SIP module: Difference between revisions

SIP module (view source)

Revision as of 08:01, 10 June 2020

1,070 bytes added , 10 June 2020

→‎Per-Call statistics

Martin.fesser

340

edits

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 related RTP statistics are shown. These include:
-* The caller and the callee (IP as well as SIP address/phone number)
+* The caller and the callee IP is shown as well as SIP address/phone number and all correlated naming informations. By clicking on "Go to call details" a separate detail page will be opened.
-* The RTP caller and callee (IP and port)
+* The Call ID column lists the call ID and all similar identifiers (e.g. P-Palladion-ID) for the SIP flow. The small filter icon allows for setting a filter for any of these identifiers. This allows for filtering all legs of one call even if several SIP flows are involved. If the filter is set, the "SIP Filter" and "SIP+RTP Filter" PCAP buttons next to the filter bar allow captures of all legs. Please note, that only the displayed ones of the current page can be captured. If there are more legs you may want to increase the visible items per table page. Up to ten legs can be captured.
-* Number of packets per direction
+* The RTP caller and callee IP and port is shown if the call is successful and a RTP stream could be initiated.
-* Jitter pre direction
+* Call initiation, established, end time and duration are displayed
+* RTP packet loss per direction is shown
+* RTP payload type per direction is shown
+* QoS tags per direction of both SIP and RTP flow are shown
+* Jitter of RTP stream per direction
 * Mean Opinion Score (MOS) per direction
+* Audio level (min, max and RMS) per direction if G.711 codec is used. The max value will show peaks whereas the RMS will show an average loudness level and gives a better impression of the whole call. The values are displayed in dbFS.
+* Expected and seen sample rate per direction is displayed and the measured bitrate of the RTP stream
+* Status of origin SIP invite request that initiated the call
+* The graph shows information of the RTP stream per direction. By using the graph selector packet rate, dropped packets, overhead (duplicated) packets, Jitter, MOS and max audio level over time can be displayed.
-The "Call ID" column lists the call ID and all similar identifiers (e.g. P-Palladion-ID) for the SIP flow. The small filter icon allows for setting a filter for any of these identifiers. This allows for filtering all legs of one call even if several SIP flows are involved. If the filter is set, the "SIP Filter" and "SIP+RTP Filter" PCAP buttons next to the filter bar allow captures for all legs. Please note, that only the displayed ones of the current page can be captured. If there are more legs you may want to increase the visible items per table page. Only up to ten legs can be captured.
+It is also possible to capture traffic for a specific SIP call by clicking on the corresponding button in the last column. The PCAP button "SIP" will only extract the SIP packets for that specific call by taking IP addresses and SIP call IDs into account. The PCAP button "SIP+RTP" allows extracting both SIP and RTP packets of that specific call.
-It is also possible to capture traffic for a specific SIP call by clicking on the corresponding button in the last column.
 To select the time range of a call, click the “Zoom” button below the PCAP button.
-When activating the RTP toggle, the table will include a second line for every call showing graphs describing the quality of the call.
+The displayed mean opinion score (MOS) of a call is calculated from other statistics of the RTP stream.
-The table contains the mean opinion score (MOS) of a call, calculated from other statistics of the RTP stream.
 Following scores are used: