forked from osmocom/wireshark
Add the missing files from Balint Reczey's patch for bug 2233.
svn path=/trunk/; revision=24253
This commit is contained in:
parent
c8276a5c91
commit
0ed6971273
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@ -0,0 +1,592 @@
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/* tap-rtp-common.c
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* RTP stream handler functions used by tshark and wireshark
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*
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* $Id$
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*
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* Copyright 2008, Ericsson AB
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* By Balint Reczey <balint.reczey@ericsson.com>
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*
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* most functions are copied from gtk/rtp_stream.c and gtk/rtp_analisys.c
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* Copyright 2003, Alcatel Business Systems
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* By Lars Ruoff <lars.ruoff@gmx.net>
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*
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* Wireshark - Network traffic analyzer
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* By Gerald Combs <gerald@wireshark.org>
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* Copyright 1998 Gerald Combs
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "globals.h"
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#include <epan/tap.h>
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#include "register.h"
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#include <string.h>
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#include <epan/rtp_pt.h>
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#include <epan/addr_resolv.h>
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#include <epan/dissectors/packet-rtp.h>
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#include "gtk/rtp_stream.h"
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#include "tap-rtp-common.h"
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/****************************************************************************/
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/* GCompareFunc style comparison function for _rtp_stream_info */
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gint rtp_stream_info_cmp(gconstpointer aa, gconstpointer bb)
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{
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const struct _rtp_stream_info* a = aa;
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const struct _rtp_stream_info* b = bb;
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if (a==b)
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return 0;
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if (a==NULL || b==NULL)
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return 1;
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if (ADDRESSES_EQUAL(&(a->src_addr), &(b->src_addr))
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&& (a->src_port == b->src_port)
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&& ADDRESSES_EQUAL(&(a->dest_addr), &(b->dest_addr))
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&& (a->dest_port == b->dest_port)
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&& (a->ssrc == b->ssrc))
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return 0;
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else
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return 1;
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}
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/****************************************************************************/
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/* when there is a [re]reading of packet's */
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void rtpstream_reset(rtpstream_tapinfo_t *tapinfo)
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{
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GList* list;
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if (tapinfo->mode == TAP_ANALYSE) {
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/* free the data items first */
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list = g_list_first(tapinfo->strinfo_list);
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while (list)
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{
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g_free(list->data);
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list = g_list_next(list);
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}
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g_list_free(tapinfo->strinfo_list);
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tapinfo->strinfo_list = NULL;
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tapinfo->nstreams = 0;
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tapinfo->npackets = 0;
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}
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++(tapinfo->launch_count);
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return;
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}
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void rtpstream_reset_cb(void *arg)
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{
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rtpstream_reset(arg);
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}
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/*
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* rtpdump file format
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*
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* The file starts with the tool to be used for playing this file,
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* the multicast/unicast receive address and the port.
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*
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* #!rtpplay1.0 224.2.0.1/3456\n
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*
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* This is followed by one binary header (RD_hdr_t) and one RD_packet_t
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* structure for each received packet. All fields are in network byte
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* order. We don't need the source IP address since we can do mapping
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* based on SSRC. This saves (a little) space, avoids non-IPv4
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* problems and privacy/security concerns. The header is followed by
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* the RTP/RTCP header and (optionally) the actual payload.
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*/
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#define RTPFILE_VERSION "1.0"
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/*
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* Write a header to the current output file.
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* The header consists of an identifying string, followed
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* by a binary structure.
