session_set_pktinfo(call->videostream->session, TRUE);
rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
call->videostream->ice_check_list = ice_session_check_list(call->ice_session, 1);
ice_check_list_set_rtp_session(call->videostream->ice_check_list, call->videostream->session);
}
call->videostream_app_evq = ortp_ev_queue_new();
rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
#ifdef TEST_EXT_RENDERER
video_stream_set_render_callback(call->videostream,rendercb,NULL);
#endif
}
#else
call->videostream=NULL;
#endif
}
void linphone_call_init_media_streams(LinphoneCall *call){
linphone_call_init_audio_stream(call);
linphone_call_init_video_stream(call);
}
static int dtmf_tab[16]={'0','1','2','3','4','5','6','7','8','9','*','#','A','B','C','D'};
static void linphone_core_dtmf_received(RtpSession* s, int dtmf, void* user_data){
LinphoneCore* lc = (LinphoneCore*)user_data;
if (dtmf<0 || dtmf>15){
ms_warning("Bad dtmf value %i",dtmf);
return;
}
if (lc->vtable.dtmf_received != NULL)
lc->vtable.dtmf_received(lc, linphone_core_get_current_call(lc), dtmf_tab[dtmf]);
}
static void parametrize_equalizer(LinphoneCore *lc, AudioStream *st){
if (st->equalizer){
MSFilter *f=st->equalizer;
int enabled=lp_config_get_int(lc->config,"sound","eq_active",0);
const char *gains=lp_config_get_string(lc->config,"sound","eq_gains",NULL);
ms_filter_call_method(f,MS_EQUALIZER_SET_ACTIVE,&enabled);
if (enabled){
if (gains){
do{
int bytes;
MSEqualizerGain g;
if (sscanf(gains,"%f:%f:%f %n",&g.frequency,&g.gain,&g.width,&bytes)==3){
ms_message("Read equalizer gains: %f(~%f) --> %f",g.frequency,g.width,g.gain);
ms_filter_call_method(f,MS_EQUALIZER_SET_GAIN,&g);
gains+=bytes;
}else break;
}while(1);
}
}
}
}
void _post_configure_audio_stream(AudioStream *st, LinphoneCore *lc, bool_t muted){
float mic_gain=lp_config_get_float(lc->config,"sound","mic_gain",1);
float thres = 0;
float recv_gain;
float ng_thres=lp_config_get_float(lc->config,"sound","ng_thres",0.05);
float ng_floorgain=lp_config_get_float(lc->config,"sound","ng_floorgain",0);
int dc_removal=lp_config_get_int(lc->config,"sound","dc_removal",0);
if (!muted)
audio_stream_set_mic_gain(st,mic_gain);
else
audio_stream_set_mic_gain(st,0);
recv_gain = lc->sound_conf.soft_play_lev;
if (recv_gain != 0) {
linphone_core_set_playback_gain_db (lc,recv_gain);
}
if (st->volsend){
ms_filter_call_method(st->volsend,MS_VOLUME_REMOVE_DC,&dc_removal);
float speed=lp_config_get_float(lc->config,"sound","el_speed",-1);
thres=lp_config_get_float(lc->config,"sound","el_thres",-1);
float force=lp_config_get_float(lc->config,"sound","el_force",-1);
int sustain=lp_config_get_int(lc->config,"sound","el_sustain",-1);
float transmit_thres=lp_config_get_float(lc->config,"sound","el_transmit_thres",-1);
MSFilter *f=NULL;
f=st->volsend;
if (speed==-1) speed=0.03;
if (force==-1) force=25;
ms_filter_call_method(f,MS_VOLUME_SET_EA_SPEED,&speed);
ms_filter_call_method(f,MS_VOLUME_SET_EA_FORCE,&force);
if (thres!=-1)
ms_filter_call_method(f,MS_VOLUME_SET_EA_THRESHOLD,&thres);
if (sustain!=-1)
ms_filter_call_method(f,MS_VOLUME_SET_EA_SUSTAIN,&sustain);
if (transmit_thres!=-1)
ms_filter_call_method(f,MS_VOLUME_SET_EA_TRANSMIT_THRESHOLD,&transmit_thres);
ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
ms_filter_call_method(st->volsend,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&ng_floorgain);
}
if (st->volrecv){
/* parameters for a limited noise-gate effect, using echo limiter threshold */
float floorgain = 1/mic_gain;
int spk_agc=lp_config_get_int(lc->config,"sound","speaker_agc_enabled",0);
ms_filter_call_method(st->volrecv, MS_VOLUME_ENABLE_AGC, &spk_agc);
ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_THRESHOLD,&ng_thres);
ms_filter_call_method(st->volrecv,MS_VOLUME_SET_NOISE_GATE_FLOORGAIN,&floorgain);
}
parametrize_equalizer(lc,st);
}
static void post_configure_audio_streams(LinphoneCall*call){
AudioStream *st=call->audiostream;
LinphoneCore *lc=call->core;
_post_configure_audio_stream(st,lc,call->audio_muted);
if (lc->vtable.dtmf_received!=NULL){
/* replace by our default action*/
audio_stream_play_received_dtmfs(call->audiostream,FALSE);
rtp_session_signal_connect(call->audiostream->session,"telephone-event",(RtpCallback)linphone_core_dtmf_received,(unsigned long)lc);
}
}
static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
int bw;
const MSList *elem;
RtpProfile *prof=rtp_profile_new("Call profile");
bool_t first=TRUE;
int remote_bw=0;
LinphoneCore *lc=call->core;
int up_ptime=0;
*used_pt=-1;
for(elem=desc->payloads;elem!=NULL;elem=elem->next){
PayloadType *pt=(PayloadType*)elem->data;
int number;
if ((pt->flags & PAYLOAD_TYPE_FLAG_CAN_SEND) && first) {
if (desc->type==SalAudio){
linphone_core_update_allocated_audio_bandwidth_in_call(call,pt);
up_ptime=linphone_core_get_upload_ptime(lc);
}
*used_pt=payload_type_get_number(pt);
first=FALSE;
}
if (desc->bandwidth>0) remote_bw=desc->bandwidth;
else if (md->bandwidth>0) {
/*case where b=AS is given globally, not per stream*/
remote_bw=md->bandwidth;
if (desc->type==SalVideo){
remote_bw=get_video_bandwidth(remote_bw,call->audio_bw);
}
}
if (desc->type==SalAudio){
bw=get_min_bandwidth(call->audio_bw,remote_bw);
}else bw=get_min_bandwidth(get_video_bandwidth(linphone_core_get_upload_bandwidth (lc),call->audio_bw),remote_bw);
if (bw>0) pt->normal_bitrate=bw*1000;
else if (desc->type==SalAudio){
pt->normal_bitrate=-1;
}
if (desc->ptime>0){
up_ptime=desc->ptime;
}
if (up_ptime>0){
char tmp[40];
snprintf(tmp,sizeof(tmp),"ptime=%i",up_ptime);
payload_type_append_send_fmtp(pt,tmp);
}
number=payload_type_get_number(pt);
if (rtp_profile_get_payload(prof,number)!=NULL){
ms_warning("A payload type with number %i already exists in profile !",number);
}else
rtp_profile_set_payload(prof,number,pt);
}
return prof;
}
static void setup_ring_player(LinphoneCore *lc, LinphoneCall *call){
int pause_time=3000;
audio_stream_play(call->audiostream,lc->sound_conf.ringback_tone);
ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
}
#define LINPHONE_RTCP_SDES_TOOL "Linphone-" LINPHONE_VERSION
static bool_t linphone_call_sound_resources_available(LinphoneCall *call){
LinphoneCore *lc=call->core;
LinphoneCall *current=linphone_core_get_current_call(lc);
return !linphone_core_is_in_conference(lc) &&
(current==NULL || current==call);
}
static int find_crypto_index_from_tag(const SalSrtpCryptoAlgo crypto[],unsigned char tag) {
int i;
for(i=0; i<SAL_CRYPTO_ALGO_MAX; i++) {
if (crypto[i].tag == tag) {
return i;
}
}
return -1;
}
static void linphone_call_start_audio_stream(LinphoneCall *call, const char *cname, bool_t muted, bool_t send_ringbacktone, bool_t use_arc){
LinphoneCore *lc=call->core;
int jitt_comp=lc->rtp_conf.audio_jitt_comp;
int used_pt=-1;
/* look for savp stream first */
const SalStreamDescription *stream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpSavp,SalAudio);
/* no savp audio stream, use avp */
if (!stream)
stream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalAudio);
if (stream && stream->dir!=SalStreamInactive && stream->rtp_port!=0){
MSSndCard *playcard=lc->sound_conf.lsd_card ?
