+/* tap-rtp-common.c
+ * RTP stream handler functions used by tshark and wireshark
+ *
+ * $Id$
+ *
+ * Copyright 2008, Ericsson AB
+ * By Balint Reczey <balint.reczey@ericsson.com>
+ *
+ * most functions are copied from gtk/rtp_stream.c and gtk/rtp_analisys.c
+ * Copyright 2003, Alcatel Business Systems
+ * By Lars Ruoff <lars.ruoff@gmx.net>
+ *
+ * Wireshark - Network traffic analyzer
+ * By Gerald Combs <gerald@wireshark.org>
+ * Copyright 1998 Gerald Combs
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "globals.h"
+
+#include <epan/tap.h>
+#include "register.h"
+#include <string.h>
+#include <epan/rtp_pt.h>
+#include <epan/addr_resolv.h>
+#include <epan/dissectors/packet-rtp.h>
+#include "gtk/rtp_stream.h"
+#include "tap-rtp-common.h"
+
+
+
+/****************************************************************************/
+/* GCompareFunc style comparison function for _rtp_stream_info */
+gint rtp_stream_info_cmp(gconstpointer aa, gconstpointer bb)
+{
+ const struct _rtp_stream_info* a = aa;
+ const struct _rtp_stream_info* b = bb;
+
+ if (a==b)
+ return 0;
+ if (a==NULL || b==NULL)
+ return 1;
+ if (ADDRESSES_EQUAL(&(a->src_addr), &(b->src_addr))
+ && (a->src_port == b->src_port)
+ && ADDRESSES_EQUAL(&(a->dest_addr), &(b->dest_addr))
+ && (a->dest_port == b->dest_port)
+ && (a->ssrc == b->ssrc))
+ return 0;
+ else
+ return 1;
+}
+
+
+/****************************************************************************/
+/* when there is a [re]reading of packet's */
+void rtpstream_reset(rtpstream_tapinfo_t *tapinfo)
+{
+ GList* list;
+
+ if (tapinfo->mode == TAP_ANALYSE) {
+ /* free the data items first */
+ list = g_list_first(tapinfo->strinfo_list);
+ while (list)
+ {
+ g_free(list->data);
+ list = g_list_next(list);
+ }
+ g_list_free(tapinfo->strinfo_list);
+ tapinfo->strinfo_list = NULL;
+ tapinfo->nstreams = 0;
+ tapinfo->npackets = 0;
+ }
+
+ ++(tapinfo->launch_count);
+
+ return;
+}
+
+void rtpstream_reset_cb(void *arg)
+{
+ rtpstream_reset(arg);
+}
+
+/*
+* rtpdump file format
+*
+* The file starts with the tool to be used for playing this file,
+* the multicast/unicast receive address and the port.
+*
+* #!rtpplay1.0 224.2.0.1/3456\n
+*
+* This is followed by one binary header (RD_hdr_t) and one RD_packet_t
+* structure for each received packet. All fields are in network byte
+* order. We don't need the source IP address since we can do mapping
+* based on SSRC. This saves (a little) space, avoids non-IPv4
+* problems and privacy/security concerns. The header is followed by
+* the RTP/RTCP header and (optionally) the actual payload.
+*/
+
+#define RTPFILE_VERSION "1.0"
+
+/*
+* Write a header to the current output file.
+* The header consists of an identifying string, followed
+* by a binary structure.
