static void notify_events_ended(RtpSession *session, telephone_event_t *events, int num){
int i;
for (i=0;i<num;i++){
if (events[i].E==1){
rtp_signal_table_emit2(&session->on_telephone_event,(gpointer)(long)events[i].event);
}
}
}
static void notify_tev(RtpSession *session, telephone_event_t *event){
OrtpEvent *ev;
OrtpEventData *evd;
rtp_signal_table_emit2(&session->on_telephone_event,(long)(long)event[0].event);
if (session->eventqs!=NULL){
ev=ortp_event_new(ORTP_EVENT_TELEPHONE_EVENT);
evd=ortp_event_get_data(ev);
evd->packet=dupmsg(session->current_tev);
evd->info.telephone_event=event[0].event;
rtp_session_dispatch_event(session,ev);
}
}
void rtp_session_rtp_parse(RtpSession *session, mblk_t *mp, uint32_t local_str_ts, struct sockaddr *addr, socklen_t addrlen)
{
int i;
int discarded;
int duplicate;
rtp_header_t *rtp;
int msgsize;
RtpStream *rtpstream=&session->rtp;
rtp_stats_t *stats=&rtpstream->stats;
msgsize=(int)(mp->b_wptr-mp->b_rptr);
if (msgsize<RTP_FIXED_HEADER_SIZE){
ortp_warning("Packet too small to be a rtp packet (%i)!",msgsize);
rtpstream->stats.bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
rtp=(rtp_header_t*)mp->b_rptr;
if (rtp->version!=2)
{
/* try to see if it is a STUN packet */
uint16_t stunlen=*((uint16_t*)(mp->b_rptr + sizeof(uint16_t)));
stunlen = ntohs(stunlen);
if (stunlen+20==mp->b_wptr-mp->b_rptr){
/* this looks like a stun packet */
if (session->eventqs!=NULL){
OrtpEvent *ev=ortp_event_new(ORTP_EVENT_STUN_PACKET_RECEIVED);
OrtpEventData *ed=ortp_event_get_data(ev);
ed->packet=mp;
memcpy(&ed->source_addr,addr,addrlen);
ed->source_addrlen=addrlen;
ed->info.socket_type = OrtpRTPSocket;
rtp_session_dispatch_event(session,ev);
return;
}
}
/* discard in two case: the packet is not stun OR nobody is interested by STUN (no eventqs) */
ortp_debug("Receiving rtp packet with version number !=2...discarded");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
/* only count non-stun packets. */
ortp_global_stats.packet_recv++;
stats->packet_recv++;
ortp_global_stats.hw_recv+=msgsize;
stats->hw_recv+=msgsize;
session->rtp.hwrcv_since_last_SR++;
session->rtcp_xr_stats.rcv_since_last_stat_summary++;
/* convert all header data from network order to host order */
rtp->seq_number=ntohs(rtp->seq_number);
rtp->timestamp=ntohl(rtp->timestamp);
rtp->ssrc=ntohl(rtp->ssrc);
/* convert csrc if necessary */
if (rtp->cc*sizeof(uint32_t) > (uint32_t) (msgsize-RTP_FIXED_HEADER_SIZE)){
ortp_debug("Receiving too short rtp packet.");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
#ifndef PERF
/* Write down the last RTP/RTCP packet reception time. */
ortp_gettimeofday(&session->last_recv_time, NULL);
#endif
for (i=0;i<rtp->cc;i++)
rtp->csrc[i]=ntohl(rtp->csrc[i]);
/*the goal of the following code is to lock on an incoming SSRC to avoid
receiving "mixed streams"*/
if (session->ssrc_set){
/*the ssrc is set, so we must check it */
if (session->rcv.ssrc!=rtp->ssrc){
if (session->inc_ssrc_candidate==rtp->ssrc){
session->inc_same_ssrc_count++;
}else{
session->inc_same_ssrc_count=0;
session->inc_ssrc_candidate=rtp->ssrc;
}
if (session->inc_same_ssrc_count>=session->rtp.ssrc_changed_thres){
/* store the sender rtp address to do symmetric RTP */
if (!session->use_connect){
if (session->rtp.gs.socket>0 && session->symmetric_rtp){
/* store the sender rtp address to do symmetric RTP */
memcpy(&session->rtp.gs.rem_addr,addr,addrlen);
session->rtp.gs.rem_addrlen=addrlen;
}
}
session->rtp.rcv_last_ts = rtp->timestamp;