lc->sound_conf.lsd_card : lc->sound_conf.play_sndcard;
MSSndCard *captcard=lc->sound_conf.capt_sndcard;
const char *playfile=lc->play_file;
const char *recfile=lc->rec_file;
call->audio_profile=make_profile(call,call->resultdesc,stream,&used_pt);
bool_t use_ec;
if (used_pt!=-1){
call->current_params.audio_codec = rtp_profile_get_payload(call->audio_profile, used_pt);
if (playcard==NULL) {
ms_warning("No card defined for playback !");
}
if (captcard==NULL) {
ms_warning("No card defined for capture !");
}
/*Replace soundcard filters by inactive file players or recorders
when placed in recvonly or sendonly mode*/
if (stream->rtp_port==0 || stream->dir==SalStreamRecvOnly){
captcard=NULL;
playfile=NULL;
}else if (stream->dir==SalStreamSendOnly){
playcard=NULL;
captcard=NULL;
recfile=NULL;
/*And we will eventually play "playfile" if set by the user*/
/*playfile=NULL;*/
}
if (send_ringbacktone){
captcard=NULL;
playfile=NULL;/* it is setup later*/
}
/*if playfile are supplied don't use soundcards*/
if (lc->use_files) {
captcard=NULL;
playcard=NULL;
}
if (call->params.in_conference){
/* first create the graph without soundcard resources*/
captcard=playcard=NULL;
}
if (!linphone_call_sound_resources_available(call)){
ms_message("Sound resources are used by another call, not using soundcard.");
captcard=playcard=NULL;
}
use_ec=captcard==NULL ? FALSE : linphone_core_echo_cancellation_enabled(lc);
if (playcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(playcard, stream->max_rate);
if (captcard && stream->max_rate>0) ms_snd_card_set_preferred_sample_rate(captcard, stream->max_rate);
audio_stream_enable_adaptive_bitrate_control(call->audiostream,use_arc);
audio_stream_start_full(
call->audiostream,
call->audio_profile,
stream->rtp_addr[0]!='\0' ? stream->rtp_addr : call->resultdesc->addr,
stream->rtp_port,
stream->rtcp_addr[0]!='\0' ? stream->rtcp_addr : call->resultdesc->addr,
linphone_core_rtcp_enabled(lc) ? (stream->rtcp_port) : 0,
used_pt,
jitt_comp,
playfile,
recfile,
playcard,
captcard,
use_ec
);
post_configure_audio_streams(call);
if (muted && !send_ringbacktone){
audio_stream_set_mic_gain(call->audiostream,0);
}
if (stream->dir==SalStreamSendOnly && playfile!=NULL){
int pause_time=500;
ms_filter_call_method(call->audiostream->soundread,MS_FILE_PLAYER_LOOP,&pause_time);
}
if (send_ringbacktone){
setup_ring_player(lc,call);
}
audio_stream_set_rtcp_information(call->audiostream, cname, LINPHONE_RTCP_SDES_TOOL);
/* valid local tags are > 0 */
if (stream->proto == SalProtoRtpSavp) {
const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
SalProtoRtpSavp,SalAudio);
int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, stream->crypto_local_tag);
if (crypto_idx >= 0) {
audio_stream_enable_strp(
call->audiostream,
stream->crypto[0].algo,
local_st_desc->crypto[crypto_idx].master_key,
stream->crypto[0].master_key);
call->audiostream_encrypted=TRUE;
} else {
ms_warning("Failed to find local crypto algo with tag: %d", stream->crypto_local_tag);
call->audiostream_encrypted=FALSE;
}
}else call->audiostream_encrypted=FALSE;
if (call->params.in_conference){
/*transform the graph to connect it to the conference filter */
bool_t mute=stream->dir==SalStreamRecvOnly;
linphone_call_add_to_conf(call, mute);
}
call->current_params.in_conference=call->params.in_conference;
}else ms_warning("No audio stream accepted ?");
}
}
static void linphone_call_start_video_stream(LinphoneCall *call, const char *cname,bool_t all_inputs_muted){
#ifdef VIDEO_ENABLED
LinphoneCore *lc=call->core;
int used_pt=-1;
/* look for savp stream first */
const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpSavp,SalVideo);