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*/
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void rtp_write_header(rtp_stream_info_t *strinfo, FILE *file)
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{
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guint32 start_sec; /* start of recording (GMT) (seconds) */
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guint32 start_usec; /* start of recording (GMT) (microseconds)*/
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guint32 source; /* network source (multicast address) */
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size_t sourcelen;
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guint16 port; /* UDP port */
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guint16 padding; /* 2 padding bytes */
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fprintf(file, "#!rtpplay%s %s/%u\n", RTPFILE_VERSION,
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get_addr_name(&(strinfo->dest_addr)),
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strinfo->dest_port);
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start_sec = g_htonl(strinfo->start_sec);
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start_usec = g_htonl(strinfo->start_usec);
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/* rtpdump only accepts guint32 as source, will be fake for IPv6 */
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memset(&source, 0, sizeof source);
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sourcelen = strinfo->src_addr.len;
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if (sourcelen > sizeof source)
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sourcelen = sizeof source;
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memcpy(&source, strinfo->src_addr.data, sourcelen);
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port = g_htons(strinfo->src_port);
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padding = 0;
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if (fwrite(&start_sec, 4, 1, file) == 0)
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return;
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if (fwrite(&start_usec, 4, 1, file) == 0)
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return;
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if (fwrite(&source, 4, 1, file) == 0)
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return;
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if (fwrite(&port, 2, 1, file) == 0)
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return;
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if (fwrite(&padding, 2, 1, file) == 0)
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return;
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}
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/* utility function for writing a sample to file in rtpdump -F dump format (.rtp)*/
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void rtp_write_sample(rtp_sample_t* sample, FILE* file)
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{
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guint16 length; /* length of packet, including this header (may
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be smaller than plen if not whole packet recorded) */
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guint16 plen; /* actual header+payload length for RTP, 0 for RTCP */
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guint32 offset; /* milliseconds since the start of recording */
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length = g_htons(sample->header.frame_length + 8);
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plen = g_htons(sample->header.frame_length);
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offset = g_htonl(sample->header.rec_time);
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if (fwrite(&length, 2, 1, file) == 0)
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return;
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if (fwrite(&plen, 2, 1, file) == 0)
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return;
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if (fwrite(&offset, 4, 1, file) == 0)
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return;
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if (fwrite(sample->frame, sample->header.frame_length, 1, file) == 0)
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return;
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}
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/****************************************************************************/
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/* whenever a RTP packet is seen by the tap listener */
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int rtpstream_packet(void *arg, packet_info *pinfo, epan_dissect_t *edt _U_, const void *arg2)
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{
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rtpstream_tapinfo_t *tapinfo = arg;
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const struct _rtp_info *rtpinfo = arg2;
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rtp_stream_info_t tmp_strinfo;
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rtp_stream_info_t *strinfo = NULL;
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GList* list;
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rtp_sample_t sample;
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struct _rtp_conversation_info *p_conv_data = NULL;
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/* gather infos on the stream this packet is part of */
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COPY_ADDRESS(&(tmp_strinfo.src_addr), &(pinfo->src));
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tmp_strinfo.src_port = pinfo->srcport;
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COPY_ADDRESS(&(tmp_strinfo.dest_addr), &(pinfo->dst));
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tmp_strinfo.dest_port = pinfo->destport;
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tmp_strinfo.ssrc = rtpinfo->info_sync_src;
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tmp_strinfo.pt = rtpinfo->info_payload_type;
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tmp_strinfo.info_payload_type_str = rtpinfo->info_payload_type_str;
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if (tapinfo->mode == TAP_ANALYSE) {
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/* check wether we already have a stream with these parameters in the list */
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list = g_list_first(tapinfo->strinfo_list);
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while (list)
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{
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if (rtp_stream_info_cmp(&tmp_strinfo, (rtp_stream_info_t*)(list->data))==0)
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{
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strinfo = (rtp_stream_info_t*)(list->data); /*found!*/
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break;
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}
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list = g_list_next(list);
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}
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/* not in the list? then create a new entry */
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if (!strinfo) {
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tmp_strinfo.npackets = 0;
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tmp_strinfo.first_frame_num = pinfo->fd->num;
