wireshark/docbook/wsug_src/WSUG_chapter_telephony.adoc

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// WSUG Chapter Telephony
[[ChTelephony]]
== Telephony
[[ChTelIntroduction]]
=== Introduction
Wireshark provides a wide range of telephony related network statistics which
can be accessed via the menu:Telephony[] menu.
These statistics range from specific signaling protocols, to analysis of
signaling and media flows. If encoded in a compatible encoding the media flow
can even be played.
The protocol specific statistics windows display detailed information of
specific protocols and might be described in a later version of this document.
Some of these statistics are described at the
{wireshark-wiki-url}Statistics pages.
[[ChTelVoipCalls]]
=== VoIP Calls
The VoIP Calls window shows a list of all detected VoIP calls in the captured
traffic. It finds calls by their signaling.
More details can be found on the {wireshark-wiki-url}VoIP_calls page.
[[ChTelANSI]]
=== ANSI
{missing}
[[ChTelGSM]]
=== GSM
{missing}
[[ChTelIAX2Analysis]]
Qt: Initial RTP playback. 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>
2014-12-13 00:51:40 +00:00
=== IAX2 Stream Analysis
Qt: Initial RTP playback. 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>
2014-12-13 00:51:40 +00:00
The “IAX2 Stream Analysis” dialog shows statistics for the forward and reverse
streams of a selected IAX2 call along with a graph.
Qt: Initial RTP playback. 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>
2014-12-13 00:51:40 +00:00
[[ChTelISUPMessages]]
=== ISUP Messages
{missing}
[[ChTelLTE]]
=== LTE
[[ChTelLTEMACTraffic]]
==== LTE MAC Traffic Statistics
Statistics of the captured LTE MAC traffic. This window will summarize the LTE
MAC traffic found in the capture.
.The “LTE MAC Traffic Statistics” window
image::wsug_graphics/ws-stats-lte-mac-traffic.png[{screenshot-attrs}]
The top pane shows statistics for common channels. Each row in the middle pane
shows statistical highlights for exactly one UE/C-RNTI. In the lower pane, you
can see the for the currently selected UE/C-RNTI the traffic broken down by
individual channel.
[[ChTelLTERLCGraph]]
==== LTE RLC Graph
{missing}
[[ChTelLTERLCTraffic]]
==== LTE RLC Traffic Statistics
Statistics of the captured LTE RLC traffic. This window will summarize the LTE
RLC traffic found in the capture.
.The “LTE RLC Traffic Statistics” window
image::wsug_graphics/ws-stats-lte-rlc-traffic.png[{screenshot-attrs}]
At the top, the check-box allows this window to include RLC PDUs found within
MAC PDUs or not. This will affect both the PDUs counted as well as the display
filters generated (see below).
The upper list shows summaries of each active UE. Each row in the lower list
shows statistical highlights for individual channels within the selected UE.
The lower part of the windows allows display filters to be generated and set for
the selected channel. Note that in the case of Acknowledged Mode channels, if a
single direction is chosen, the generated filter will show data in that
direction and control PDUs in the opposite direction.
[[ChTelMTP3]]
=== MTP3
{missing}
[[ChTelOsmux]]
=== Osmux
{missing}
[[ChTelRTPAnalysis]]
=== RTP Analysis
The RTP analysis function takes the selected RTP stream (and the reverse stream,
if possible) and generates a list of statistics on it.
.The “RTP Stream Analysis” window
image::wsug_graphics/ws-tel-rtpstream-analysis.png[{screenshot-attrs}]
Starting with basic data as packet number and sequence number, further
statistics are created based on arrival time, delay, jitter, packet size, etc.
Besides the per packet statistics, the lower pane shows the overall statistics,
with minimums and maximums for delta, jitter and clock skew. Also an indication
of lost packets is included.
The RTP Stream Analysis window further provides the option to save the RTP
payload (as raw data or, if in a PCM encoding, in an Audio file). Other options
a to export and plot various statistics on the RTP streams.
[[ChTelRtpPlayer]]
The RTP Player window lets you play back RTP audio data. In order to use
this feature your version of Wireshark must support audio and the codecs
used by each RTP stream.
More details can be found on the
{wireshark-wiki-url}VoIP_calls page.
[[ChTelRTSP]]
=== RTSP
In the Real Time Streaming Protocol (RTSP) menu the user can check the Packet Counter window. It shows Total RTCP Packets and divided into RTSP Response Packets, RTSP Request Packets and Other RTSP packets. The user can filter, copy or save the data into a file.
{missing}
[[ChTelSCTP]]
=== SCTP
{missing}
[[ChTelSMPPOperations]]
=== SMPP Operations
{missing}
[[ChTelUCPMessages]]
=== UCP Messages
{missing}
[[ChTelH225]]
=== H.225
{missing}
[[ChTelSIPFlows]]
=== SIP Flows
Session Initiation Protocol (SIP) Flows window shows the list of all captured SIP transactions, such as client registrations, messages, calls and so on.
NOTE: This window will list both complete and in-progress SIP transactions.
User's operations in the window:
* Filtering the captured data. To do so, click the btn:[Prepare Filter] button.
* Checking the sequence diagram. To do so, click the btn:[Flow Sequence] button.
* Listen to the captured RTP stream if a decoder for the payload exists. To do so, click the btn:[Play Streams] button.
* Copy the data in the `SCV` or `YAML` format.
[[ChTelSIPStatistics]]
=== SIP Statistics
SIP Statistics window shows captured SIP transactions. It is divided into SIP Responses and SIP Requests. In this window the user can filter, copy or save the statistics into a file.
[[ChTelWAPWSPPacketCounter]]
=== WAP-WSP Packet Counter
{missing}
// End of WSUG Chapter Telephony