submit quality report on call end instead of on call release and add doxygen documentation

parent 7553aa64
......@@ -794,11 +794,13 @@ void linphone_call_set_state(LinphoneCall *call, LinphoneCallState cstate, const
if (lc->vtable.call_state_changed)
lc->vtable.call_state_changed(lc,call,cstate,message);
if (cstate==LinphoneCallReleased){
if (cstate==LinphoneCallEnd){
if (call->log->status == LinphoneCallSuccess)
linphone_reporting_publish(call);
}
if (cstate==LinphoneCallReleased){
if (call->op!=NULL) {
/*transfer the last error so that it can be obtained even in Released state*/
if (call->non_op_error.reason==SalReasonNone){
......@@ -2718,7 +2720,7 @@ static void handle_ice_events(LinphoneCall *call, OrtpEvent *ev){
void linphone_call_stats_fill(LinphoneCallStats *stats, MediaStream *ms, OrtpEvent *ev){
OrtpEventType evt=ortp_event_get_type(ev);
OrtpEventData *evd=ortp_event_get_data(ev);
if (evt == ORTP_EVENT_RTCP_PACKET_RECEIVED) {
stats->round_trip_delay = rtp_session_get_round_trip_propagation(ms->sessions.rtp_session);
if(stats->received_rtcp != NULL)
......@@ -2753,11 +2755,11 @@ void linphone_call_handle_stream_events(LinphoneCall *call, int stream_index){
MediaStream *ms=stream_index==0 ? (MediaStream *)call->audiostream : (MediaStream *)call->videostream; /*assumption to remove*/
OrtpEvQueue *evq;
OrtpEvent *ev;
if (ms==NULL) return;
/* Ensure there is no dangling ICE check list. */
if (call->ice_session == NULL) ms->ice_check_list = NULL;
switch(ms->type){
case AudioStreamType:
audio_stream_iterate((AudioStream*)ms);
......@@ -2776,10 +2778,10 @@ void linphone_call_handle_stream_events(LinphoneCall *call, int stream_index){
while ((evq=stream_index==0 ? call->audiostream_app_evq : call->videostream_app_evq) && (NULL != (ev=ortp_ev_queue_get(evq)))){
OrtpEventType evt=ortp_event_get_type(ev);
OrtpEventData *evd=ortp_event_get_data(ev);
linphone_call_stats_fill(&call->stats[stream_index],ms,ev);
linphone_call_notify_stats_updated(call,stream_index);
if (evt == ORTP_EVENT_ZRTP_ENCRYPTION_CHANGED){
if (ms->type==AudioStreamType)
linphone_call_audiostream_encryption_changed(call, evd->info.zrtp_stream_encrypted);
......
......@@ -26,12 +26,20 @@ Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
extern "C"{
#endif
/**
* Linphone quality report sub object storing address related information (ip / port / MAC).
*/
typedef struct reporting_addr {
char * ip;
int port;
uint32_t ssrc;
} reporting_addr_t;
/**
* Linphone quality report sub object storing media metrics information as required by RFC035.
*/
typedef struct reporting_content_metrics {
// timestamps - mandatory
struct {
......@@ -45,18 +53,13 @@ typedef struct reporting_content_metrics {
char * payload_desc; // mime type
int sample_rate; // clock rate
int frame_duration; // to check (ptime?) - audio only
// int frame_ocets;
// int frames_per_sec;
// int packets_per_sec;
char * fmtp;
int packet_loss_concealment; // in voip metrics - audio only
// char * silence_suppression_state;
} session_description;
// jitter buffet - optional
struct {
int adaptive; // constant
// int rate; // constant
int nominal; // no may vary during the call <- average? worst score?
int max; // no may vary during the call <- average?
int abs_max; // constant
......@@ -64,25 +67,14 @@ typedef struct reporting_content_metrics {
// packet loss - optional
struct {
float network_packet_loss_rate; // voip metrics (loss rate) + conversion
float jitter_buffer_discard_rate; //idem
float network_packet_loss_rate;
float jitter_buffer_discard_rate;
} packet_loss;
// burst gap loss - optional
// (no) currently not implemented
// struct {
// int burst_loss_density;
// int burst_duration;
// float gap_loss_density;
// int gap_duration;
// int min_gap_threshold;
// } burst_gap_loss;
// delay - optional
struct {
int round_trip_delay; // no - vary
int end_system_delay; // no - not implemented yet
// int one_way_delay;
int symm_one_way_delay; // no - vary (depends on round_trip_delay) + not implemented (depends on end_system_delay)
int interarrival_jitter; // no - not implemented yet
int mean_abs_jitter; // to check
......@@ -92,7 +84,6 @@ typedef struct reporting_content_metrics {
struct {
int level; // no - vary
int noise_level; // no - vary
// int residual_echo_return_loss;
} signal;
// quality estimates - optional
......@@ -101,20 +92,14 @@ typedef struct reporting_content_metrics {
int rcq; //voip metrics R factor - no - vary or avg in [0..120]
float moslq; // no - vary or avg - voip metrics - in [0..4.9]
float moscq; // no - vary or avg - voip metrics - in [0..4.9]
// int extri;
// int extro;
// char * rlqestalg;
// char * rcqestalg;
// char * moslqestalg;
// char * moscqestalg;
// char * extriestalg;
// char * extroutestalg;
// char * qoestalg;
} quality_estimates;
} reporting_content_metrics_t;
/**
* Linphone quality report main object created by function linphone_reporting_new().
* It contains all fields required by RFC6035
*/
typedef struct reporting_session_report {
struct {
char * call_id;
......@@ -138,10 +123,40 @@ typedef struct reporting_session_report {
reporting_session_report_t * linphone_reporting_new();
void linphone_reporting_destroy(reporting_session_report_t * report);
/**
* Fill media information about a given call. This function must be called before
* stopping the media stream.
* @param call #LinphoneCall object to consider
* @param stats_type the media type (LINPHONE_CALL_STATS_AUDIO or LINPHONE_CALL_STATS_VIDEO)
*
*/
void linphone_reporting_update(LinphoneCall * call, int stats_type);
/**
* Fill IP information about a given call. This function must be called each
* time state is 'LinphoneCallStreamsRunning' since IP might be updated (if we
* found a direct route between caller and callee for example).
* @param call #LinphoneCall object to consider
*
*/
void linphone_reporting_update_ip(LinphoneCall * call);
/**
* Publish the report on the call end.
* @param call #LinphoneCall object to consider
*
*/
void linphone_reporting_publish(LinphoneCall* call);
/**
* Update publish report data with fresh RTCP stats, if needed.
* @param call #LinphoneCall object to consider
* @param stats_type the media type (LINPHONE_CALL_STATS_AUDIO or LINPHONE_CALL_STATS_VIDEO)
*
*/
void linphone_reporting_call_stats_updated(LinphoneCall *call, int stats_type);
#ifdef __cplusplus
}
#endif
......
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