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tmp_strinfo.start_sec = (guint32) pinfo->fd->abs_ts.secs;
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tmp_strinfo.start_usec = pinfo->fd->abs_ts.nsecs/1000;
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tmp_strinfo.start_rel_sec = (guint32) pinfo->fd->rel_ts.secs;
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tmp_strinfo.start_rel_usec = pinfo->fd->rel_ts.nsecs/1000;
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tmp_strinfo.tag_vlan_error = 0;
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tmp_strinfo.tag_diffserv_error = 0;
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tmp_strinfo.vlan_id = 0;
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tmp_strinfo.problem = FALSE;
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/* reset RTP stats */
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tmp_strinfo.rtp_stats.first_packet = TRUE;
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tmp_strinfo.rtp_stats.max_delta = 0;
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tmp_strinfo.rtp_stats.max_jitter = 0;
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tmp_strinfo.rtp_stats.mean_jitter = 0;
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tmp_strinfo.rtp_stats.delta = 0;
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tmp_strinfo.rtp_stats.diff = 0;
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tmp_strinfo.rtp_stats.jitter = 0;
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tmp_strinfo.rtp_stats.bandwidth = 0;
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tmp_strinfo.rtp_stats.total_bytes = 0;
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tmp_strinfo.rtp_stats.bw_start_index = 0;
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tmp_strinfo.rtp_stats.bw_index = 0;
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tmp_strinfo.rtp_stats.timestamp = 0;
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tmp_strinfo.rtp_stats.max_nr = 0;
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tmp_strinfo.rtp_stats.total_nr = 0;
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tmp_strinfo.rtp_stats.sequence = 0;
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tmp_strinfo.rtp_stats.start_seq_nr = 0;
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tmp_strinfo.rtp_stats.stop_seq_nr = 0;
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tmp_strinfo.rtp_stats.cycles = 0;
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tmp_strinfo.rtp_stats.under = FALSE;
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tmp_strinfo.rtp_stats.start_time = 0;
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tmp_strinfo.rtp_stats.time = 0;
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tmp_strinfo.rtp_stats.reg_pt = PT_UNDEFINED;
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/* Get the Setup frame number who set this RTP stream */
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p_conv_data = p_get_proto_data(pinfo->fd, proto_get_id_by_filter_name("rtp"));
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if (p_conv_data)
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tmp_strinfo.setup_frame_number = p_conv_data->frame_number;
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else
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tmp_strinfo.setup_frame_number = 0xFFFFFFFF;
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strinfo = g_malloc(sizeof(rtp_stream_info_t));
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*strinfo = tmp_strinfo; /* memberwise copy of struct */
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tapinfo->strinfo_list = g_list_append(tapinfo->strinfo_list, strinfo);
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}
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/* get RTP stats for the packet */
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rtp_packet_analyse(&(strinfo->rtp_stats), pinfo, rtpinfo);
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if (strinfo->rtp_stats.flags & STAT_FLAG_WRONG_TIMESTAMP
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|| strinfo->rtp_stats.flags & STAT_FLAG_WRONG_SEQ)
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strinfo->problem = TRUE;
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/* increment the packets counter for this stream */
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++(strinfo->npackets);
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strinfo->stop_rel_sec = (guint32) pinfo->fd->rel_ts.secs;
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strinfo->stop_rel_usec = pinfo->fd->rel_ts.nsecs/1000;
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/* increment the packets counter of all streams */
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++(tapinfo->npackets);
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return 1; /* refresh output */
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}
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else if (tapinfo->mode == TAP_SAVE) {
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if (rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_fwd)==0) {
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/* XXX - what if rtpinfo->info_all_data_present is
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FALSE, so that we don't *have* all the data? */
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sample.header.rec_time =
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(pinfo->fd->abs_ts.nsecs/1000 + 1000000 - tapinfo->filter_stream_fwd->start_usec)/1000
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+ (guint32) (pinfo->fd->abs_ts.secs - tapinfo->filter_stream_fwd->start_sec - 1)*1000;
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sample.header.frame_length = rtpinfo->info_data_len;
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sample.frame = rtpinfo->info_data;
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rtp_write_sample(&sample, tapinfo->save_file);
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}
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}
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else if (tapinfo->mode == TAP_MARK) {
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if (rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_fwd)==0
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|| rtp_stream_info_cmp(&tmp_strinfo, tapinfo->filter_stream_rev)==0)
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{
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cf_mark_frame(&cfile, pinfo->fd);
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}
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}
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return 0;
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}
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typedef struct _key_value {
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guint32 key;
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guint32 value;
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} key_value;
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/* RTP sampling clock rates for fixed payload types as defined in
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http://www.iana.org/assignments/rtp-parameters */
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static const key_value clock_map[] = {
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{PT_PCMU, 8000},
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{PT_1016, 8000},
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{PT_G721, 8000},
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{PT_GSM, 8000},
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{PT_G723, 8000},
