![AndersBroman](/assets/img/avatar_default.png)
detection) Bug: 11478 Change-Id: Ieff015e5bf55facc732beae6a40b76d2050a4aac Reviewed-on: https://code.wireshark.org/review/20696 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Anders Broman <a.broman58@gmail.com>
737 lines
25 KiB
C
737 lines
25 KiB
C
/* 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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* 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 ui/gtk/rtp_stream.c and ui/gtk/rtp_analysis.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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include <glib.h>
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#include <math.h>
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#include "globals.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/proto_data.h>
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#include <epan/dissectors/packet-rtp.h>
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#include <wsutil/pint.h>
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#include "rtp_stream.h"
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#include "tap-rtp-common.h"
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/* XXX: are changes needed to properly handle situations where
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info_all_data_present == FALSE ?
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E.G., when captured frames are truncated.
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*/
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/****************************************************************************/
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/* Type for storing and writing rtpdump information */
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typedef struct st_rtpdump_info {
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double rec_time; /**< milliseconds since start of recording */
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guint16 num_samples; /**< number of bytes in *frame */
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const guint8 *samples; /**< data bytes */
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} rtpdump_info_t;
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/****************************************************************************/
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/* GCompareFunc style comparison function for _rtp_stream_info */
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static gint rtp_stream_info_cmp(gconstpointer aa, gconstpointer bb)
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{
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const struct _rtp_stream_info* a = (const struct _rtp_stream_info*)aa;
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const struct _rtp_stream_info* b = (const struct _rtp_stream_info*)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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return;
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}
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void rtpstream_reset_cb(void *arg)
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{
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rtpstream_tapinfo_t *ti =(rtpstream_tapinfo_t *)arg;
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if (ti->tap_reset) {
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/* Give listeners a chance to cleanup references. */
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ti->tap_reset(ti);
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}
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rtpstream_reset(ti);
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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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char* addr_str = address_to_display(NULL, &(strinfo->dest_addr));
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fprintf(file, "#!rtpplay%s %s/%u\n", RTPFILE_VERSION,
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addr_str,
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strinfo->dest_port);
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wmem_free(NULL, addr_str);
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start_sec = g_htonl(strinfo->start_fd->abs_ts.secs);
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start_usec = g_htonl(strinfo->start_fd->abs_ts.nsecs / 1000000);
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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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static void rtp_write_sample(rtpdump_info_t* rtpdump_info, 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(rtpdump_info->num_samples + 8);
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plen = g_htons(rtpdump_info->num_samples);
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offset = g_htonl(rtpdump_info->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(rtpdump_info->samples, rtpdump_info->num_samples, 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 = (rtpstream_tapinfo_t *)arg;
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const struct _rtp_info *rtpinfo = (const struct _rtp_info *)arg2;
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rtp_stream_info_t new_stream_info;
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rtp_stream_info_t *stream_info = NULL;
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GList* list;
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rtpdump_info_t rtpdump_info;
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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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memset(&new_stream_info, 0, sizeof(rtp_stream_info_t));
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copy_address(&(new_stream_info.src_addr), &(pinfo->src));
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new_stream_info.src_port = pinfo->srcport;
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copy_address(&(new_stream_info.dest_addr), &(pinfo->dst));
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new_stream_info.dest_port = pinfo->destport;
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new_stream_info.ssrc = rtpinfo->info_sync_src;
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new_stream_info.payload_type = rtpinfo->info_payload_type;
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new_stream_info.payload_type_name = g_strdup(rtpinfo->info_payload_type_str);
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if (tapinfo->mode == TAP_ANALYSE) {
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/* check whether 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(&new_stream_info, (rtp_stream_info_t*)(list->data))==0)
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{
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stream_info = (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 (!stream_info) {
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new_stream_info.start_fd = pinfo->fd;
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new_stream_info.start_rel_time = pinfo->rel_ts;
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/* reset RTP stats */
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new_stream_info.rtp_stats.first_packet = TRUE;
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new_stream_info.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 = (struct _rtp_conversation_info *)p_get_proto_data(wmem_file_scope(), pinfo, proto_get_id_by_filter_name("rtp"), 0);
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if (p_conv_data)
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new_stream_info.setup_frame_number = p_conv_data->frame_number;
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else
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new_stream_info.setup_frame_number = 0xFFFFFFFF;
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stream_info = g_new(rtp_stream_info_t,1);
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*stream_info = new_stream_info; /* memberwise copy of struct */
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tapinfo->strinfo_list = g_list_append(tapinfo->strinfo_list, stream_info);
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}
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/* get RTP stats for the packet */
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rtp_packet_analyse(&(stream_info->rtp_stats), pinfo, rtpinfo);
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if (stream_info->rtp_stats.flags & STAT_FLAG_WRONG_TIMESTAMP
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|| stream_info->rtp_stats.flags & STAT_FLAG_WRONG_SEQ)
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stream_info->problem = TRUE;
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/* increment the packets counter for this stream */
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++(stream_info->packet_count);
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stream_info->stop_rel_time = pinfo->rel_ts;
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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(&new_stream_info, 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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rtpdump_info.rec_time = nstime_to_msec(&pinfo->abs_ts) -
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nstime_to_msec(&tapinfo->filter_stream_fwd->start_fd->abs_ts);
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rtpdump_info.num_samples = rtpinfo->info_data_len;
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rtpdump_info.samples = rtpinfo->info_data;
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rtp_write_sample(&rtpdump_info, tapinfo->save_file);
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}
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}
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else if (tapinfo->mode == TAP_MARK && tapinfo->tap_mark_packet) {
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if (rtp_stream_info_cmp(&new_stream_info, tapinfo->filter_stream_fwd)==0
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|| rtp_stream_info_cmp(&new_stream_info, tapinfo->filter_stream_rev)==0)
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{
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tapinfo->tap_mark_packet(tapinfo, 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 0;
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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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NOTE: Please keep the mimetypes in case insensitive alphabetical order.
