#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;
+ {
+ const LinphoneCallParams *params=linphone_call_get_current_params(call);
+ if (params->has_video && !call->videostream_encrypted) {
+ return FALSE;
+ }
}
#endif
#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;
+
+ ms_message("Video stream is %s", encrypted ? "encrypted" : "not encrypted");
call->videostream_encrypted=encrypted;
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,¶ms);
+ {
+ const LinphoneCallParams *params=linphone_call_get_current_params(call);
+ if (params->has_video) {
+ OrtpZrtpParams params;
+ ms_message("Trying to enable encryption on video stream");
+ params.zid_file=NULL; //unused
+ video_stream_enable_zrtp(call->videostream,call->audiostream,¶ms);
+ }
}
#endif
}
l=ms_list_append(l,pt);
md->streams[0].payloads=l;
- // if ZRTP is enabled, put the hello hash into the audiostream's desc
- if (call->audiostream && call->audiostream->ms.zrtp_context!=NULL){
- ortp_zrtp_get_hello_hash(call->audiostream->ms.zrtp_context,
- md->streams[0].zrtp_hello_hash,
- sizeof(md->streams[0].zrtp_hello_hash));
- ms_message("Audio stream zrtp hash: %s", md->streams[0].zrtp_hello_hash);
- }
-
if (call->params.has_video){
md->n_active_streams++;
md->streams[1].rtp_port=call->video_port;
md->streams[1].type=SalVideo;
l=make_codec_list(lc,lc->codecs_conf.video_codecs,0,NULL,-1);
md->streams[1].payloads=l;
- // if ZRTP is enabled, put the hello hash into the audiostream's desc
- if (call->videostream->ms.zrtp_context!=NULL){
- ortp_zrtp_get_hello_hash(call->videostream->ms.zrtp_context,
- md->streams[1].zrtp_hello_hash,
- sizeof(md->streams[1].zrtp_hello_hash));
- ms_message("Video stream zrtp hash: %s", md->streams[1].zrtp_hello_hash);
- }
}
if (md->n_total_streams < md->n_active_streams)
md->n_total_streams = md->n_active_streams;
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);
_linphone_call_params_copy(&call->params,params);
sal_op_set_custom_header(call->op,call->params.custom_headers);
}
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);
void linphone_call_init_media_streams(LinphoneCall *call){
linphone_call_init_audio_stream(call);
linphone_call_init_video_stream(call);
-
- // moved from linphone_call_start_media_streams, because ZRTP needs to be
- // at least partially initialized so that the SDP can contain 'zrtp-hash'
- 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=call->core->zrtp_secrets_cache;
- audio_stream_enable_zrtp(call->audiostream,¶ms);
- } else if (call->params.media_encryption==LinphoneMediaEncryptionSRTP){
- call->current_params.media_encryption=linphone_call_are_all_streams_encrypted(call) ?
- LinphoneMediaEncryptionSRTP : LinphoneMediaEncryptionNone;
- }
}
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.video_codec = rtp_profile_get_payload(call->video_profile, used_pt);
call->current_params.has_video=TRUE;
video_stream_enable_adaptive_bitrate_control(call->videostream,
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);
SalProtoRtpAvp,SalVideo);
#endif
+ call->current_params.audio_codec = NULL;
+ call->current_params.video_codec = NULL;
+
if ((call->audiostream == NULL) && (call->videostream == NULL)) {
ms_fatal("start_media_stream() called without prior init !");
return;
call->playing_ringbacktone=send_ringbacktone;
call->up_bw=linphone_core_get_upload_bandwidth(lc);
- // ZRTP was initialized in linphone_call_init_media_streams with a
- // partially iniitalized RtpSession, and now needs to get an update
if (call->params.media_encryption==LinphoneMediaEncryptionZRTP) {
- ortp_zrtp_start_engine(call->audiostream->ms.zrtp_context,call->audiostream->ms.session);
+ 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,¶ms);
+ }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
}
#endif //BUILD_UPNP
-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);
+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) {
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);
}
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;
}
/**
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) {
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)
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) {
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)
projects / linphone / blobdiff