if (call->audiostream==NULL){
ms_error("linphone_call_set_authentication_token_verified(): No audio stream");
}
- if (call->audiostream->ortpZrtpContext==NULL){
+ if (call->audiostream->ms.zrtp_context==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);
+ ortp_zrtp_sas_verified(call->audiostream->ms.zrtp_context);
}else if (call->auth_token_verified && !verified){
- ortp_zrtp_sas_reset_verified(call->audiostream->ortpZrtpContext);
+ ortp_zrtp_sas_reset_verified(call->audiostream->ms.zrtp_context);
}
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){
+static MSList *make_codec_list(LinphoneCore *lc, const MSList *codecs, int bandwidth_limit,int* max_sample_rate, int nb_codecs_limit){
MSList *l=NULL;
const MSList *it;
+ int nb = 0;
if (max_sample_rate) *max_sample_rate=0;
for(it=codecs;it!=NULL;it=it->next){
PayloadType *pt=(PayloadType*)it->data;
}
if (linphone_core_check_payload_type_usability(lc,pt)){
l=ms_list_append(l,payload_type_clone(pt));
+ nb++;
if (max_sample_rate && payload_type_get_rate(pt)>*max_sample_rate) *max_sample_rate=payload_type_get_rate(pt);
}
}
+ if ((nb_codecs_limit > 0) && (nb >= nb_codecs_limit)) break;
}
return l;
}
static void update_media_description_from_stun(SalMediaDescription *md, const StunCandidate *ac, const StunCandidate *vc){
- if (ac->port!=0){
- strcpy(md->streams[0].rtp_addr,ac->addr);
- md->streams[0].rtp_port=ac->port;
- if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->nstreams==1){
- strcpy(md->addr,ac->addr);
+ int i;
+ for (i = 0; i < md->n_active_streams; i++) {
+ if ((md->streams[i].type == SalAudio) && (ac->port != 0)) {
+ strcpy(md->streams[0].rtp_addr,ac->addr);
+ md->streams[0].rtp_port=ac->port;
+ if ((ac->addr[0]!='\0' && vc->addr[0]!='\0' && strcmp(ac->addr,vc->addr)==0) || md->n_active_streams==1){
+ strcpy(md->addr,ac->addr);
+ }
+ }
+ if ((md->streams[i].type == SalVideo) && (vc->port != 0)) {
+ strcpy(md->streams[1].rtp_addr,vc->addr);
+ md->streams[1].rtp_port=vc->port;
}
}
- if (vc->port!=0){
- strcpy(md->streams[1].rtp_addr,vc->addr);
- md->streams[1].rtp_port=vc->port;
- }
-
}
void linphone_call_make_local_media_description(LinphoneCore *lc, LinphoneCall *call){
const char *username=linphone_address_get_username (addr);
SalMediaDescription *md=sal_media_description_new();
bool_t keep_srtp_keys=lp_config_get_int(lc->config,"sip","keep_srtp_keys",0);
-
- if (call->ping_time>0) {
- linphone_core_adapt_to_network(lc,call->ping_time,&call->params);
- }
+
+ linphone_core_adapt_to_network(lc,call->ping_time,&call->params);
md->session_id=(old_md ? old_md->session_id : (rand() & 0xfff));
md->session_ver=(old_md ? (old_md->session_ver+1) : (rand() & 0xfff));
- md->nstreams=1;
+ md->n_total_streams=(old_md ? old_md->n_total_streams : 1);
+ md->n_active_streams=1;
strncpy(md->addr,call->localip,sizeof(md->addr));
strncpy(md->username,username,sizeof(md->username));
md->streams[0].ptime=call->params.down_ptime;
else
md->streams[0].ptime=linphone_core_get_download_ptime(lc);
- 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=make_codec_list(lc,lc->codecs_conf.audio_codecs,call->params.audio_bw,&md->streams[0].max_rate,-1);
+ pt=payload_type_clone(rtp_profile_get_payload_from_mime(lc->default_profile,"telephone-event"));
l=ms_list_append(l,pt);
md->streams[0].payloads=l;
if (call->params.has_video){
- md->nstreams++;
+ md->n_active_streams++;
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);
+ l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL,-1);
md->streams[1].payloads=l;