/* no savp audio stream, use avp */
if (!vstream)
vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalVideo);
/* shutdown preview */
if (lc->previewstream!=NULL) {
video_preview_stop(lc->previewstream);
lc->previewstream=NULL;
}
if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->rtp_port!=0) {
const char *rtp_addr=vstream->rtp_addr[0]!='\0' ? vstream->rtp_addr : call->resultdesc->addr;
const char *rtcp_addr=vstream->rtcp_addr[0]!='\0' ? vstream->rtcp_addr : call->resultdesc->addr;
call->video_profile=make_profile(call,call->resultdesc,vstream,&used_pt);
if (used_pt!=-1){
call->current_params.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
VideoStreamDir dir=VideoStreamSendRecv;
MSWebCam *cam=lc->video_conf.device;
bool_t is_inactive=FALSE;
call->current_params.has_video=TRUE;
video_stream_enable_adaptive_bitrate_control(call->videostream,
linphone_core_adaptive_rate_control_enabled(lc));
video_stream_set_sent_video_size(call->videostream,linphone_core_get_preferred_video_size(lc));
video_stream_enable_self_view(call->videostream,lc->video_conf.selfview);
if (lc->video_window_id!=0)
video_stream_set_native_window_id(call->videostream,lc->video_window_id);
if (lc->preview_window_id!=0)
video_stream_set_native_preview_window_id (call->videostream,lc->preview_window_id);
video_stream_use_preview_video_window (call->videostream,lc->use_preview_window);
if (vstream->dir==SalStreamSendOnly && lc->video_conf.capture ){
cam=get_nowebcam_device();
dir=VideoStreamSendOnly;
}else if (vstream->dir==SalStreamRecvOnly && lc->video_conf.display ){
dir=VideoStreamRecvOnly;
}else if (vstream->dir==SalStreamSendRecv){
if (lc->video_conf.display && lc->video_conf.capture)
dir=VideoStreamSendRecv;
else if (lc->video_conf.display)
dir=VideoStreamRecvOnly;
else
dir=VideoStreamSendOnly;
}else{
ms_warning("video stream is inactive.");
/*either inactive or incompatible with local capabilities*/
is_inactive=TRUE;
}
if (call->camera_active==FALSE || all_inputs_muted){
cam=get_nowebcam_device();
}
if (!is_inactive){
call->log->video_enabled = TRUE;
video_stream_set_direction (call->videostream, dir);
ms_message("%s lc rotation:%d\n", __FUNCTION__, lc->device_rotation);
video_stream_set_device_rotation(call->videostream, lc->device_rotation);
video_stream_start(call->videostream,
call->video_profile, rtp_addr, vstream->rtp_port,
rtcp_addr, linphone_core_rtcp_enabled(lc) ? (vstream->rtcp_port) : 0,
used_pt, lc->rtp_conf.audio_jitt_comp, cam);
video_stream_set_rtcp_information(call->videostream, cname,LINPHONE_RTCP_SDES_TOOL);
}
if (vstream->proto == SalProtoRtpSavp) {
const SalStreamDescription *local_st_desc=sal_media_description_find_stream(call->localdesc,
SalProtoRtpSavp,SalVideo);
video_stream_enable_strp(
call->videostream,
vstream->crypto[0].algo,
local_st_desc->crypto[0].master_key,
vstream->crypto[0].master_key
);
call->videostream_encrypted=TRUE;
}else{
call->videostream_encrypted=FALSE;
}
}else ms_warning("No video stream accepted.");
}else{
ms_warning("No valid video stream defined.");
}
#endif
}
void linphone_call_start_media_streams(LinphoneCall *call, bool_t all_inputs_muted, bool_t send_ringbacktone){
LinphoneCore *lc=call->core;
call->current_params.audio_codec = NULL;
call->current_params.video_codec = NULL;
LinphoneAddress *me=linphone_core_get_primary_contact_parsed(lc);
char *cname;
bool_t use_arc=linphone_core_adaptive_rate_control_enabled(lc);
#ifdef VIDEO_ENABLED
const SalStreamDescription *vstream=sal_media_description_find_stream(call->resultdesc,
SalProtoRtpAvp,SalVideo);
#endif
if(call->audiostream == NULL)
{