+*/
+void rtp_write_header(rtp_stream_info_t *strinfo, FILE *file)
+{
+ guint32 start_sec; /* start of recording (GMT) (seconds) */
+ guint32 start_usec; /* start of recording (GMT) (microseconds)*/
+ guint32 source; /* network source (multicast address) */
+ size_t sourcelen;
+ guint16 port; /* UDP port */
+ guint16 padding; /* 2 padding bytes */
+
+ fprintf(file, "#!rtpplay%s %s/%u\n", RTPFILE_VERSION,
+ get_addr_name(&(strinfo->dest_addr)),
+ strinfo->dest_port);
+
+ start_sec = g_htonl(strinfo->start_sec);
+ start_usec = g_htonl(strinfo->start_usec);
+ /* rtpdump only accepts guint32 as source, will be fake for IPv6 */
+ memset(&source, 0, sizeof source);
+ sourcelen = strinfo->src_addr.len;
+ if (sourcelen > sizeof source)
+ sourcelen = sizeof source;
+ memcpy(&source, strinfo->src_addr.data, sourcelen);
+ port = g_htons(strinfo->src_port);
+ padding = 0;
+
+ if (fwrite(&start_sec, 4, 1, file) == 0)
+ return;
+ if (fwrite(&start_usec, 4, 1, file) == 0)
+ return;
+ if (fwrite(&source, 4, 1, file) == 0)
+ return;
+ if (fwrite(&port, 2, 1, file) == 0)
+ return;
+ if (fwrite(&padding, 2, 1, file) == 0)
+ return;
+}
+
+/* utility function for writing a sample to file in rtpdump -F dump format (.rtp)*/
+void rtp_write_sample(rtp_sample_t* sample, FILE* file)
+{
+ guint16 length; /* length of packet, including this header (may
+ be smaller than plen if not whole packet recorded) */
+ guint16 plen; /* actual header+payload length for RTP, 0 for RTCP */
+ guint32 offset; /* milliseconds since the start of recording */
+
+ length = g_htons(sample->header.frame_length + 8);
+ plen = g_htons(sample->header.frame_length);
+ offset = g_htonl(sample->header.rec_time);
+
+ if (fwrite(&length, 2, 1, file) == 0)
+ return;
+ if (fwrite(&plen, 2, 1, file) == 0)
+ return;
+ if (fwrite(&offset, 4, 1, file) == 0)
+ return;
+ if (fwrite(sample->frame, sample->header.frame_length, 1, file) == 0)
+ return;
+}
+
+
+/****************************************************************************/
+/* whenever a RTP packet is seen by the tap listener */
+int rtpstream_packet(void *arg, packet_info *pinfo, epan_dissect_t *edt _U_, const void *arg2)
+{
+ rtpstream_tapinfo_t *tapinfo = arg;
+ const struct _rtp_info *rtpinfo = arg2;
+ rtp_stream_info_t tmp_strinfo;
+ rtp_stream_info_t *strinfo = NULL;
+ GList* list;
+ rtp_sample_t sample;
+
+ struct _rtp_conversation_info *p_conv_data = NULL;
+
+ /* gather infos on the stream this packet is part of */
+ COPY_ADDRESS(&(tmp_strinfo.src_addr), &(pinfo->src));
+ tmp_strinfo.src_port = pinfo->srcport;
+ COPY_ADDRESS(&(tmp_strinfo.dest_addr), &(pinfo->dst));
+ tmp_strinfo.dest_port = pinfo->destport;
+ tmp_strinfo.ssrc = rtpinfo->info_sync_src;
+ tmp_strinfo.pt = rtpinfo->info_payload_type;
+ tmp_strinfo.info_payload_type_str = rtpinfo->info_payload_type_str;
+
+ if (tapinfo->mode == TAP_ANALYSE) {
+ /* check wether we already have a stream with these parameters in the list */
+ list = g_list_first(tapinfo->strinfo_list);
+ while (list)
+ {
+ if (rtp_stream_info_cmp(&tmp_strinfo, (rtp_stream_info_t*)(list->data))==0)
+ {
+ strinfo = (rtp_stream_info_t*)(list->data); /*found!*/
+ break;
+ }
+ list = g_list_next(list);
+ }
+
+ /* not in the list? then create a new entry */
+ if (!strinfo) {
+ tmp_strinfo.npackets = 0;
+ tmp_strinfo.first_frame_num = pinfo->fd->num;
+ tmp_strinfo.start_sec = (guint32) pinfo->fd->abs_ts.secs;
+ tmp_strinfo.start_usec = pinfo->fd->abs_ts.nsecs/1000;
