two different variables for types voip_rtp_stream_info_t (strinfo) and
voip_calls_info_t (callsinfo) instead of using strinfo for both.
svn path=/trunk/; revision=24038
est. Use g_ascii_strcasecmp() and g_ascii_strncasecmp(), and supply our
own versions if they're missing from GLib (as is the case with GLib
1.x).
In the code to build the list of named fields for Diameter, don't use
g_strdown(); do our own g_ascii_-style upper-case to lower-case mapping
in the hash function and use g_ascii_strcasecmp() in the compare
function.
We do this because there is no guarantee that toupper(), tolower(), and
functions that use them will, for example, map between "I" and "i" in
all locales; in Turkish locales, for example, there are, in both
upper case and lower case, versions of "i" with and without a dot, and
the upper-case version of "i" is "I"-with-a-dot and the lower-case
version of "I" is "i"-without-a-dot. This causes strings that should
match not to match.
This finishes fixing bug 2010 - an earlier checkin prevented the crash
(as there are other ways to produce the same crash, e.g. a bogus
dictionary.xml file), but didn't fix the case-insensitive string matching.
svn path=/trunk/; revision=23623
H.225
- change RysMessage_vals to h225_Rasmessage_vals
- use #.PDU directive for H323-UserInformation and RasMessage instead of implementing it by hands
- register RasMessage_PDU as "h225.ras" dissector for calling it from H.460
asn2wrs make PDUs exportable
svn path=/trunk/; revision=23226
I fixed a bug in packet_unistim.c which was causing a 'malformed packet' to
appear when an open stream command is read with no endpoint ip (as is sometimes
done) I added a simple msg_len check to avoid this.
I also added UFTP to the unistim dissector and adjusted the unistim call-detection
in voip-info.c to start on an open stream as well as keypresses..
svn path=/trunk/; revision=22929
find attached several trivial diffs for viewing RTP SSRC values in Hex
rather than Dec at various places in the UI.
Also includes change from BASE_DEC to BASE_HEX_DEC for corresponding RTP
and RTCP dissector header fields.
svn path=/trunk/; revision=22017
* add SUA to the "VoIP Calls" tap.
* propagate changes to packet-sccp.h to other dissectors
From Neil Piercy:
* add SLR, DLR and CAUSE to COL_INFO
svn path=/trunk/; revision=21126
- instead of wrongly using the h248 call counter use tapinfo's counter
graph_analysis.c:
- beautify the code (I was trying to the bug that got fixed by bzeroing m3ua tap data however this looks better!).
svn path=/trunk/; revision=21063
Create two new files (ws_strsplit.[ch]) that use GTK2 code to override
the buggy g_strsplit() function when compiling for GTK1. Include this
work-around function (ws_strsplit) in libwireshark.def. Add notes on usage
to README.developer. Include epan/ws_strsplit.h in all files that use
g_strsplit().
svn path=/trunk/; revision=20804
Find attached a patch for this bug. The problem was actually in the "Voip Calls" logic, when the first RTP packet was after the last signaling packet (e.g. a call connected and the release not captured), that caused the RTP to not be added to the graph list and therefor to the player.
svn path=/trunk/; revision=19667
In tests for whether we're built with the RTP player, test
HAVE_LIBPORTAUDIO first - hopefully we can eventually get rid of the
test for the GTK+ version number.
svn path=/trunk/; revision=19641
player, so that we get declartions of rtp_player_init() and
add_rtp_packet().
Constify the first argument to add_rtp_packet(), as it's passed a
pointer to a const value.
svn path=/trunk/; revision=19272
- Change the "listen_rtp" to "rtp_player"
- Change from a plugin to be part of the core
- By default it will not compile with the rtp_player. In order to
compile it is necessary to:
+ For windows: uncomment the line
"PORTAUDIO_DIR=$(WIRESHARK_LIBS)\portaudio_v18_1" in config.nmake
+ For linux: using the "--with-portaudio=yes"
svn path=/trunk/; revision=19094
I'm not sure if this will help with the problems that Keith French is
seeing, but when I loaded some of my old H.323 traces, one of them would
assert/abort. This patch fixes that assertion (looks like it was obviously
asserting on the wrong pointer variable).
svn path=/trunk/; revision=18791
Mike Oliveras has indicated that for MGCP voip calls, 2 seconds may be a
better timeout for still matching DLCX requests to a hung-up endpoint,
as in this patch.
svn path=/trunk/; revision=18662
this is actually a false positive in coverity and can not trigger since htis pointer can not be null here
but verifying this before the varialbe is dereferenced does not hurt in case
the file is changed and this contract is broken.
checking the
svn path=/trunk/; revision=18542
Fix the part where the DeleteConnection messages and
responses were not being included in the graphs.
There is a test that allows messages such as these for 1 second after
the call has been hung up, but the time calculation was wrong.
svn path=/trunk/; revision=18479
From Cvetan Ivanov:
voip_calls.c
It seems to me that in gtk/voip_calls.c tmp_h323info->guid is pointer itself, therefore:
memcmp(&tmp_h323info->guid
should in fact read:
memcmp(tmp_h323info->guid
svn path=/trunk/; revision=18424
H225.cnf
I noticed is that the voip call flow graph does not have a label for the setupAck packet. I traced this to the empty frame_label.
voip_calls.c
It seems to me that in gtk/voip_calls.c tmp_h323info->guid is pointer itself, therefore:
memcmp(&tmp_h323info->guid
should in fact read:
memcmp(tmp_h323info->guid
svn path=/trunk/; revision=18304
of 16 bytes. Use "sizeof" for the size of e_guid_t's, and use structure
assignment to copy GUID values.
Make functions such as append_h225ras_call() and new_h225ras_call() take
pointers to e_guid_t's as arguments.
Define GUID_LEN in epan/guid-utils.h and use it as the length of a GUID
in a packet. (Note that "sizeof e_guid_t" is not guaranteed to be 16,
although it is guaranteed to be the size of an e_guid_t.)
When constructing a display filter that matches a GUID, use
guid_to_str() to construct the string for the GUID.
svn path=/trunk/; revision=17676
a patch for the VoipCalls to fix a couple of issues:
- a problem with the RTP Events (RFC2833) not been handle correctly
- Display the RTP stream in time order when the setup frame is after the
RTP stream.
- fix a init issue that caused the H245 packet to not been displayed
correctly.
svn path=/trunk/; revision=17383
Stephen Fisher