session->rcv.ssrc=rtp->ssrc;
rtp_signal_table_emit(&session->on_ssrc_changed);
}else{
/*discard the packet*/
ortp_debug("Receiving packet with unknown ssrc.");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
} else{
/* The SSRC change must not happen if we still receive
ssrc from the initial source. */
session->inc_same_ssrc_count=0;
}
}else{
session->ssrc_set=TRUE;
session->rcv.ssrc=rtp->ssrc;
if (!session->use_connect){
if (session->rtp.gs.socket>0 && session->symmetric_rtp){
/* store the sender rtp address to do symmetric RTP */
memcpy(&session->rtp.gs.rem_addr,addr,addrlen);
session->rtp.gs.rem_addrlen=addrlen;
}
}
}
/* update some statistics */
{
poly32_t *extseq=(poly32_t*)&rtpstream->hwrcv_extseq;
if (rtp->seq_number>extseq->split.lo){
extseq->split.lo=rtp->seq_number;
}else if (rtp->seq_number<200 && extseq->split.lo>((1<<16) - 200)){
/* this is a check for sequence number looping */
extseq->split.lo=rtp->seq_number;
extseq->split.hi++;
}
/* the first sequence number received should be initialized at the beginning
or at any resync, so that the first receiver reports contains valid loss rate*/
if (!(session->flags & RTP_SESSION_RECV_SEQ_INIT)){
rtp_session_set_flag(session, RTP_SESSION_RECV_SEQ_INIT);
rtpstream->hwrcv_seq_at_last_SR=rtp->seq_number-1;
session->rtcp_xr_stats.rcv_seq_at_last_stat_summary=rtp->seq_number-1;
}
if (stats->packet_recv==1){
session->rtcp_xr_stats.first_rcv_seq=extseq->one;
}
session->rtcp_xr_stats.last_rcv_seq=extseq->one;
}
/* check for possible telephone events */
if (rtp_profile_is_telephone_event(session->snd.profile, rtp->paytype)){
queue_packet(&session->rtp.tev_rq,session->rtp.max_rq_size,mp,rtp,&discarded,&duplicate);
stats->discarded+=discarded;
ortp_global_stats.discarded+=discarded;
stats->packet_dup_recv+=duplicate;
ortp_global_stats.packet_dup_recv+=duplicate;
session->rtcp_xr_stats.discarded_count += discarded;
session->rtcp_xr_stats.dup_since_last_stat_summary += duplicate;
return;
}
/* check for possible payload type change, in order to update accordingly our clock-rate dependant
parameters */
if (session->hw_recv_pt!=rtp->paytype){
rtp_session_update_payload_type(session,rtp->paytype);
}
/* Drop the packets while the RTP_SESSION_FLUSH flag is set. */
if (session->flags & RTP_SESSION_FLUSH) {
freemsg(mp);
return;
}
jitter_control_new_packet(&session->rtp.jittctl,rtp->timestamp,local_str_ts);
update_rtcp_xr_stat_summary(session, mp, local_str_ts);
if (session->flags & RTP_SESSION_FIRST_PACKET_DELIVERED) {
/* detect timestamp important jumps in the future, to workaround stupid rtp senders */
if (RTP_TIMESTAMP_IS_NEWER_THAN(rtp->timestamp,session->rtp.rcv_last_ts+session->rtp.ts_jump)){
ortp_warning("rtp_parse: timestamp jump in the future detected.");
rtp_signal_table_emit2(&session->on_timestamp_jump,&rtp->timestamp);
}
else if (RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(session->rtp.rcv_last_ts,rtp->timestamp)
|| RTP_SEQ_IS_STRICTLY_GREATER_THAN(session->rtp.rcv_last_seq,rtp->seq_number)){
/* don't queue packets older than the last returned packet to the application, or whose sequence number
is behind the last packet returned to the application*/
/* Call timstamp jumb in case of
* large negative Ts jump or if ts is set to 0
*/
if ( RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(session->rtp.rcv_last_ts, rtp->timestamp + session->rtp.ts_jump) ){
ortp_warning("rtp_parse: negative timestamp jump detected");