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*/
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static const mimetype_and_clock mimetype_and_clock_map[] = {
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{"AMR", 8000}, /* [RFC4867][RFC3267] */
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{"AMR-WB", 16000}, /* [RFC4867][RFC3267] */
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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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{"EVRC", 8000}, /* [RFC3558] */
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{"EVRC0", 8000}, /* [RFC4788] */
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{"EVRC1", 8000}, /* [RFC4788] */
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{"EVRCB", 8000}, /* [RFC4788] */
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{"EVRCB0", 8000}, /* [RFC4788] */
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{"EVRCB1", 8000}, /* [RFC4788] */
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{"EVRCWB", 16000}, /* [RFC5188] */
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{"EVRCWB0", 16000}, /* [RFC5188] */
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{"EVRCWB1", 16000}, /* [RFC5188] */
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{"G7221", 16000}, /* [RFC3047] */
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{"G726-16", 8000}, /* [RFC3551][RFC4856] */
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{"G726-24", 8000}, /* [RFC3551][RFC4856] */
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{"G726-32", 8000}, /* [RFC3551][RFC4856] */
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{"G726-40", 8000}, /* [RFC3551][RFC4856] */
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{"G729D", 8000}, /* [RFC3551][RFC4856] */
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{"G729E", 8000}, /* [RFC3551][RFC4856] */
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{"GSM-EFR", 8000}, /* [RFC3551] */
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{"H263-1998", 90000}, /* [RFC2429],[RFC3555] */
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{"H263-2000", 90000}, /* [RFC2429],[RFC3555] */
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{"H264", 90000}, /* [RFC3984] */
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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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{"mpa-robust", 90000}, /* [RFC3119] */
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{"pointer", 90000}, /* [RFC2862] */
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{"raw", 90000}, /* [RFC4175] */
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{"red", 1000}, /* [RFC4102] */
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{"SMV", 8000}, /* [RFC3558] */
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{"SMV0", 8000}, /* [RFC3558] */
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{"t140", 1000}, /* [RFC4103] */
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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(const gchar *payload_type_str)
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{
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int i;
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/* Search for matching mimetype in reverse order to avoid false matches
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* when pt_mime_name_str is the prefix of payload_type_str */
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for (i = NUM_DYN_CLOCK_VALUES - 1; i > -1 ; 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 0;
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}
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/****************************************************************************/
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void
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rtp_packet_analyse(tap_rtp_stat_t *statinfo,
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packet_info *pinfo,
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const struct _rtp_info *rtpinfo)
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{
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double current_time;
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double current_jitter;
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double current_diff = 0;