}
-
- for(i=0; i<md->nstreams; i++) {
+ if (md->n_total_streams < md->n_active_streams)
+ md->n_total_streams = md->n_active_streams;
+
+ /* Deactivate inactive streams. */
+ for (i = md->n_active_streams; i < md->n_total_streams; i++) {
+ md->streams[i].rtp_port = 0;
+ md->streams[i].rtcp_port = 0;
+ md->streams[i].proto = SalProtoRtpAvp;
+ md->streams[i].type = old_md->streams[i].type;
+ md->streams[i].dir = SalStreamInactive;
+ l = make_codec_list(lc, lc->codecs_conf.video_codecs, 0, NULL, 1);
+ md->streams[i].payloads = l;
+ }
+
+ for(i=0; i<md->n_active_streams; i++) {
if (md->streams[i].proto == SalProtoRtpSavp) {
if (keep_srtp_keys && old_md && old_md->streams[i].proto==SalProtoRtpSavp){
int j;
linphone_core_update_local_media_description_from_ice(md, call->ice_session);
linphone_core_update_ice_state_in_call_stats(call);
}
+#ifdef BUILD_UPNP
+ if(call->upnp_session != NULL) {
+ linphone_core_update_local_media_description_from_upnp(md, call->upnp_session);
+ linphone_core_update_upnp_state_in_call_stats(call);
+ }
+#endif //BUILD_UPNP
linphone_address_destroy(addr);
call->localdesc=md;
if (old_md) sal_media_description_unref(old_md);
stats->received_rtcp = NULL;
stats->sent_rtcp = NULL;
stats->ice_state = LinphoneIceStateNotActivated;
+#ifdef BUILD_UPNP
+ stats->upnp_state = LinphoneUpnpStateIdle;
+#else
+ stats->upnp_state = LinphoneUpnpStateNotAvailable;
+#endif //BUILD_UPNP
}
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_core_get_local_ip(lc,NULL,call->localip);
linphone_call_init_common(call,from,to);
- call->params=*params;
+ _linphone_call_params_copy(&call->params,params);
+ sal_op_set_custom_header(call->op,call->params.custom_headers);
+ call->params.custom_headers=NULL;
+
if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) {
call->ice_session = ice_session_new();
ice_session_set_role(call->ice_session, IR_Controlling);
if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseStun) {
call->ping_time=linphone_core_run_stun_tests(call->core,call);
}
+#ifdef BUILD_UPNP
+ if (linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseUpnp) {
+ if(!lc->rtp_conf.disable_upnp) {
+ call->upnp_session = linphone_upnp_session_new(call);
+ }
+ }
+#endif //BUILD_UPNP
call->camera_active=params->has_video;
discover_mtu(lc,linphone_address_get_domain (to));
LinphoneCall * linphone_call_new_incoming(LinphoneCore *lc, LinphoneAddress *from, LinphoneAddress *to, SalOp *op){
LinphoneCall *call=ms_new0(LinphoneCall,1);
char *from_str;
+ const SalMediaDescription *md;
call->dir=LinphoneCallIncoming;
sal_op_set_user_pointer(op,call);
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);
+#ifdef BUILD_UPNP
+ if (lc->upnp != NULL && linphone_core_get_firewall_policy(lc)==LinphonePolicyUseUpnp &&
+ linphone_upnp_context_get_state(lc->upnp) == LinphoneUpnpStateOk) {
+#else //BUILD_UPNP
+ {
+#endif //BUILD_UPNP
+ /*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_core_get_local_ip(lc,NULL,call->localip);
linphone_call_init_common(call, from, to);
call->log->call_id=ms_strdup(sal_op_get_call_id(op)); /*must be known at that time*/
linphone_core_init_default_params(lc, &call->params);
+ md=sal_call_get_remote_media_description(op);
call->params.has_video &= !!lc->video_policy.automatically_accept;
- call->params.has_video &= linphone_core_media_description_contains_video_stream(sal_call_get_remote_media_description(op));
+ if (md) {
+ // It is licit to receive an INVITE without SDP
+ // In this case WE chose the media parameters according to policy.