+ tmp_strinfo.start_rel_sec = (guint32) pinfo->fd->rel_ts.secs;
+ tmp_strinfo.start_rel_usec = pinfo->fd->rel_ts.nsecs/1000;
+ tmp_strinfo.tag_vlan_error = 0;
+ tmp_strinfo.tag_diffserv_error = 0;
+ tmp_strinfo.vlan_id = 0;
+ tmp_strinfo.problem = FALSE;
+
+ /* reset RTP stats */
+ tmp_strinfo.rtp_stats.first_packet = TRUE;
+ tmp_strinfo.rtp_stats.max_delta = 0;
+ tmp_strinfo.rtp_stats.max_jitter = 0;
+ tmp_strinfo.rtp_stats.mean_jitter = 0;
+ tmp_strinfo.rtp_stats.delta = 0;
+ tmp_strinfo.rtp_stats.diff = 0;
+ tmp_strinfo.rtp_stats.jitter = 0;
+ tmp_strinfo.rtp_stats.bandwidth = 0;
+ tmp_strinfo.rtp_stats.total_bytes = 0;
+ tmp_strinfo.rtp_stats.bw_start_index = 0;
+ tmp_strinfo.rtp_stats.bw_index = 0;
+ tmp_strinfo.rtp_stats.timestamp = 0;
+ tmp_strinfo.rtp_stats.max_nr = 0;
+ tmp_strinfo.rtp_stats.total_nr = 0;
+ tmp_strinfo.rtp_stats.sequence = 0;
+ tmp_strinfo.rtp_stats.start_seq_nr = 0;
+ tmp_strinfo.rtp_stats.stop_seq_nr = 0;
+ tmp_strinfo.rtp_stats.cycles = 0;
+ tmp_strinfo.rtp_stats.under = FALSE;
+ tmp_strinfo.rtp_stats.start_time = 0;
+ tmp_strinfo.rtp_stats.time = 0;
+ tmp_strinfo.rtp_stats.reg_pt = PT_UNDEFINED;
+
+ /* Get the Setup frame number who set this RTP stream */
+ p_conv_data = p_get_proto_data(pinfo->fd, proto_get_id_by_filter_name("rtp"));
+ if (p_conv_data)
+ tmp_strinfo.setup_frame_number = p_conv_data->frame_number;
+ else
+ tmp_strinfo.setup_frame_number = 0xFFFFFFFF;
+
+ strinfo = g_malloc(sizeof(rtp_stream_info_t));
+ *strinfo = tmp_strinfo; /* memberwise copy of struct */
+ tapinfo->strinfo_list = g_list_append(tapinfo->strinfo_list, strinfo);
+ }
+
+ /* get RTP stats for the packet */
+ rtp_packet_analyse(&(strinfo->rtp_stats), pinfo, rtpinfo);
+ if (strinfo->rtp_stats.flags & STAT_FLAG_WRONG_TIMESTAMP
+ || strinfo->rtp_stats.flags & STAT_FLAG_WRONG_SEQ)
+ strinfo->problem = TRUE;
+
+
+ /* increment the packets counter for this stream */
+ ++(strinfo->npackets);
+ strinfo->stop_rel_sec = (guint32) pinfo->fd->rel_ts.secs;
+ strinfo->stop_rel_usec = pinfo->fd->rel_ts.nsecs/1000;
+
+ /* increment the packets counter of all streams */
+ ++(tapinfo->npackets);
+
+ return 1; /* refresh output */
+ }
+ else if (tapinfo->mode == TAP_SAVE) {
+ if (rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_fwd)==0) {
+ /* XXX - what if rtpinfo->info_all_data_present is
+ FALSE, so that we don't *have* all the data? */
+ sample.header.rec_time =
+ (pinfo->fd->abs_ts.nsecs/1000 + 1000000 - tapinfo->filter_stream_fwd->start_usec)/1000
+ + (guint32) (pinfo->fd->abs_ts.secs - tapinfo->filter_stream_fwd->start_sec - 1)*1000;
+ sample.header.frame_length = rtpinfo->info_data_len;
+ sample.frame = rtpinfo->info_data;
+ rtp_write_sample(&sample, tapinfo->save_file);
+ }
+ }
+ else if (tapinfo->mode == TAP_MARK) {
+
+ if (rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_fwd)==0
+ || rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_rev)==0)
+ {
+ cf_mark_frame(&cfile, pinfo->fd);
+ }
+ }
+
+ return 0;
+}
+
+
+typedef struct _key_value {
+ guint32 key;
+ guint32 value;
+} key_value;
+
+
+/* RTP sampling clock rates for fixed payload types as defined in
+ http://www.iana.org/assignments/rtp-parameters */
+static const key_value clock_map[] = {
+ {PT_PCMU, 8000},
+ {PT_1016, 8000},
+ {PT_G721, 8000},
+ {PT_GSM, 8000},
+ {PT_G723, 8000},
+ {PT_DVI4_8000, 8000},
+ {PT_DVI4_16000, 16000},
+ {PT_LPC, 8000},
+ {PT_PCMA, 8000},
+ {PT_G722, 8000},
+ {PT_L16_STEREO, 44100},
+ {PT_L16_MONO, 44100},
+ {PT_QCELP, 8000},
+ {PT_CN, 8000},
+ {PT_MPA, 90000},
+ {PT_G728, 8000},
+ {PT_G728, 8000},
+ {PT_DVI4_11025, 11025},
+ {PT_DVI4_22050, 22050},
+ {PT_G729, 8000},
+ {PT_CN_OLD, 8000},
+ {PT_CELB, 90000},
+ {PT_JPEG, 90000},
+ {PT_NV, 90000},
+ {PT_H261, 90000},
+ {PT_MPV, 90000},
+ {PT_MP2T, 90000},
+ {PT_H263, 90000},
+};
+
+#define NUM_CLOCK_VALUES (sizeof clock_map / sizeof clock_map[0])
+
+static guint32
+get_clock_rate(guint32 key)
+{
+ size_t i;
+
+ for (i = 0; i < NUM_CLOCK_VALUES; i++) {
+ if (clock_map[i].key == key)
+ return clock_map[i].value;
+ }
+ return 1;
+}
+
+typedef struct _mimetype_and_clock {
+ const gchar *pt_mime_name_str;
+ guint32 value;
+} mimetype_and_clock;
+/* RTP sampling clock rates for
+ "In addition to the RTP payload formats (encodings) listed in the RTP
+ Payload Types table, there are additional payload formats that do not
+ have static RTP payload types assigned but instead use dynamic payload
+ type number assignment. Each payload format is named by a registered
+ MIME subtype"
+ http://www.iana.org/assignments/rtp-parameters.
+*/
+static const mimetype_and_clock mimetype_and_clock_map[] = {
+ {"AMR", 8000}, /* [RFC3267] */
+ {"AMR-WB", 16000}, /* [RFC3267] */
+ {"EVRC", 8000}, /* [RFC3558] */
+ {"EVRC0", 8000}, /* [RFC3558] */
+ {"G7221", 16000}, /* [RFC3047] */
+ {"G726-16", 8000}, /* [RFC3551] */
+ {"G726-24", 8000}, /* [RFC3551] */
+ {"G726-32", 8000}, /* [RFC3551] */
+ {"G726-40", 8000}, /* [RFC3551] */
+ {"G729D", 8000}, /* [RFC3551] */
+ {"G729E", 8000}, /* [RFC3551] */
+ {"GSM-EFR", 8000}, /* [RFC3551] */
+ {"mpa-robust", 90000}, /* [RFC3119] */
+ {"SMV", 8000}, /* [RFC3558] */
+ {"SMV0", 8000}, /* [RFC3558] */
+ {"red", 1000}, /* [RFC4102] */
+ {"t140", 1000}, /* [RFC4103] */
+ {"BMPEG", 90000}, /* [RFC2343],[RFC3555] */
+ {"BT656", 90000}, /* [RFC2431],[RFC3555] */
+ {"DV", 90000}, /* [RFC3189] */
+ {"H263-1998", 90000}, /* [RFC2429],[RFC3555] */
+ {"H263-2000", 90000}, /* [RFC2429],[RFC3555] */
+ {"MP1S", 90000}, /* [RFC2250],[RFC3555] */
+ {"MP2P", 90000}, /* [RFC2250],[RFC3555] */
+ {"MP4V-ES", 90000}, /* [RFC3016] */
+ {"pointer", 90000}, /* [RFC2862] */
+ {"raw", 90000}, /* [RFC4175] */
+ {"telephone-event", 8000}, /* [RFC4733] */
+};
+
+#define NUM_DYN_CLOCK_VALUES (sizeof mimetype_and_clock_map / sizeof mimetype_and_clock_map[0])
+
+static guint32
+get_dyn_pt_clock_rate(gchar *payload_type_str)
+{
+ size_t i;
+
+ for (i = 0; i < NUM_DYN_CLOCK_VALUES; i++) {
+ if (g_ascii_strncasecmp(mimetype_and_clock_map[i].pt_mime_name_str,payload_type_str,(strlen(mimetype_and_clock_map[i].pt_mime_name_str))) == 0)
+ return mimetype_and_clock_map[i].value;
+ }
+
+ return 1;
+}
+
+/****************************************************************************/
+int rtp_packet_analyse(tap_rtp_stat_t *statinfo,
+ packet_info *pinfo,
+ const struct _rtp_info *rtpinfo)
+{
+ double current_time;
+ double current_jitter;
+ double current_diff;
+ guint32 clock_rate;
+
+ statinfo->flags = 0;
+ /* check payload type */
+ if (rtpinfo->info_payload_type == PT_CN
+ || rtpinfo->info_payload_type == PT_CN_OLD)
+ statinfo->flags |= STAT_FLAG_PT_CN;
+ if (statinfo->pt == PT_CN
+ || statinfo->pt == PT_CN_OLD)
+ statinfo->flags |= STAT_FLAG_FOLLOW_PT_CN;
+ if (rtpinfo->info_payload_type != statinfo->pt)
+ statinfo->flags |= STAT_FLAG_PT_CHANGE;
+ statinfo->pt = rtpinfo->info_payload_type;
+ /*
+ * XXX - should "get_clock_rate()" return 0 for unknown
+ * payload types, presumably meaning that we should
+ * just ignore this packet?