ms_fatal("start_media_stream() called without prior init !");
return;
}
cname=linphone_address_as_string_uri_only(me);
#if defined(VIDEO_ENABLED)
if (vstream!=NULL && vstream->dir!=SalStreamInactive && vstream->payloads!=NULL){
/*when video is used, do not make adaptive rate control on audio, it is stupid.*/
use_arc=FALSE;
}
#endif
linphone_call_start_audio_stream(call,cname,all_inputs_muted,send_ringbacktone,use_arc);
call->current_params.has_video=FALSE;
if (call->videostream!=NULL) {
linphone_call_start_video_stream(call,cname,all_inputs_muted);
}
call->all_muted=all_inputs_muted;
call->playing_ringbacktone=send_ringbacktone;
call->up_bw=linphone_core_get_upload_bandwidth(lc);
if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
OrtpZrtpParams params;
/*will be set later when zrtp is activated*/
call->current_params.media_encryption=LinphoneMediaEncryptionNone;
params.zid_file=lc->zrtp_secrets_cache;
audio_stream_enable_zrtp(call->audiostream,&params);
}else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
}
/*also reflect the change if the "wished" params, in order to avoid to propose SAVP or video again
* further in the call, for example during pause,resume, conferencing reINVITEs*/
linphone_call_fix_call_parameters(call);
if ((call->ice_session != NULL) && (ice_session_state(call->ice_session) != IS_Completed)) {
ice_session_start_connectivity_checks(call->ice_session);
}
goto end;
end:
ms_free(cname);
linphone_address_destroy(me);
}
void linphone_call_start_media_streams_for_ice_gathering(LinphoneCall *call){
audio_stream_prepare_sound(call->audiostream, NULL, NULL);
#ifdef VIDEO_ENABLED
if (call->videostream) {
video_stream_prepare_video(call->videostream);
}
#endif
}
void linphone_call_delete_ice_session(LinphoneCall *call){
if (call->ice_session != NULL) {
ice_session_destroy(call->ice_session);
call->ice_session = NULL;
if (call->audiostream != NULL) call->audiostream->ice_check_list = NULL;
if (call->videostream != NULL) call->videostream->ice_check_list = NULL;
}
}
static void linphone_call_log_fill_stats(LinphoneCallLog *log, AudioStream *st){
audio_stream_get_local_rtp_stats (st,&log->local_stats);
log->quality=audio_stream_get_average_quality_rating(st);
}
void linphone_call_stop_media_streams(LinphoneCall *call){
if (call->audiostream!=NULL) {
call->audiostream->ice_check_list = NULL;
rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
ortp_ev_queue_flush(call->audiostream_app_evq);
ortp_ev_queue_destroy(call->audiostream_app_evq);
call->audiostream_app_evq=NULL;
if (call->audiostream->ec){
const char *state_str=NULL;
ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_STATE_STRING,&state_str);
if (state_str){
ms_message("Writing echo canceler state, %i bytes",(int)strlen(state_str));
lp_config_set_string(call->core->config,"sound","ec_state",state_str);
}
}
linphone_call_log_fill_stats (call->log,call->audiostream);
if (call->endpoint){
linphone_call_remove_from_conf(call);
}
audio_stream_stop(call->audiostream);
call->audiostream=NULL;
}
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL){
call->videostream->ice_check_list = NULL;
rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
ortp_ev_queue_flush(call->videostream_app_evq);
ortp_ev_queue_destroy(call->videostream_app_evq);
call->videostream_app_evq=NULL;
video_stream_stop(call->videostream);
call->videostream=NULL;
}
#endif
ms_event_queue_skip(call->core->msevq);
if (call->audio_profile){
rtp_profile_clear_all(call->audio_profile);
rtp_profile_destroy(call->audio_profile);
call->audio_profile=NULL;
}
if (call->video_profile){
rtp_profile_clear_all(call->video_profile);