+ call->params.has_video &= linphone_core_media_description_contains_video_stream(md);
+ }
switch (linphone_core_get_firewall_policy(call->core)) {
case LinphonePolicyUseIce:
call->ice_session = ice_session_new();
case LinphonePolicyUseStun:
call->ping_time=linphone_core_run_stun_tests(call->core,call);
/* No break to also destroy ice session in this case. */
+ break;
+ case LinphonePolicyUseUpnp:
+#ifdef BUILD_UPNP
+ if(!lc->rtp_conf.disable_upnp) {
+ call->upnp_session = linphone_upnp_session_new(call);
+ if (call->upnp_session != NULL) {
+ linphone_call_init_media_streams(call);
+ if (linphone_core_update_upnp_from_remote_media_description(call, sal_call_get_remote_media_description(op))<0) {
+ /* uPnP port mappings failed, proceed with the call anyway. */
+ linphone_call_delete_upnp_session(call);
+ }
+ }
+ }
+#endif //BUILD_UPNP
+ break;
default:
break;
}
static void linphone_call_destroy(LinphoneCall *obj)
{
+#ifdef BUILD_UPNP
+ linphone_call_delete_upnp_session(obj);
+#endif //BUILD_UPNP
linphone_call_delete_ice_session(obj);
if (obj->op!=NULL) {
sal_op_release(obj->op);
if (obj->auth_token) {
ms_free(obj->auth_token);
}
-
+ linphone_call_params_uninit(&obj->params);
ms_free(obj);
}
/**
* Returns current parameters associated to the call.
**/
-const LinphoneCallParams * linphone_call_get_current_params(const LinphoneCall *call){
+const LinphoneCallParams * linphone_call_get_current_params(LinphoneCall *call){
+ if (call->params.record_file)
+ call->current_params.record_file=call->params.record_file;
return &call->current_params;
}
}else if (vsd){
cp->has_video=is_video_active(vsd);
}
+ if (!cp->has_video){
+ if (md->bandwidth>0 && md->bandwidth<=linphone_core_get_edge_bw(call->core)){
+ cp->low_bandwidth=TRUE;
+ }
+ }
+ cp->custom_headers=(SalCustomHeader*)sal_op_get_custom_header(call->op);
return cp;
}
}
return NULL;
}
+/**
+ * Returns the far end's sip contact as a string, if available.
+**/
+const char *linphone_call_get_remote_contact(LinphoneCall *call){
+ if (call->op){
+ return sal_op_get_remote_contact(call->op);
+ }
+ return NULL;
+}
+
/**
* Returns true if this calls has received a transfer that has not been
* executed yet.
**/
void linphone_call_enable_camera (LinphoneCall *call, bool_t enable){
#ifdef VIDEO_ENABLED
- if (call->videostream!=NULL && call->videostream->ticker!=NULL){
+ if (call->videostream!=NULL && call->videostream->ms.ticker!=NULL){
LinphoneCore *lc=call->core;
MSWebCam *nowebcam=get_nowebcam_device();
if (call->camera_active!=enable && lc->video_conf.device!=nowebcam){
#endif
}
+#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
+
+
/**
* Take a photo of currently received video and write it into a jpeg file.
**/
cp->has_video=enabled;
}
+/**
+ * Returns the audio codec used in the call, described as a PayloadType structure.
+**/
const PayloadType* linphone_call_params_get_used_audio_codec(const LinphoneCallParams *cp) {
return cp->audio_codec;
}
+
+/**
+ * Returns the video codec used in the call, described as a PayloadType structure.