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double nominaltime;
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double arrivaltime; /* Time relative to start_time */
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double expected_time;
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double absskew;
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guint32 clock_rate;
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/* Store the current time */
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current_time = nstime_to_msec(&pinfo->rel_ts);
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/* Is this the first packet we got in this direction? */
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if (statinfo->first_packet) {
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/* Save the MAC address of the first RTP frame */
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if( pinfo->dl_src.type == AT_ETHER){
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copy_address(&(statinfo->first_packet_mac_addr), &(pinfo->dl_src));
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}
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statinfo->start_seq_nr = rtpinfo->info_seq_num;
|
|
statinfo->stop_seq_nr = rtpinfo->info_seq_num;
|
|
statinfo->seq_num = rtpinfo->info_seq_num;
|
|
statinfo->start_time = current_time;
|
|
statinfo->timestamp = rtpinfo->info_timestamp;
|
|
statinfo->first_timestamp = rtpinfo->info_timestamp;
|
|
statinfo->time = current_time;
|
|
statinfo->lastnominaltime = 0;
|
|
statinfo->pt = rtpinfo->info_payload_type;
|
|
statinfo->reg_pt = rtpinfo->info_payload_type;
|
|
if (pinfo->net_src.type == AT_IPv6) {
|
|
statinfo->bw_history[statinfo->bw_index].bytes = rtpinfo->info_data_len + 48;
|
|
} else {
|
|
statinfo->bw_history[statinfo->bw_index].bytes = rtpinfo->info_data_len + 28;
|
|
}
|
|
statinfo->bw_history[statinfo->bw_index].time = current_time;
|
|
statinfo->bw_index++;
|
|
if (pinfo->net_src.type == AT_IPv6) {
|
|
statinfo->total_bytes += rtpinfo->info_data_len + 48;
|
|
} else {
|
|
statinfo->total_bytes += rtpinfo->info_data_len + 28;
|
|
}
|
|
statinfo->bandwidth = (double)(statinfo->total_bytes*8)/1000;
|
|
/* Not needed ? initialised to zero? */
|
|
statinfo->delta = 0;
|
|
statinfo->jitter = 0;
|
|
statinfo->diff = 0;
|
|
|
|
statinfo->total_nr++;
|
|
statinfo->flags |= STAT_FLAG_FIRST;
|
|
if (rtpinfo->info_marker_set) {
|
|
statinfo->flags |= STAT_FLAG_MARKER;
|
|
}
|
|
statinfo->first_packet_num = pinfo->num;
|
|
statinfo->first_packet = FALSE;
|
|
return;
|
|
}
|
|
|
|
/* Reset flags */
|
|
statinfo->flags = 0;
|
|
#if 0
|
|
/*According to bug https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=11478
|
|
* this code causes problems. A better solution is needed if there is need for the functionality */
|
|
/* Try to detect duplicated packets due to mirroring/span ports by comparing src MAC addresses.
|
|
* Chek for duplicates (src mac differs from first_packet_mac_addr) */
|
|
*/
|
|
if( pinfo->dl_src.type == AT_ETHER){
|
|
if(!addresses_equal(&(statinfo->first_packet_mac_addr), &(pinfo->dl_src))){
|
|
statinfo->flags |= STAT_FLAG_DUP_PKT;
|
|
statinfo->delta = current_time-(statinfo->time);
|
|
return;
|
|
}
|
|
}
|
|
#endif
|
|
/* 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 we wrap */
|
|
else if ( (statinfo->seq_num == 65535) && (rtpinfo->info_seq_num == 0) )
|
|
statinfo->seq_num = rtpinfo->info_seq_num;
|
|
/* Lost packets. If the prev seq is enormously larger than the cur seq
|
|
* we assume that instead of being massively late we lost the packet(s)
|
|
* that would have indicated the sequence number wrapping. An imprecise
|
|
* heuristic at best, but it seems to work well enough.
|
|
* https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=5958 */
|
|
else if (statinfo->seq_num+1 < rtpinfo->info_seq_num || statinfo->seq_num - rtpinfo->info_seq_num > 0xFF00) {
|
|
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;
|
|
}
|
|
|
|
/* 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;
|
|
|
|
/*
|
|
* Return for unknown payload types
|
|
* Ignore jitter calculation for clockrate = 0
|
|
*/
|
|
if (statinfo->pt < 96 ){
|
|
clock_rate = get_clock_rate(statinfo->pt);
|
|
}else{ /* Dynamic PT */
|
|
if ( rtpinfo->info_payload_type_str != NULL ){
|
|
/* Is it a "telephone-event" ?
|
|
* Timestamp is not increased for telepone-event packets impacting
|
|
* calculation of Jitter Skew and clock drift.