rtp_profile_destroy(call->video_profile);
call->video_profile=NULL;
}
}
void linphone_call_enable_echo_cancellation(LinphoneCall *call, bool_t enable) {
if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
bool_t bypass_mode = !enable;
ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_SET_BYPASS_MODE,&bypass_mode);
}
}
bool_t linphone_call_echo_cancellation_enabled(LinphoneCall *call) {
if (call!=NULL && call->audiostream!=NULL && call->audiostream->ec){
bool_t val;
ms_filter_call_method(call->audiostream->ec,MS_ECHO_CANCELLER_GET_BYPASS_MODE,&val);
return !val;
} else {
return linphone_core_echo_cancellation_enabled(call->core);
}
}
void linphone_call_enable_echo_limiter(LinphoneCall *call, bool_t val){
if (call!=NULL && call->audiostream!=NULL ) {
if (val) {
const char *type=lp_config_get_string(call->core->config,"sound","el_type","mic");
if (strcasecmp(type,"mic")==0)
audio_stream_enable_echo_limiter(call->audiostream,ELControlMic);
else if (strcasecmp(type,"full")==0)
audio_stream_enable_echo_limiter(call->audiostream,ELControlFull);
} else {
audio_stream_enable_echo_limiter(call->audiostream,ELInactive);
}
}
}
bool_t linphone_call_echo_limiter_enabled(const LinphoneCall *call){
if (call!=NULL && call->audiostream!=NULL ){
return call->audiostream->el_type !=ELInactive ;
} else {
return linphone_core_echo_limiter_enabled(call->core);
}
}
/**
* @addtogroup call_misc
* @{
**/
/**
* Returns the measured sound volume played locally (received from remote).
* It is expressed in dbm0.
**/
float linphone_call_get_play_volume(LinphoneCall *call){
AudioStream *st=call->audiostream;
if (st && st->volrecv){
float vol=0;
ms_filter_call_method(st->volrecv,MS_VOLUME_GET,&vol);
return vol;
}
return LINPHONE_VOLUME_DB_LOWEST;
}
/**
* Returns the measured sound volume recorded locally (sent to remote).
* It is expressed in dbm0.
**/
float linphone_call_get_record_volume(LinphoneCall *call){
AudioStream *st=call->audiostream;
if (st && st->volsend && !call->audio_muted && call->state==LinphoneCallStreamsRunning){
float vol=0;
ms_filter_call_method(st->volsend,MS_VOLUME_GET,&vol);
return vol;
}
return LINPHONE_VOLUME_DB_LOWEST;
}
/**
* Obtain real-time quality rating of the call
*
* Based on local RTP statistics and RTCP feedback, a quality rating is computed and updated
* during all the duration of the call. This function returns its value at the time of the function call.
* It is expected that the rating is updated at least every 5 seconds or so.
* The rating is a floating point number comprised between 0 and 5.
*
* 4-5 = good quality <br>
* 3-4 = average quality <br>
* 2-3 = poor quality <br>
* 1-2 = very poor quality <br>
* 0-1 = can't be worse, mostly unusable <br>
*
* @returns The function returns -1 if no quality measurement is available, for example if no
* active audio stream exist. Otherwise it returns the quality rating.
**/
float linphone_call_get_current_quality(LinphoneCall *call){
if (call->audiostream){
return audio_stream_get_quality_rating(call->audiostream);
}
return -1;
}
/**
* Returns call quality averaged over all the duration of the call.
*
* See linphone_call_get_current_quality() for more details about quality measurement.
**/
float linphone_call_get_average_quality(LinphoneCall *call){
if (call->audiostream){
return audio_stream_get_average_quality_rating(call->audiostream);
}
return -1;
}
/**
* Access last known statistics for audio stream, for a given call.
**/
const LinphoneCallStats *linphone_call_get_audio_stats(const LinphoneCall *call) {
return &call->stats[LINPHONE_CALL_STATS_AUDIO];
}
/**
* Access last known statistics for video stream, for a given call.