+**/
const PayloadType* linphone_call_params_get_used_video_codec(const LinphoneCallParams *cp) {
return cp->video_codec;
}
+/**
+ * @ingroup call_control
+ * Use to know if this call has been configured in low bandwidth mode.
+ * This mode can be automatically discovered thanks to a stun server when activate_edge_workarounds=1 in section [net] of configuration file.
+ * An application that would have reliable way to know network capacity may not use activate_edge_workarounds=1 but instead manually configure
+ * low bandwidth mode with linphone_call_params_enable_low_bandwidth().
+ * <br> When enabled, this param may transform a call request with video in audio only mode.
+ * @return TRUE if low bandwidth has been configured/detected
+ */
bool_t linphone_call_params_low_bandwidth_enabled(const LinphoneCallParams *cp) {
return cp->low_bandwidth;
}
+
+/**
+ * @ingroup call_control
+ * Indicate low bandwith mode.
+ * Configuring a call to low bandwidth mode will result in the core to activate several settings for the call in order to ensure that bitrate usage
+ * is lowered to the minimum possible. Typically, ptime (packetization time) will be increased, audio codec's output bitrate will be targetted to 20kbit/s provided
+ * that it is achievable by the codec selected after SDP handshake. Video is automatically disabled.
+ *
+**/
+void linphone_call_params_enable_low_bandwidth(LinphoneCallParams *cp, bool_t enabled){
+ cp->low_bandwidth=enabled;
+}
+
/**
* Returns whether video is enabled.
**/
return cp->has_video;
}
+/**
+ * Returns kind of media encryption selected for the call.
+**/
enum LinphoneMediaEncryption linphone_call_params_get_media_encryption(const LinphoneCallParams *cp) {
return cp->media_encryption;
}
+/**
+ * Set requested media encryption for a call.
+**/
void linphone_call_params_set_media_encryption(LinphoneCallParams *cp, enum LinphoneMediaEncryption e) {
cp->media_encryption = e;
}
cp->real_early_media=enabled;
}
+/**
+ * Indicates whether sending of early media was enabled.
+**/
bool_t linphone_call_params_early_media_sending_enabled(const LinphoneCallParams *cp){
return cp->real_early_media;
}
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);
+void linphone_call_params_add_custom_header(LinphoneCallParams *params, const char *header_name, const char *header_value){
+ params->custom_headers=sal_custom_header_append(params->custom_headers,header_name,header_value);
+}
+
+const char *linphone_call_params_get_custom_header(const LinphoneCallParams *params, const char *header_name){
+ return sal_custom_header_find(params->custom_headers,header_name);
+}
+
+void _linphone_call_params_copy(LinphoneCallParams *ncp, const LinphoneCallParams *cp){
+ memcpy(ncp,cp,sizeof(LinphoneCallParams));
+ if (cp->record_file) ncp->record_file=ms_strdup(cp->record_file);
+ /*
+ * The management of the custom headers is not optimal. We copy everything while ref counting would be more efficient.
+ */
+ if (cp->custom_headers) ncp->custom_headers=sal_custom_header_clone(cp->custom_headers);
}
-#endif
/**
- *
+ * Copy existing LinphoneCallParams to a new LinphoneCallParams object.
**/
LinphoneCallParams * linphone_call_params_copy(const LinphoneCallParams *cp){
LinphoneCallParams *ncp=ms_new0(LinphoneCallParams,1);
- memcpy(ncp,cp,sizeof(LinphoneCallParams));
+ _linphone_call_params_copy(ncp,cp);
return ncp;
}
+void linphone_call_params_uninit(LinphoneCallParams *p){
+ if (p->record_file) ms_free(p->record_file);
+ if (p->custom_headers) sal_custom_header_free(p->custom_headers);
+}
+
/**
- *
+ * Destroy LinphoneCallParams.
