From 0ed69712733a236c115778593fc89a999ed734ed Mon Sep 17 00:00:00 2001 From: Guy Harris Date: Sun, 3 Feb 2008 12:36:37 +0000 Subject: [PATCH] Add the missing files from Balint Reczey's patch for bug 2233. svn path=/trunk/; revision=24253 --- tap-rtp-common.c | 592 +++++++++++++++++++++++++++++++++++++++++++++++ tap-rtp-common.h | 49 ++++ 2 files changed, 641 insertions(+) create mode 100644 tap-rtp-common.c create mode 100644 tap-rtp-common.h diff --git a/tap-rtp-common.c b/tap-rtp-common.c new file mode 100644 index 0000000000..76e8163583 --- /dev/null +++ b/tap-rtp-common.c @@ -0,0 +1,592 @@ +/* tap-rtp-common.c + * RTP stream handler functions used by tshark and wireshark + * + * $Id$ + * + * Copyright 2008, Ericsson AB + * By Balint Reczey + * + * most functions are copied from gtk/rtp_stream.c and gtk/rtp_analisys.c + * Copyright 2003, Alcatel Business Systems + * By Lars Ruoff + * + * Wireshark - Network traffic analyzer + * By Gerald Combs + * 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 +#include "register.h" +#include +#include +#include +#include +#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)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; +} + + diff --git a/tap-rtp-common.h b/tap-rtp-common.h new file mode 100644 index 0000000000..b7b07a8bfe --- /dev/null +++ b/tap-rtp-common.h @@ -0,0 +1,49 @@ +/* tap-rtp-common.h + * RTP streams handler functions used by tshark and wireshark + * + * $Id$ + * + * Copyright 2008, Ericsson AB + * By Balint Reczey + * + * most functions are copied from gtk/rtp_stream.c and gtk/rtp_analisys.c + * Copyright 2003, Alcatel Business Systems + * By Lars Ruoff + * + * Wireshark - Network traffic analyzer + * By Gerald Combs + * 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. + */ + +#ifndef TAP_RTP_COMMON_H_INCLUDED +#define TAP_RTP_COMMON_H_INCLUDED + +#include "gtk/rtp_stream.h" + +gint rtp_stream_info_cmp(gconstpointer, gconstpointer); +void rtpstream_reset_cb(void*); +void rtp_write_header(rtp_stream_info_t*, FILE*); +void rtp_write_sample(rtp_sample_t*, FILE*); +int rtpstream_packet(void*, packet_info*, epan_dissect_t *, const void *); + +/* The one and only global rtpstream_tapinfo_t structure for tshark and wireshark. + */ +static rtpstream_tapinfo_t the_tapinfo_struct = + {0, NULL, 0, TAP_ANALYSE, NULL, NULL, NULL, 0, FALSE}; + + +#endif /*TAP_RTP_COMMON_H_INCLUDED*/