rtp_signal_table_emit2(&session->on_timestamp_jump, &rtp->timestamp);
}
ortp_debug("rtp_parse: discarding too old packet (ts=%i)",rtp->timestamp);
freemsg(mp);
stats->outoftime++;
ortp_global_stats.outoftime++;
session->rtcp_xr_stats.discarded_count++;
return;
}
}
if (queue_packet(&session->rtp.rq,session->rtp.max_rq_size,mp,rtp,&discarded,&duplicate))
jitter_control_update_size(&session->rtp.jittctl,&session->rtp.rq);
stats->discarded+=discarded;
ortp_global_stats.discarded+=discarded;
stats->packet_dup_recv+=duplicate;
ortp_global_stats.packet_dup_recv+=duplicate;
session->rtcp_xr_stats.discarded_count += discarded;
session->rtcp_xr_stats.dup_since_last_stat_summary += duplicate;
if ((discarded == 0) && (duplicate == 0)) {
session->rtcp_xr_stats.rcv_count++;
}
}
void rtp_session_rtp_parse(RtpSession *session, mblk_t *mp, uint32_t local_str_ts, struct sockaddr *addr, socklen_t addrlen)
{
int i;
rtp_header_t *rtp;
int msgsize;
RtpStream *rtpstream=&session->rtp;
rtp_stats_t *stats=&rtpstream->stats;
return_if_fail(mp!=NULL);
msgsize=msgdsize(mp);
if (msgsize<RTP_FIXED_HEADER_SIZE){
ortp_warning("Packet too small to be a rtp packet (%i)!",msgsize);
rtpstream->stats.bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
rtp=(rtp_header_t*)mp->b_rptr;
if (rtp->version!=2)
{
/* try to see if it is a STUN packet */
uint16_t stunlen=*((uint16_t*)(mp->b_rptr + sizeof(uint16_t)));
stunlen = ntohs(stunlen);
if (stunlen+20==mp->b_wptr-mp->b_rptr){
/* this looks like a stun packet */
if (session->eventqs!=NULL){
OrtpEvent *ev=ortp_event_new(ORTP_EVENT_STUN_PACKET_RECEIVED);
OrtpEventData *ed=ortp_event_get_data(ev);
ed->packet=mp;
ed->ep=rtp_endpoint_new(addr,addrlen);
rtp_session_dispatch_event(session,ev);
return;
}
}
freemsg(mp);
return;
}
/* only count non-stun packets. */
ortp_global_stats.packet_recv++;
stats->packet_recv++;
ortp_global_stats.hw_recv+=msgsize;
stats->hw_recv+=msgsize;
session->rtp.hwrcv_since_last_SR++;
if (rtp->version!=2)
{
/* discard*/
ortp_debug("Receiving rtp packet with version number !=2...discarded");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
/* convert all header data from network order to host order */
rtp->seq_number=ntohs(rtp->seq_number);
rtp->timestamp=ntohl(rtp->timestamp);
rtp->ssrc=ntohl(rtp->ssrc);
/* convert csrc if necessary */
if (rtp->cc*sizeof(uint32_t) > (uint32_t) (msgsize-RTP_FIXED_HEADER_SIZE)){
ortp_debug("Receiving too short rtp packet.");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
/* Write down the last RTP/RTCP packet reception time. */
gettimeofday(&session->last_recv_time, NULL);
for (i=0;i<rtp->cc;i++)
rtp->csrc[i]=ntohl(rtp->csrc[i]);
if (session->rcv.ssrc!=0)
{
/*the ssrc is set, so we must check it */
if (session->rcv.ssrc!=rtp->ssrc){
/*ortp_debug("rtp_parse: bad ssrc - %i",rtp->ssrc);*/
session->rcv.ssrc=rtp->ssrc;
rtp_signal_table_emit(&session->on_ssrc_changed);
}
}else session->rcv.ssrc=rtp->ssrc;
/* update some statistics */
{
poly32_t *extseq=(poly32_t*)&rtpstream->hwrcv_extseq;
if (rtp->seq_number>extseq->split.lo){
extseq->split.lo=rtp->seq_number;
}else if (rtp->seq_number<200 && extseq->split.lo>((1<<16) - 200)){
/* this is a check for sequence number looping */
extseq->split.lo=rtp->seq_number;
extseq->split.hi++;
}
}
/* check for possible telephone events */
if (rtp->paytype==session->rcv.telephone_events_pt){
split_and_queue(&session->rtp.tev_rq,session->rtp.max_rq_size,mp,rtp,&i);
stats->discarded+=i;
ortp_global_stats.discarded+=i;
return;