+ */
+ if (statinfo->pt < 96 ){
+ clock_rate = get_clock_rate(statinfo->pt);
+ }else{ /* dynamic PT */
+ if ( rtpinfo->info_payload_type_str != NULL )
+ clock_rate = get_dyn_pt_clock_rate(rtpinfo-> info_payload_type_str);
+ else
+ clock_rate = 1;
+ }
+
+ /* store the current time and calculate the current jitter */
+ current_time = nstime_to_sec(&pinfo->fd->rel_ts);
+ current_diff = fabs (current_time - (statinfo->time) - ((double)(rtpinfo->info_timestamp)-(double)(statinfo->timestamp))/clock_rate);
+ current_jitter = statinfo->jitter + ( current_diff - statinfo->jitter)/16;
+ statinfo->delta = current_time-(statinfo->time);
+ statinfo->jitter = current_jitter;
+ statinfo->diff = current_diff;
+
+ /* calculate the BW in Kbps adding the IP+UDP header to the RTP -> 20bytes(IP)+8bytes(UDP) = 28bytes */
+ statinfo->bw_history[statinfo->bw_index].bytes = rtpinfo->info_data_len + 28;
+ statinfo->bw_history[statinfo->bw_index].time = current_time;
+ /* check if there are more than 1sec in the history buffer to calculate BW in bps. If so, remove those for the calculation */
+ while ((statinfo->bw_history[statinfo->bw_start_index].time+1)<current_time){
+ statinfo->total_bytes -= statinfo->bw_history[statinfo->bw_start_index].bytes;
+ statinfo->bw_start_index++;
+ if (statinfo->bw_start_index == BUFF_BW) statinfo->bw_start_index=0;
+ };
+ statinfo->total_bytes += rtpinfo->info_data_len + 28;
+ statinfo->bandwidth = (double)(statinfo->total_bytes*8)/1000;
+ statinfo->bw_index++;
+ if (statinfo->bw_index == BUFF_BW) statinfo->bw_index = 0;
+
+
+ /* is this the first packet we got in this direction? */
+ if (statinfo->first_packet) {
+ statinfo->start_seq_nr = rtpinfo->info_seq_num;
+ statinfo->start_time = current_time;
+ statinfo->delta = 0;
+ statinfo->jitter = 0;
+ statinfo->diff = 0;
+ statinfo->flags |= STAT_FLAG_FIRST;
+ statinfo->first_packet = FALSE;
+ }
+ /* is it a packet with the mark bit set? */
+ if (rtpinfo->info_marker_set) {
+ statinfo->delta_timestamp = rtpinfo->info_timestamp - statinfo->timestamp;
+ if (rtpinfo->info_timestamp > statinfo->timestamp){
+ statinfo->flags |= STAT_FLAG_MARKER;
+ }
+ else{
+ statinfo->flags |= STAT_FLAG_WRONG_TIMESTAMP;
+ }
+ }
+ /* is it a regular packet? */
+ if (!(statinfo->flags & STAT_FLAG_FIRST)
+ && !(statinfo->flags & STAT_FLAG_MARKER)
+ && !(statinfo->flags & STAT_FLAG_PT_CN)
+ && !(statinfo->flags & STAT_FLAG_WRONG_TIMESTAMP)
+ && !(statinfo->flags & STAT_FLAG_FOLLOW_PT_CN)) {
+ /* include it in maximum delta calculation */
+ if (statinfo->delta > statinfo->max_delta) {
+ statinfo->max_delta = statinfo->delta;
+ statinfo->max_nr = pinfo->fd->num;
+ }
+ /* maximum and mean jitter calculation */
+ if (statinfo->jitter > statinfo->max_jitter) {
+ statinfo->max_jitter = statinfo->jitter;
+ }
+ statinfo->mean_jitter = (statinfo->mean_jitter*statinfo->total_nr + current_diff) / (statinfo->total_nr+1);
+ }
+ /* regular payload change? (CN ignored) */
+ if (!(statinfo->flags & STAT_FLAG_FIRST)
+ && !(statinfo->flags & STAT_FLAG_PT_CN)) {