**/
const LinphoneCallStats *linphone_call_get_video_stats(const LinphoneCall *call) {
return &call->stats[LINPHONE_CALL_STATS_VIDEO];
}
/**
* @}
**/
static void display_bandwidth(RtpSession *as, RtpSession *vs){
ms_message("bandwidth usage: audio=[d=%.1f,u=%.1f] video=[d=%.1f,u=%.1f] kbit/sec",
(as!=NULL) ? (rtp_session_compute_recv_bandwidth(as)*1e-3) : 0,
(as!=NULL) ? (rtp_session_compute_send_bandwidth(as)*1e-3) : 0,
(vs!=NULL) ? (rtp_session_compute_recv_bandwidth(vs)*1e-3) : 0,
(vs!=NULL) ? (rtp_session_compute_send_bandwidth(vs)*1e-3) : 0);
}
static void linphone_core_disconnected(LinphoneCore *lc, LinphoneCall *call){
char temp[256];
char *from=NULL;
if(call)
from = linphone_call_get_remote_address_as_string(call);
if (from)
{
snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
free(from);
}
else
{
snprintf(temp,sizeof(temp),"Remote end seems to have disconnected, the call is going to be closed.");
}
if (lc->vtable.display_warning!=NULL)
lc->vtable.display_warning(lc,temp);
linphone_core_terminate_call(lc,call);
}
static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
OrtpEventType evt=ortp_event_get_type(ev);
OrtpEventData *evd=ortp_event_get_data(ev);
if (evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) {
switch (ice_session_state(call->ice_session)) {
case IS_Completed:
if (ice_session_role(call->ice_session) == IR_Controlling) {
ice_session_select_candidates(call->ice_session);
linphone_core_update_call(call->core, call, &call->current_params);
}
break;
case IS_Failed:
linphone_call_delete_ice_session(call);
break;
default:
break;
}
} else if (evt == ORTP_EVENT_ICE_GATHERING_FINISHED) {
if (evd->info.ice_processing_successful==TRUE) {
ice_session_compute_candidates_foundations(call->ice_session);
ice_session_eliminate_redundant_candidates(call->ice_session);
ice_session_choose_default_candidates(call->ice_session);
} else {
linphone_call_delete_ice_session(call);
}
switch (call->state) {
case LinphoneCallStreamsRunning:
linphone_core_start_update_call(call->core, call);
break;
case LinphoneCallUpdatedByRemote:
linphone_core_start_accept_call_update(call->core, call);
break;
case LinphoneCallOutgoingInit:
linphone_call_stop_media_streams(call);
linphone_core_proceed_with_invite_if_ready(call->core, call, NULL);
break;
default:
linphone_call_stop_media_streams(call);
linphone_core_notify_incoming_call(call->core, call);
break;
}
} else if (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) {
linphone_core_start_accept_call_update(call->core, call);
} else if (evt == ORTP_EVENT_ICE_RESTART_NEEDED) {
ice_session_restart(call->ice_session);
ice_session_set_role(call->ice_session, IR_Controlling);
linphone_core_update_call(call->core, call, &call->current_params);
}
}
void linphone_call_background_tasks(LinphoneCall *call, bool_t one_second_elapsed){
LinphoneCore* lc = call->core;
int disconnect_timeout = linphone_core_get_nortp_timeout(call->core);
bool_t disconnected=FALSE;
if (call->state==LinphoneCallStreamsRunning && one_second_elapsed){
RtpSession *as=NULL,*vs=NULL;
float audio_load=0, video_load=0;
if (call->audiostream!=NULL){
as=call->audiostream->session;
if (call->audiostream->ticker)
audio_load=ms_ticker_get_average_load(call->audiostream->ticker);
}
if (call->videostream!=NULL){
if (call->videostream->ticker)
video_load=ms_ticker_get_average_load(call->videostream->ticker);
vs=call->videostream->session;
}
display_bandwidth(as,vs);
ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
}
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL) {
OrtpEvent *ev;
/* Ensure there is no dangling ICE check list. */
if (call->ice_session == NULL) call->videostream->ice_check_list = NULL;
// Beware that the application queue should not depend on treatments fron the
// mediastreamer queue.