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{PT_DVI4_8000, 8000},
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{PT_DVI4_16000, 16000},
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{PT_LPC, 8000},
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{PT_PCMA, 8000},
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{PT_G722, 8000},
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{PT_L16_STEREO, 44100},
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{PT_L16_MONO, 44100},
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{PT_QCELP, 8000},
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{PT_CN, 8000},
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{PT_MPA, 90000},
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{PT_G728, 8000},
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{PT_G728, 8000},
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{PT_DVI4_11025, 11025},
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{PT_DVI4_22050, 22050},
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{PT_G729, 8000},
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{PT_CN_OLD, 8000},
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{PT_CELB, 90000},
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{PT_JPEG, 90000},
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{PT_NV, 90000},
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{PT_H261, 90000},
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{PT_MPV, 90000},
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{PT_MP2T, 90000},
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{PT_H263, 90000},
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};
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#define NUM_CLOCK_VALUES (sizeof clock_map / sizeof clock_map[0])
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static guint32
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get_clock_rate(guint32 key)
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{
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size_t i;
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for (i = 0; i < NUM_CLOCK_VALUES; i++) {
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if (clock_map[i].key == key)
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return clock_map[i].value;
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}
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return 1;
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}
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typedef struct _mimetype_and_clock {
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const gchar *pt_mime_name_str;
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guint32 value;
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} mimetype_and_clock;
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/* RTP sampling clock rates for
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"In addition to the RTP payload formats (encodings) listed in the RTP
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Payload Types table, there are additional payload formats that do not
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have static RTP payload types assigned but instead use dynamic payload
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type number assignment. Each payload format is named by a registered
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MIME subtype"
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http://www.iana.org/assignments/rtp-parameters.
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*/
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static const mimetype_and_clock mimetype_and_clock_map[] = {
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{"AMR", 8000}, /* [RFC3267] */
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{"AMR-WB", 16000}, /* [RFC3267] */
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{"EVRC", 8000}, /* [RFC3558] */
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{"EVRC0", 8000}, /* [RFC3558] */
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{"G7221", 16000}, /* [RFC3047] */
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{"G726-16", 8000}, /* [RFC3551] */
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{"G726-24", 8000}, /* [RFC3551] */
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{"G726-32", 8000}, /* [RFC3551] */
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{"G726-40", 8000}, /* [RFC3551] */
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{"G729D", 8000}, /* [RFC3551] */
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{"G729E", 8000}, /* [RFC3551] */
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{"GSM-EFR", 8000}, /* [RFC3551] */
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{"mpa-robust", 90000}, /* [RFC3119] */
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{"SMV", 8000}, /* [RFC3558] */
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{"SMV0", 8000}, /* [RFC3558] */
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{"red", 1000}, /* [RFC4102] */
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{"t140", 1000}, /* [RFC4103] */
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{"BMPEG", 90000}, /* [RFC2343],[RFC3555] */
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{"BT656", 90000}, /* [RFC2431],[RFC3555] */
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{"DV", 90000}, /* [RFC3189] */
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{"H263-1998", 90000}, /* [RFC2429],[RFC3555] */
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{"H263-2000", 90000}, /* [RFC2429],[RFC3555] */
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{"MP1S", 90000}, /* [RFC2250],[RFC3555] */
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{"MP2P", 90000}, /* [RFC2250],[RFC3555] */
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{"MP4V-ES", 90000}, /* [RFC3016] */
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{"pointer", 90000}, /* [RFC2862] */
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{"raw", 90000}, /* [RFC4175] */
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{"telephone-event", 8000}, /* [RFC4733] */
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};
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#define NUM_DYN_CLOCK_VALUES (sizeof mimetype_and_clock_map / sizeof mimetype_and_clock_map[0])
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static guint32
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get_dyn_pt_clock_rate(gchar *payload_type_str)
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{
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size_t i;
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for (i = 0; i < NUM_DYN_CLOCK_VALUES; i++) {
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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)
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return mimetype_and_clock_map[i].value;
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}
|
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return 1;
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}
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/****************************************************************************/
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int rtp_packet_analyse(tap_rtp_stat_t *statinfo,
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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;
|
||||
}
|
||||
|
||||
|
|
@ -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 <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.
|
||||
*/
|
||||
|
||||
#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*/
|
Loading…
Reference in New Issue