}
/* check for possible payload type change, in order to update accordingly our clock-rate dependant
parameters */
if (session->hw_recv_pt!=rtp->paytype){
rtp_session_update_payload_type(session,rtp->paytype);
}
if (session->flags & RTP_SESSION_FIRST_PACKET_DELIVERED) {
int32_t slide=0;
int32_t safe_delay=0;
jitter_control_new_packet(&session->rtp.jittctl,rtp->timestamp,local_str_ts,&slide,&safe_delay);
session->rtp.rcv_diff_ts=session->rtp.hwrcv_diff_ts + slide - safe_delay;
ortp_debug(" rcv_diff_ts=%i", session->rtp.rcv_diff_ts);
/* detect timestamp important jumps in the future, to workaround stupid rtp senders */
if (RTP_TIMESTAMP_IS_NEWER_THAN(rtp->timestamp,session->rtp.rcv_last_ts+session->rtp.ts_jump)){
ortp_debug("rtp_parse: timestamp jump ?");
rtp_signal_table_emit2(&session->on_timestamp_jump,(long)&rtp->timestamp);
}
else if (RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(session->rtp.rcv_last_ts,rtp->timestamp)){
/* don't queue packets older than the last returned packet to the application*/
/* Call timstamp jumb in case of
* large negative Ts jump or if ts is set to 0
*/
if ( RTP_TIMESTAMP_IS_STRICTLY_NEWER_THAN(session->rtp.rcv_last_ts, rtp->timestamp + session->rtp.ts_jump) ){
ortp_warning("rtp_parse: negative timestamp jump");
rtp_signal_table_emit2(&session->on_timestamp_jump,
(long)&rtp->timestamp);
}
ortp_debug("rtp_parse: discarding too old packet (ts=%i)",rtp->timestamp);
freemsg(mp);
stats->outoftime++;
ortp_global_stats.outoftime++;
return;
}
}
split_and_queue(&session->rtp.rq,session->rtp.max_rq_size,mp,rtp,&i);
stats->discarded+=i;
ortp_global_stats.discarded+=i;
}
/* for high level telephony event callback */
void rtp_session_check_telephone_events(RtpSession *session, mblk_t *m0)
{
telephone_event_t *events,*evbuf;
int num;
int i;
mblk_t *mp;
rtp_header_t *hdr;
mblk_t *cur_tev;
hdr=(rtp_header_t*)m0->b_rptr;
mp=m0->b_cont;
num=(mp->b_wptr-mp->b_rptr)/sizeof(telephone_event_t);
events=(telephone_event_t*)mp->b_rptr;
if (hdr->markbit==1)
{
/* this is a start of new events. Store the event buffer for later use*/
if (session->current_tev!=NULL) {
freemsg(session->current_tev);
session->current_tev=NULL;
}
session->current_tev=copymsg(m0);
/* handle the case where the events are short enough to end within the packet that has the marker bit*/
notify_events_ended(session,events,num);
}
/* whatever there is a markbit set or not, we parse the packet and compare it to previously received one */
cur_tev=session->current_tev;
if (cur_tev!=NULL)
{
/* first compare timestamp, they must be identical */
if (((rtp_header_t*)cur_tev->b_rptr)->timestamp==
((rtp_header_t*)m0->b_rptr)->timestamp)
{
evbuf=(telephone_event_t*)cur_tev->b_cont;
for (i=0;i<num;i++)
{
if (events[i].E==1)
{
/* update events that have ended */
if (evbuf[i].E==0){
evbuf[i].E=1;
/* this is a end of event, report it */
rtp_signal_table_emit2(&session->on_telephone_event,(gpointer)(long)events[i].event);
}
}
}
}
else
{
/* timestamp are not identical: this is not the same events*/
if (session->current_tev!=NULL) {
freemsg(session->current_tev);
session->current_tev=NULL;
}
session->current_tev=dupmsg(m0);
}
}
else
{
/* there is no pending events, but we did not received marked bit packet
either the sending implementation is not compliant, either it has been lost,
we must deal with it anyway.*/
session->current_tev=copymsg(m0);
/* inform the application if there are tone ends */
notify_events_ended(session,events,num);
}
}