**/
void linphone_call_params_destroy(LinphoneCallParams *p){
+ linphone_call_params_uninit(p);
ms_free(p);
}
+
/**
* @}
**/
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);
+ ms_filter_call_method_noarg(call->videostream->ms.decoder, MS_VIDEO_DECODER_RESET_FIRST_IMAGE_NOTIFICATION);
#endif
}
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);
+ rtp_session_set_transports(audiostream->ms.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);
+ rtp_session_set_pktinfo(audiostream->ms.session, TRUE);
+ rtp_session_set_symmetric_rtp(audiostream->ms.session, FALSE);
if (ice_session_check_list(call->ice_session, 0) == NULL) {
ice_session_add_check_list(call->ice_session, ice_check_list_new());
}
- audiostream->ice_check_list = ice_session_check_list(call->ice_session, 0);
- ice_check_list_set_rtp_session(audiostream->ice_check_list, audiostream->session);
+ audiostream->ms.ice_check_list = ice_session_check_list(call->ice_session, 0);
+ ice_check_list_set_rtp_session(audiostream->ms.ice_check_list, audiostream->ms.session);
}
call->audiostream_app_evq = ortp_ev_queue_new();
- rtp_session_register_event_queue(audiostream->session,call->audiostream_app_evq);
+ rtp_session_register_event_queue(audiostream->ms.session,call->audiostream_app_evq);
}
void linphone_call_init_video_stream(LinphoneCall *call){
if (dscp!=-1)
video_stream_set_dscp(call->videostream,dscp);
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 (video_recv_buf_size>0) rtp_session_set_recv_buf_size(call->videostream->ms.session,video_recv_buf_size);
if( lc->video_conf.displaytype != NULL)
video_stream_set_display_filter_name(call->videostream,lc->video_conf.displaytype);
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);
+ rtp_session_set_transports(call->videostream->ms.session,vrtp,vrtcp);
}
if ((linphone_core_get_firewall_policy(lc) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
- rtp_session_set_pktinfo(call->videostream->session, TRUE);
- rtp_session_set_symmetric_rtp(call->videostream->session, FALSE);
+ rtp_session_set_pktinfo(call->videostream->ms.session, TRUE);
+ rtp_session_set_symmetric_rtp(call->videostream->ms.session, FALSE);
if (ice_session_check_list(call->ice_session, 1) == NULL) {
ice_session_add_check_list(call->ice_session, ice_check_list_new());
}
- 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->ms.ice_check_list = ice_session_check_list(call->ice_session, 1);
+ ice_check_list_set_rtp_session(call->videostream->ms.ice_check_list, call->videostream->ms.session);
}
call->videostream_app_evq = ortp_ev_queue_new();
- rtp_session_register_event_queue(call->videostream->session,call->videostream_app_evq);
+ rtp_session_register_event_queue(call->videostream->ms.session,call->videostream_app_evq);
#ifdef TEST_EXT_RENDERER
video_stream_set_render_callback(call->videostream,rendercb,NULL);
#endif
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);*/
}
+ if (call->record_active)
+ linphone_call_start_recording(call);
}
static RtpProfile *make_profile(LinphoneCall *call, const SalMediaDescription *md, const SalStreamDescription *desc, int *used_pt){
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_enable_adaptive_jittcomp(call->audiostream, linphone_core_audio_adaptive_jittcomp_enabled(lc));
+ if (!call->params.in_conference && call->params.record_file)
+ audio_stream_mixed_record_open(call->audiostream,call->params.record_file);
audio_stream_start_full(
call->audiostream,
call->audio_profile,
}
audio_stream_set_rtcp_information(call->audiostream, cname, rtcp_tool);
- /* valid local tags are > 0 */
+ /* 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;
- }
