path: root/ui/voip_calls.h
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2017-09-21Complete move of tap-sequence-analysis.c functionality to sequence_analysis.cMichael Mann1-2/+1
Since dissectors are now populating the timestamp of the seq_analysis_item_t structure within the tap function, don't have the sequence_anaylsis redo it when writing an ASCII file. This removes the need for the capture_file parameter and simplifies the logic a bit. Also just have GUI register the tap itself. It will provide for some more flexibility in the future. Change-Id: I55b2f951b977ea70ac9f7eb4929245b0779e5f0e Reviewed-on: https://code.wireshark.org/review/23650 Petri-Dish: Michael Mann <mmann78@netscape.net> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2015-12-09Qt: various fixes to VoIP calls / RTP player windowsPascal Quantin1-2/+2
- Flush any remaining tapped packets before emitting captureFileRetapFinished(). This ensures that all packets have been treated before returning from retapPackets(). - Remove VoIP tap listeners when captureFileRetapFinished() is emitted. This avoid summing stats each time the RTP player is opened, leading to wrong information in VoIP calls window - Change voip_calls_tapinfo_t redraw member from a boolean to bitmap so as to identify which tap should call the tapinfo->tap_draw() callback. This allows fixing a race condition where the RTP player can be empty in Qt UI - Reset some more statistics in voip_calls_reset_all_taps() Change-Id: Ie7681702c81d338185c1813f2d340a437edf3a04 Reviewed-on: https://code.wireshark.org/review/12474 Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com> Petri-Dish: Pascal Quantin <pascal.quantin@gmail.com> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Anders Broman <a.broman58@gmail.com>
2015-10-05Fix warnings introduced by "Qt: Initial RTP playback"Peter Wu1-6/+1
Change-Id: I28ae077be535f32ef81ac370d6782033f219017d Reviewed-on: https://code.wireshark.org/review/10777 Reviewed-by: Peter Wu <peter@lekensteyn.nl> Petri-Dish: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Anders Broman <a.broman58@gmail.com> Reviewed-by: Michael Mann <mmann78@netscape.net>
2015-10-02Qt: Initial RTP playback.Gerald Combs1-3/+17
Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-07-26Avoid duplicate SIP and Q.931 calls in VoIP Calls list in case of Q.931Michal Pazdera1-0/+1
transported over SIP as described in RFC 3204, 3.2 QSIG Media Type. Change-Id: Ida30a7b115e60fa64d30cfc1f4b7c11be724f8ee Reviewed-on: https://code.wireshark.org/review/9479 Reviewed-by: Anders Broman <a.broman58@gmail.com>
2015-02-13Replace tabs by spaces when editor modelines has "expandtab"Bill Meier1-71/+71
Change-Id: If7a6f2697be732ae4f94ed8b845fd293c32510f7 Also: tabs-stops should be 8 Reviewed-on: https://code.wireshark.org/review/7100 Reviewed-by: Bill Meier <wmeier@newsguy.com>
2014-12-20Consolidate RTP stream structs.Gerald Combs1-1/+1
Consolidate the three different RTP stream structs in ui/rtp_stream.h, ui/gtk/rtp_player.c, and ui/voip_calls.c into one. Make the member names a bit more consistent. Document what each GList contains. Use nstime_t for timestamps since that's what we get from the frame data. Use g_new0 to initialize our structs. Change-Id: I2b3f8f2051394a6a98a5c7bc49c117f07161d031 Reviewed-on: https://code.wireshark.org/review/5843 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-09Qt: Initial VoIP Calls dialog.Gerald Combs1-1/+17
Add Telephony menu items for VoIP Calls and SIP Flows. Put VoIP Calls at the top, since that seems to be the primary item. Add configure-time checks for QtMultimediaWidgets in anticipation of adding a VoIP playback dialog. Add an icon for the playback button. (Yes, I've been avoiding GNOME-level gratuitous icons so far but this is one of the rare occiasions where it makes sense.) Add a help link define for the VoIP calls dialog. Change-Id: I5d0799685c598ad9af76fe9667f8ea7d14b66050 Reviewed-on: https://code.wireshark.org/review/5674 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-11-19voip_calls: Fix tap callbacks.Gerald Combs1-1/+1
We don't need to call the VoIP tap reset and draw callbacks repeately. Do so only once from the RTP tap. Packet callbacks should return a gboolean. Clean up some function names and make some static. Change-Id: I5c934ce8ce7f279861e8cc73235bbfc27d7fe622 Reviewed-on: https://code.wireshark.org/review/5396 Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-11-18voip_calls: Move to ui.Gerald Combs1-0/+254
Move voip_calls.[ch] to ui. Add callbacks to voip_calls_tapinfo_t. Remove unused function definitions. Change-Id: Ib12db7053d53afa81ef2a66dc0cfe681bc624dd2 Reviewed-on: https://code.wireshark.org/review/5379 Reviewed-by: Gerald Combs <gerald@wireshark.org>