video_stream_iterate(call->videostream);
while (call->videostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->videostream_app_evq)))){
OrtpEventType evt=ortp_event_get_type(ev);
OrtpEventData *evd=ortp_event_get_data(ev);
if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
linphone_call_videostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->session);
if(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp != NULL)
freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp);
call->stats[LINPHONE_CALL_STATS_VIDEO].received_rtcp = evd->packet;
evd->packet = NULL;
if (lc->vtable.call_stats_updated)
lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->session), sizeof(jitter_stats_t));
if(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp != NULL)
freemsg(call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp);
call->stats[LINPHONE_CALL_STATS_VIDEO].sent_rtcp = evd->packet;
evd->packet = NULL;
if (lc->vtable.call_stats_updated)
lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_VIDEO]);
} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
|| (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
handle_ice_events(call, ev);
}
ortp_event_destroy(ev);
}
}
#endif
if (call->audiostream!=NULL) {
OrtpEvent *ev;
/* Ensure there is no dangling ICE check list. */
if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
// Beware that the application queue should not depend on treatments fron the
// mediastreamer queue.
audio_stream_iterate(call->audiostream);
while (call->audiostream_app_evq && (NULL != (ev=ortp_ev_queue_get(call->audiostream_app_evq)))){
OrtpEventType evt=ortp_event_get_type(ev);
OrtpEventData *evd=ortp_event_get_data(ev);
if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
} else if (evt == ORTP_EVENT_ZRTP_SAS_READY) {
linphone_call_audiostream_auth_token_ready(call, evd->info.zrtp_sas.sas, evd->info.zrtp_sas.verified);
} else if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->session);
if(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp != NULL)
freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp);
call->stats[LINPHONE_CALL_STATS_AUDIO].received_rtcp = evd->packet;
evd->packet = NULL;
if (lc->vtable.call_stats_updated)
lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
} else if (evt == ORTP_EVENT_RTCP_PACKET_EMITTED) {
memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->session), sizeof(jitter_stats_t));
if(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp != NULL)
freemsg(call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp);
call->stats[LINPHONE_CALL_STATS_AUDIO].sent_rtcp = evd->packet;
evd->packet = NULL;
if (lc->vtable.call_stats_updated)
lc->vtable.call_stats_updated(lc, call, &call->stats[LINPHONE_CALL_STATS_AUDIO]);
} else if ((evt == ORTP_EVENT_ICE_SESSION_PROCESSING_FINISHED) || (evt == ORTP_EVENT_ICE_GATHERING_FINISHED)
|| (evt == ORTP_EVENT_ICE_LOSING_PAIRS_COMPLETED) || (evt == ORTP_EVENT_ICE_RESTART_NEEDED)) {
handle_ice_events(call, ev);
}
ortp_event_destroy(ev);
}
}
if (call->state==LinphoneCallStreamsRunning && one_second_elapsed && call->audiostream!=NULL && disconnect_timeout>0 )
disconnected=!audio_stream_alive(call->audiostream,disconnect_timeout);
if (disconnected)
linphone_core_disconnected(call->core,call);
}
void linphone_call_log_completed(LinphoneCall *call){
LinphoneCore *lc=call->core;
call->log->duration=time(NULL)-call->start_time;
if (call->log->status==LinphoneCallMissed){
char *info;
lc->missed_calls++;
info=ortp_strdup_printf(ngettext("You have missed %i call.",
"You have missed %i calls.", lc->missed_calls),
lc->missed_calls);
if (lc->vtable.display_status!=NULL)
lc->vtable.display_status(lc,info);
ms_free(info);
}
lc->call_logs=ms_list_prepend(lc->call_logs,(void *)call->log);
if (ms_list_size(lc->call_logs)>lc->max_call_logs){
MSList *elem,*prevelem=NULL;
/*find the last element*/
for(elem=lc->call_logs;elem!=NULL;elem=elem->next){
prevelem=elem;
}
elem=prevelem;
linphone_call_log_destroy((LinphoneCallLog*)elem->data);
lc->call_logs=ms_list_remove_link(lc->call_logs,elem);
}
if (lc->vtable.call_log_updated!=NULL){
lc->vtable.call_log_updated(lc,call->log);
}
call_logs_write_to_config_file(lc);
}
LinphoneCallState linphone_call_get_transfer_state(LinphoneCall *call) {
return call->transfer_state;
}
void linphone_call_set_transfer_state(LinphoneCall* call, LinphoneCallState state) {
if (state != call->transfer_state) {
LinphoneCore* lc = call->core;
call->transfer_state = state;
if (lc->vtable.transfer_state_changed)
lc->vtable.transfer_state_changed(lc, call, state);
}
}
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