+ 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_srtp(
+ 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 */
#endif
}
+void linphone_call_update_crypto_parameters(LinphoneCall *call, SalMediaDescription *old_md, SalMediaDescription *new_md) {
+ SalStreamDescription *old_stream;
+ SalStreamDescription *new_stream;
+ int i;
+
+ old_stream = sal_media_description_find_stream(old_md, SalProtoRtpSavp, SalAudio);
+ new_stream = sal_media_description_find_stream(new_md, SalProtoRtpSavp, SalAudio);
+ if (old_stream && new_stream) {
+ const SalStreamDescription *local_st_desc = sal_media_description_find_stream(call->localdesc, SalProtoRtpSavp, SalAudio);
+ if (local_st_desc) {
+ int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, new_stream->crypto_local_tag);
+ if (crypto_idx >= 0) {
+ audio_stream_enable_srtp(call->audiostream, new_stream->crypto[0].algo, local_st_desc->crypto[crypto_idx].master_key, new_stream->crypto[0].master_key);
+ call->audiostream_encrypted = TRUE;
+ } else {
+ ms_warning("Failed to find local crypto algo with tag: %d", new_stream->crypto_local_tag);
+ call->audiostream_encrypted = FALSE;
+ }
+ for (i = 0; i < SAL_CRYPTO_ALGO_MAX; i++) {
+ old_stream->crypto[i].tag = new_stream->crypto[i].tag;
+ old_stream->crypto[i].algo = new_stream->crypto[i].algo;
+ strncpy(old_stream->crypto[i].master_key, new_stream->crypto[i].master_key, sizeof(old_stream->crypto[i].master_key) - 1);
+ }
+ }
+ }
+
+#ifdef VIDEO_ENABLED
+ old_stream = sal_media_description_find_stream(old_md, SalProtoRtpSavp, SalVideo);
+ new_stream = sal_media_description_find_stream(new_md, SalProtoRtpSavp, SalVideo);
+ if (old_stream && new_stream) {
+ const SalStreamDescription *local_st_desc = sal_media_description_find_stream(call->localdesc, SalProtoRtpSavp, SalVideo);
+ if (local_st_desc) {
+ int crypto_idx = find_crypto_index_from_tag(local_st_desc->crypto, new_stream->crypto_local_tag);
+ if (crypto_idx >= 0) {
+ video_stream_enable_strp(call->videostream, new_stream->crypto[0].algo, local_st_desc->crypto[crypto_idx].master_key, new_stream->crypto[0].master_key);
+ call->videostream_encrypted = TRUE;
+ } else {
+ ms_warning("Failed to find local crypto algo with tag: %d", new_stream->crypto_local_tag);
+ call->videostream_encrypted = FALSE;
+ }
+ for (i = 0; i < SAL_CRYPTO_ALGO_MAX; i++) {
+ old_stream->crypto[i].tag = new_stream->crypto[i].tag;
+ old_stream->crypto[i].algo = new_stream->crypto[i].algo;
+ strncpy(old_stream->crypto[i].master_key, new_stream->crypto[i].master_key, sizeof(old_stream->crypto[i].master_key) - 1);
+ }
+ }
+ }
+#endif
+}
+
+void linphone_call_update_remote_session_id_and_ver(LinphoneCall *call) {
+ SalMediaDescription *remote_desc = sal_call_get_remote_media_description(call->op);
+ if (remote_desc) {
+ call->remote_session_id = remote_desc->session_id;
+ call->remote_session_ver = remote_desc->session_ver;
+ }
+}
+
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;
+ if (call->audiostream != NULL) call->audiostream->ms.ice_check_list = NULL;
+ if (call->videostream != NULL) call->videostream->ms.ice_check_list = NULL;
call->stats[LINPHONE_CALL_STATS_AUDIO].ice_state = LinphoneIceStateNotActivated;
call->stats[LINPHONE_CALL_STATS_VIDEO].ice_state = LinphoneIceStateNotActivated;
}
}
-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);
+#ifdef BUILD_UPNP
+void linphone_call_delete_upnp_session(LinphoneCall *call){
+ if(call->upnp_session!=NULL) {
+ linphone_upnp_session_destroy(call->upnp_session);
+ call->upnp_session=NULL;
+ }
+}
+#endif //BUILD_UPNP
+
+static void linphone_call_log_fill_stats(LinphoneCallLog *log, MediaStream *st){