+ if ((statinfo->pt != statinfo->reg_pt)
+ && (statinfo->reg_pt != PT_UNDEFINED)) {
+ statinfo->flags |= STAT_FLAG_REG_PT_CHANGE;
+ }
+ }
+
+ /* set regular payload*/
+ if (!(statinfo->flags & STAT_FLAG_PT_CN)) {
+ statinfo->reg_pt = statinfo->pt;
+ }
+
+
+ /* When calculating expected rtp packets the seq number can wrap around
+ * so we have to count the number of cycles
+ * Variable cycles counts the wraps around in forwarding connection and
+ * under is flag that indicates where we are
+ *
+ * XXX how to determine number of cycles with all possible lost, late
+ * and duplicated packets without any doubt? It seems to me, that
+ * because of all possible combination of late, duplicated or lost
+ * packets, this can only be more or less good approximation
+ *
+ * There are some combinations (rare but theoretically possible),
+ * where below code won't work correctly - statistic may be wrong then.
+ */
+
+ /* so if the current sequence number is less than the start one
+ * we assume, that there is another cycle running */
+ if ((rtpinfo->info_seq_num < statinfo->start_seq_nr) && (statinfo->under == FALSE)){
+ statinfo->cycles++;
+ statinfo->under = TRUE;
+ }
+ /* what if the start seq nr was 0? Then the above condition will never
+ * be true, so we add another condition. XXX The problem would arise
+ * if one of the packets with seq nr 0 or 65535 would be lost or late */
+ else if ((rtpinfo->info_seq_num == 0) && (statinfo->stop_seq_nr == 65535) &&
+ (statinfo->under == FALSE)){
+ statinfo->cycles++;
+ statinfo->under = TRUE;
+ }
+ /* the whole round is over, so reset the flag */
+ else if ((rtpinfo->info_seq_num > statinfo->start_seq_nr) && (statinfo->under != FALSE)) {
+ statinfo->under = FALSE;
+ }
+
+ /* Since it is difficult to count lost, duplicate or late packets separately,
+ * we would like to know at least how many times the sequence number was not ok */
+
+ /* if the current seq number equals the last one or if we are here for
+ * the first time, then it is ok, we just store the current one as the last one */
+ if ( (statinfo->seq_num+1 == rtpinfo->info_seq_num) || (statinfo->flags & STAT_FLAG_FIRST) )
+ statinfo->seq_num = rtpinfo->info_seq_num;
+ /* if the first one is 65535. XXX same problem as above: if seq 65535 or 0 is lost... */
+ else if ( (statinfo->seq_num == 65535) && (rtpinfo->info_seq_num == 0) )
+ statinfo->seq_num = rtpinfo->info_seq_num;
+ /* lost packets */
+ else if (statinfo->seq_num+1 < rtpinfo->info_seq_num) {
+ statinfo->seq_num = rtpinfo->info_seq_num;
+ statinfo->sequence++;
+ statinfo->flags |= STAT_FLAG_WRONG_SEQ;
+ }
+ /* late or duplicated */
+ else if (statinfo->seq_num+1 > rtpinfo->info_seq_num) {
+ statinfo->sequence++;
+ statinfo->flags |= STAT_FLAG_WRONG_SEQ;
+ }
+ statinfo->time = current_time;
+ statinfo->timestamp = rtpinfo->info_timestamp;
+ statinfo->stop_seq_nr = rtpinfo->info_seq_num;
+ statinfo->total_nr++;
+
+ return 0;
+}
+
+
git.samba.org / obnox / wireshark / wip.git / commitdiff