|
|
* see 2.2.1 of RFC 4733
|
|
*/
|
|
if (g_ascii_strncasecmp("telephone-event",rtpinfo->info_payload_type_str,(strlen("telephone-event")))==0){
|
|
clock_rate = 0;
|
|
statinfo->flags |= STAT_FLAG_PT_T_EVENT;
|
|
}else{
|
|
if(rtpinfo->info_payload_rate !=0){
|
|
clock_rate = rtpinfo->info_payload_rate;
|
|
}else{
|
|
clock_rate = get_dyn_pt_clock_rate(rtpinfo->info_payload_type_str);
|
|
}
|
|
}
|
|
}else{
|
|
clock_rate = 0;
|
|
}
|
|
}
|
|
|
|
/* Handle wraparound ? */
|
|
arrivaltime = current_time - statinfo->start_time;
|
|
|
|
nominaltime = (double)(guint32_wraparound_diff(rtpinfo->info_timestamp, statinfo->first_timestamp));
|
|
|
|
/* Can only analyze defined sampling rates */
|
|
if (clock_rate != 0) {
|
|
statinfo->clock_rate = clock_rate;
|
|
/* Convert from sampling clock to ms */
|
|
nominaltime = nominaltime /(clock_rate/1000);
|
|
|
|
/* Calculate the current jitter(in ms) */
|
|
if (!statinfo->first_packet) {
|
|
expected_time = statinfo->time + (nominaltime - statinfo->lastnominaltime);
|
|
current_diff = fabs(current_time - expected_time);
|
|
current_jitter = (15 * statinfo->jitter + current_diff) / 16;
|
|
|
|
statinfo->delta = current_time-(statinfo->time);
|
|
statinfo->jitter = current_jitter;
|
|
statinfo->diff = current_diff;
|
|
}
|
|
statinfo->lastnominaltime = nominaltime;
|
|
/* Calculate skew, i.e. absolute jitter that also catches clock drift
|
|
* Skew is positive if TS (nominal) is too fast
|
|
*/
|
|
statinfo->skew = nominaltime - arrivaltime;
|
|
absskew = fabs(statinfo->skew);
|
|
if(absskew > fabs(statinfo->max_skew)){
|
|
statinfo->max_skew = statinfo->skew;
|
|
}
|
|
/* Gather data for calculation of average, minimum and maximum framerate based on timestamp */
|
|
#if 0
|
|
if (numPackets > 0 && (!hardPayloadType || !alternatePayloadType)) {
|
|
/* Skip first packet and possibly alternate payload type packets */
|
|
double dt;
|
|
dt = nominaltime - statinfo->lastnominaltime;
|
|
sumdt += 1.0 * dt;
|
|
numdt += (dt != 0 ? 1 : 0);
|
|
mindt = (dt < mindt ? dt : mindt);
|
|
maxdt = (dt > maxdt ? dt : maxdt);
|
|
}
|
|
#endif
|
|
/* Gather data for calculation of skew least square */
|
|
statinfo->sumt += 1.0 * current_time;
|
|
statinfo->sumTS += 1.0 * nominaltime;
|
|
statinfo->sumt2 += 1.0 * current_time * current_time;
|
|
statinfo->sumtTS += 1.0 * current_time * nominaltime;
|
|
}
|
|
|
|
/* Calculate the BW in Kbps adding the IP+UDP header to the RTP -> 20bytes(IP) + 8bytes(UDP) */
|
|
if (pinfo->net_src.type == AT_IPv6) {
|
|
statinfo->bw_history[statinfo->bw_index].bytes = rtpinfo->info_data_len + 48;
|
|
} else {
|
|
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+1000/* ms */)<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;
|
|
};
|
|
/* IP hdr + UDP + RTP */
|
|
if (pinfo->net_src.type == AT_IPv6){
|
|
statinfo->total_bytes += rtpinfo->info_data_len + 48;
|
|
}else{
|
|
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;
|
|
|
|
|
|
/* Used by GTK code only */
|
|
statinfo->delta_timestamp = guint32_wraparound_diff(rtpinfo->info_timestamp, statinfo->timestamp);
|
|
|
|
/* Is it a packet with the mark bit set? */
|
|
if (rtpinfo->info_marker_set) {
|
|
statinfo->flags |= STAT_FLAG_MARKER;
|
|
}
|
|
|
|
/* Difference can be negative. We don't expect difference bigger than 31 bits. Difference don't care about wrap around. */
|
|
gint32 tsdelta=rtpinfo->info_timestamp - statinfo->timestamp;
|
|
if (tsdelta < 0) {
|
|
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->num;
|
|
}
|
|
if (clock_rate != 0) {
|
|
/* 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;
|
|
}
|
|
|
|
statinfo->time = current_time;
|
|
statinfo->timestamp = rtpinfo->info_timestamp;
|
|
statinfo->stop_seq_nr = rtpinfo->info_seq_num;
|
|
statinfo->total_nr++;
|
|
statinfo->last_payload_len = rtpinfo->info_payload_len - rtpinfo->info_padding_count;
|
|
|
|
return;
|
|
}
|
|
|
|
/*
|
|
* Editor modelines - http://www.wireshark.org/tools/modelines.html
|
|
*
|
|
* Local variables:
|
|
* c-basic-offset: 8
|
|
* tab-width: 8
|
|
* indent-tabs-mode: t
|
|
* End:
|
|
*
|
|
* vi: set shiftwidth=8 tabstop=8 noexpandtab:
|
|
* :indentSize=8:tabSize=8:noTabs=false:
|
|
*/
|