+ float quality=media_stream_get_average_quality_rating(st);
+ if (quality>=0){
+ if (log->quality!=-1){
+ log->quality*=quality/5.0;
+ }else log->quality=quality;
+ }
}
void linphone_call_stop_audio_stream(LinphoneCall *call) {
if (call->audiostream!=NULL) {
- rtp_session_unregister_event_queue(call->audiostream->session,call->audiostream_app_evq);
+ rtp_session_unregister_event_queue(call->audiostream->ms.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;
lp_config_set_string(call->core->config,"sound","ec_state",state_str);
}
}
- linphone_call_log_fill_stats (call->log,call->audiostream);
+ audio_stream_get_local_rtp_stats(call->audiostream,&call->log->local_stats);
+ linphone_call_log_fill_stats (call->log,(MediaStream*)call->audiostream);
if (call->endpoint){
linphone_call_remove_from_conf(call);
}
void linphone_call_stop_video_stream(LinphoneCall *call) {
#ifdef VIDEO_ENABLED
if (call->videostream!=NULL){
- rtp_session_unregister_event_queue(call->videostream->session,call->videostream_app_evq);
+ rtp_session_unregister_event_queue(call->videostream->ms.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;
+ linphone_call_log_fill_stats(call->log,(MediaStream*)call->videostream);
video_stream_stop(call->videostream);
call->videostream=NULL;
}
* active audio stream exist. Otherwise it returns the quality rating.
**/
float linphone_call_get_current_quality(LinphoneCall *call){
+ float audio_rating=-1;
+ float video_rating=-1;
+ float result;
if (call->audiostream){
- return audio_stream_get_quality_rating(call->audiostream);
+ audio_rating=media_stream_get_quality_rating((MediaStream*)call->audiostream)/5.0;
}
- return -1;
+ if (call->videostream){
+ video_rating=media_stream_get_quality_rating((MediaStream*)call->videostream)/5.0;
+ }
+ if (audio_rating<0 && video_rating<0) result=-1;
+ else if (audio_rating<0) result=video_rating*5.0;
+ else if (video_rating<0) result=audio_rating*5.0;
+ else result=audio_rating*video_rating*5.0;
+ return result;
}
/**
return -1;
}
+static void update_local_stats(LinphoneCallStats *stats, MediaStream *stream){
+ const MSQualityIndicator *qi=media_stream_get_quality_indicator(stream);
+ if (qi) {
+ stats->local_late_rate=ms_quality_indicator_get_local_late_rate(qi);
+ stats->local_loss_rate=ms_quality_indicator_get_local_loss_rate(qi);
+ }
+}
+
/**
* 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];
+const LinphoneCallStats *linphone_call_get_audio_stats(LinphoneCall *call) {
+ LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_AUDIO];
+ if (call->audiostream){
+ update_local_stats(stats,(MediaStream*)call->audiostream);
+ }
+ return stats;
}
/**
* 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];
+const LinphoneCallStats *linphone_call_get_video_stats(LinphoneCall *call) {
+ LinphoneCallStats *stats=&call->stats[LINPHONE_CALL_STATS_VIDEO];
+ if (call->videostream){
+ update_local_stats(stats,(MediaStream*)call->videostream);
+ }
+ return stats;
}
+/**
+ * Enable recording of the call (voice-only).
+ * This function must be used before the call parameters are assigned to the call.
+ * The call recording can be started and paused after the call is established with
+ * linphone_call_start_recording() and linphone_call_pause_recording().
+ * @param cp the call parameters
+ * @param path path and filename of the file where audio is written.
+**/
+void linphone_call_params_set_record_file(LinphoneCallParams *cp, const char *path){
+ if (cp->record_file){
+ ms_free(cp->record_file);
+ cp->record_file=NULL;
+ }
+ if (path) cp->record_file=ms_strdup(path);
+}
+
+/**
+ * Retrieves the path for the audio recoding of the call.
