forked from mirrors/linphone-iphone
submit quality report on call end instead of on call release and add doxygen documentation
This commit is contained in:
Gautier Pelloux-Prayer
parent
7553aa6492
commit
2b5e2f2a81
2 changed files with 53 additions and 36 deletions
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@ -794,11 +794,13 @@ void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const
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if (lc->vtable.call_state_changed)
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lc->vtable.call_state_changed(lc,call,cstate,message);
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if (cstate==LinphoneCallReleased){
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if (cstate==LinphoneCallEnd){
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if (call->log->status == LinphoneCallSuccess)
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linphone_reporting_publish(call);
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}
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if (cstate==LinphoneCallReleased){
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if (call->op!=NULL) {
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/*transfer the last error so that it can be obtained even in Released state*/
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if (call->non_op_error.reason==SalReasonNone){
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@ -2718,7 +2720,7 @@ static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
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void linphone_call_stats_fill(LinphoneCallStats *stats, MediaStream *ms, OrtpEvent *ev){
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OrtpEventType evt=ortp_event_get_type(ev);
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OrtpEventData *evd=ortp_event_get_data(ev);
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if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
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stats->round_trip_delay = rtp_session_get_round_trip_propagation(ms->sessions.rtp_session);
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if(stats->received_rtcp != NULL)
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@ -2753,11 +2755,11 @@ void linphone_call_handle_stream_events(LinphoneCall *call, int stream_index){
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MediaStream *ms=stream_index==0 ? (MediaStream *)call->audiostream : (MediaStream *)call->videostream; /*assumption to remove*/
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OrtpEvQueue *evq;
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OrtpEvent *ev;
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if (ms==NULL) return;
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/* Ensure there is no dangling ICE check list. */
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if (call->ice_session == NULL) ms->ice_check_list = NULL;
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switch(ms->type){
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case AudioStreamType:
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audio_stream_iterate((AudioStream*)ms);
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@ -2776,10 +2778,10 @@ void linphone_call_handle_stream_events(LinphoneCall *call, int stream_index){
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while ((evq=stream_index==0 ? call->audiostream_app_evq : call->videostream_app_evq) && (NULL != (ev=ortp_ev_queue_get(evq)))){
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OrtpEventType evt=ortp_event_get_type(ev);
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OrtpEventData *evd=ortp_event_get_data(ev);
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linphone_call_stats_fill(&call->stats[stream_index],ms,ev);
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linphone_call_notify_stats_updated(call,stream_index);
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if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
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if (ms->type==AudioStreamType)
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linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
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@ -26,12 +26,20 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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extern "C"{
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#endif
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/**
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* Linphone quality report sub object storing address related information (ip / port / MAC).
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*/
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typedef struct reporting_addr {
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char * ip;
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int port;
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uint32_t ssrc;
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} reporting_addr_t;
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/**
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* Linphone quality report sub object storing media metrics information as required by RFC035.
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*/
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typedef struct reporting_content_metrics {
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// timestamps - mandatory
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struct {
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@ -45,18 +53,13 @@ typedef struct reporting_content_metrics {
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char * payload_desc; // mime type
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int sample_rate; // clock rate
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int frame_duration; // to check (ptime?) - audio only
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// int frame_ocets;
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// int frames_per_sec;
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// int packets_per_sec;
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char * fmtp;
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int packet_loss_concealment; // in voip metrics - audio only
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// char * silence_suppression_state;
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} session_description;
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// jitter buffet - optional
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struct {
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int adaptive; // constant
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// int rate; // constant
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int nominal; // no may vary during the call <- average? worst score?
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int max; // no may vary during the call <- average?
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int abs_max; // constant
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@ -64,25 +67,14 @@ typedef struct reporting_content_metrics {
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// packet loss - optional
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struct {
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float network_packet_loss_rate; // voip metrics (loss rate) + conversion
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float jitter_buffer_discard_rate; //idem
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float network_packet_loss_rate;
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float jitter_buffer_discard_rate;
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} packet_loss;
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// burst gap loss - optional
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// (no) currently not implemented
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// struct {
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// int burst_loss_density;
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// int burst_duration;
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// float gap_loss_density;
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// int gap_duration;
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// int min_gap_threshold;
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// } burst_gap_loss;
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// delay - optional
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struct {
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int round_trip_delay; // no - vary
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int end_system_delay; // no - not implemented yet
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// int one_way_delay;
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int symm_one_way_delay; // no - vary (depends on round_trip_delay) + not implemented (depends on end_system_delay)
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int interarrival_jitter; // no - not implemented yet
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int mean_abs_jitter; // to check
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@ -92,7 +84,6 @@ typedef struct reporting_content_metrics {
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struct {
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int level; // no - vary
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int noise_level; // no - vary
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// int residual_echo_return_loss;
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} signal;
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// quality estimates - optional
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@ -101,20 +92,14 @@ typedef struct reporting_content_metrics {
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int rcq; //voip metrics R factor - no - vary or avg in [0..120]
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float moslq; // no - vary or avg - voip metrics - in [0..4.9]
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float moscq; // no - vary or avg - voip metrics - in [0..4.9]
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// int extri;
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// int extro;
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// char * rlqestalg;
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// char * rcqestalg;
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// char * moslqestalg;
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// char * moscqestalg;
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// char * extriestalg;
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// char * extroutestalg;
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// char * qoestalg;
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} quality_estimates;
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} reporting_content_metrics_t;
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/**
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* Linphone quality report main object created by function linphone_reporting_new().
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* It contains all fields required by RFC6035
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*/
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typedef struct reporting_session_report {
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struct {
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char * call_id;
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@ -138,10 +123,40 @@ typedef struct reporting_session_report {
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reporting_session_report_t * linphone_reporting_new();
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void linphone_reporting_destroy(reporting_session_report_t * report);
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/**
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* Fill media information about a given call. This function must be called before
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* stopping the media stream.
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* @param call #LinphoneCall object to consider
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* @param stats_type the media type (LINPHONE_CALL_STATS_AUDIO or LINPHONE_CALL_STATS_VIDEO)
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*
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*/
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void linphone_reporting_update(LinphoneCall * call, int stats_type);
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/**
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* Fill IP information about a given call. This function must be called each
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* time state is 'LinphoneCallStreamsRunning' since IP might be updated (if we
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* found a direct route between caller and callee for example).
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* @param call #LinphoneCall object to consider
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*
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*/
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void linphone_reporting_update_ip(LinphoneCall * call);
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/**
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* Publish the report on the call end.
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* @param call #LinphoneCall object to consider
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*
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*/
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void linphone_reporting_publish(LinphoneCall* call);
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/**
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* Update publish report data with fresh RTCP stats, if needed.
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* @param call #LinphoneCall object to consider
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* @param stats_type the media type (LINPHONE_CALL_STATS_AUDIO or LINPHONE_CALL_STATS_VIDEO)
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*
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*/
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void linphone_reporting_call_stats_updated(LinphoneCall *call, int stats_type);
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#ifdef __cplusplus
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}
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#endif
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