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/*
The oRTP library is an RTP (Realtime Transport Protocol - rfc1889) stack.
Copyright (C) 2001 Simon MORLAT simon.morlat@linphone.org
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <ortp.h>
#include "rtpmod.h"
#include "jitterctl.h"
void split_and_queue(queue_t *q, int maxrqsz, mblk_t *mp, rtp_header_t *rtp, int *discarded)
{
mblk_t *mdata,*tmp;
int header_size;
*discarded=0;
header_size=RTP_FIXED_HEADER_SIZE+ (4*rtp->cc);
if ((mp->b_wptr - mp->b_rptr)==header_size){
ortp_debug("Rtp packet contains no data.");
(*discarded)++;
freemsg(mp);
return;
}
/* creates a new mblk_t to be linked with the rtp header*/
mdata=dupb(mp);
mp->b_wptr=mp->b_rptr+header_size;
mdata->b_rptr+=header_size;
/* link proto with data */
mp->b_cont=mdata;
/* and then add the packet to the queue */
rtp_putq(q,mp);
/* make some checks: q size must not exceed RtpStream::max_rq_size */
while (q->q_mcount > maxrqsz)
{
/* remove the oldest mblk_t */
tmp=getq(q);
if (mp!=NULL)
{
ortp_debug("rtp_putq: Queue is full. Discarding message with ts=%i",((rtp_header_t*)mp->b_rptr)->timestamp);
freemsg(tmp);
(*discarded)++;
}
}
}
void rtp_parse(RtpSession *session, mblk_t *mp, guint32 local_str_ts)
{
gint i;
rtp_header_t *rtp;
int msgsize;
RtpStream *rtpstream=&session->rtp;
rtp_stats_t *stats=&rtpstream->stats;
g_return_if_fail(mp!=NULL);
msgsize=msgdsize(mp);
ortp_global_stats.hw_recv+=msgsize;
stats->hw_recv+=msgsize;
ortp_global_stats.packet_recv++;
stats->packet_recv++;
session->rtp.hwrcv_since_last_SR++;
rtp=(rtp_header_t*)mp->b_rptr;
if (rtp->version!=2)
{
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(guint32) > (msgsize-RTP_FIXED_HEADER_SIZE)){
ortp_debug("Receiving too short rtp packet.");
stats->bad++;
ortp_global_stats.bad++;
freemsg(mp);
return;
}
for (i=0;i<rtp->cc;i++)
rtp->csrc[i]=ntohl(rtp->csrc[i]);
if (session->recv_ssrc!=0)
{
/*the ssrc is set, so we must check it */
if (session->recv_ssrc!=rtp->ssrc){
/*ortp_debug("rtp_parse: bad ssrc - %i",rtp->ssrc);*/
session->recv_ssrc=rtp->ssrc;
rtp_signal_table_emit(&session->on_ssrc_changed);
}
}else session->recv_ssrc=rtp->ssrc;
/* update some statistics */
if (rtp->seq_number>rtpstream->hwrcv_extseq.split.lo){
rtpstream->hwrcv_extseq.split.lo=rtp->seq_number;
}else if (rtp->seq_number<200 && rtpstream->hwrcv_extseq.split.lo>((1<<16) - 200)){
/* this is a check for sequence number looping */
rtpstream->hwrcv_extseq.split.lo=rtp->seq_number;
rtpstream->hwrcv_extseq.split.hi++;
}
/* check for possible telephone events */
if (rtp->paytype==session->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;
}
if (!(session->flags & RTP_SESSION_RECV_SYNC)){
gint32 slide=0;
gint32 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,&rtp->timestamp);
}
else if (RTP_TIMESTAMP_IS_NEWER_THAN(session->rtp.rcv_last_ts,rtp->timestamp)){
/* avoid very old packet to enqueued, because the user is no more supposed to get them */
ortp_debug("rtp_parse: silently discarding very 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;
}
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