+**/
+const char *linphone_call_params_get_record_file(const LinphoneCallParams *cp){
+ return cp->record_file;
+}
+
+/**
+ * Start call recording.
+ * The output file where audio is recorded must be previously specified with linphone_call_params_set_record_file().
+**/
+void linphone_call_start_recording(LinphoneCall *call){
+ if (!call->params.record_file){
+ ms_error("linphone_call_start_recording(): no output file specified. Use linphone_call_params_set_record_file().");
+ return;
+ }
+ if (call->audiostream && !call->params.in_conference){
+ audio_stream_mixed_record_start(call->audiostream);
+ }
+ call->record_active=TRUE;
+}
+
+/**
+ * Stop call recording.
+**/
+void linphone_call_stop_recording(LinphoneCall *call){
+ if (call->audiostream && !call->params.in_conference){
+ audio_stream_mixed_record_stop(call->audiostream);
+ }
+ call->record_active=FALSE;
+}
/**
* @}
if (from)
{
snprintf(temp,sizeof(temp),"Remote end %s seems to have disconnected, the call is going to be closed.",from);
- free(from);
+ ms_free(from);
}
else
{
if (lc->vtable.display_warning!=NULL)
lc->vtable.display_warning(lc,temp);
linphone_core_terminate_call(lc,call);
+ linphone_core_play_named_tone(lc,LinphoneToneCallFailed);
}
static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
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);
+ as=call->audiostream->ms.session;
+ if (call->audiostream->ms.ticker)
+ audio_load=ms_ticker_get_average_load(call->audiostream->ms.ticker);
}
if (call->videostream!=NULL){
- if (call->videostream->ticker)
- video_load=ms_ticker_get_average_load(call->videostream->ticker);
- vs=call->videostream->session;
+ if (call->videostream->ms.ticker)
+ video_load=ms_ticker_get_average_load(call->videostream->ms.ticker);
+ vs=call->videostream->ms.session;
}
report_bandwidth(call,as,vs);
ms_message("Thread processing load: audio=%f\tvideo=%f",audio_load,video_load);
}
+
+#ifdef BUILD_UPNP
+ linphone_upnp_call_process(call);
+#endif //BUILD_UPNP
+
#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;
+ if (call->ice_session == NULL) call->videostream->ms.ice_check_list = NULL;
// Beware that the application queue should not depend on treatments fron the
// mediastreamer queue.
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);
+ call->stats[LINPHONE_CALL_STATS_VIDEO].round_trip_delay = rtp_session_get_round_trip_propagation(call->videostream->ms.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;
+ update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
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));
+ memcpy(&call->stats[LINPHONE_CALL_STATS_VIDEO].jitter_stats, rtp_session_get_jitter_stats(call->videostream->ms.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;
+ update_local_stats(&call->stats[LINPHONE_CALL_STATS_VIDEO],(MediaStream*)call->videostream);
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)
OrtpEvent *ev;
/* Ensure there is no dangling ICE check list. */
- if (call->ice_session == NULL) call->audiostream->ice_check_list = NULL;
+ if (call->ice_session == NULL) call->audiostream->ms.ice_check_list = NULL;
// Beware that the application queue should not depend on treatments fron the
// mediastreamer queue.
} 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);
+ call->stats[LINPHONE_CALL_STATS_AUDIO].round_trip_delay = rtp_session_get_round_trip_propagation(call->audiostream->ms.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;
+ update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
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));
+ memcpy(&call->stats[LINPHONE_CALL_STATS_AUDIO].jitter_stats, rtp_session_get_jitter_stats(call->audiostream->ms.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;
+ update_local_stats(&call->stats[LINPHONE_CALL_STATS_AUDIO],(MediaStream*)call->audiostream);
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)
}
}
+/**
+ * Returns true if the call is part of the conference.
+ * @ingroup conferencing
+**/
bool_t linphone_call_is_in_conference(const LinphoneCall *call) {
return call->params.in_conference;
}
projects / linphone / blobdiff