325
edits
SIP module: Difference between revisions
Jump to navigation
Jump to search
→Per-Call statistics
Remco.derooy (talk | contribs) m (→SIP dashboard) |
Remco.derooy (talk | contribs) |
||
| Line 23: | Line 23: | ||
== Per-Call statistics == | == Per-Call statistics == | ||
The SIP calls tab contains a table listing all SIP calls in the selected interval. | The SIP calls tab, contains a table listing all SIP calls in realtime or from the selected interval. | ||
{| class="wikitable sortable" | {| class="wikitable sortable" | ||
| Line 32: | Line 32: | ||
For each call, the call statistics and the related RTP statistics are shown. These include: | For each call, the call statistics and the related RTP statistics are shown. These include: | ||
* The caller and the callee | * The IP for caller and the callee is shown, as well as the phone number/SIP address and all correlated naming informations.<br> | ||
* 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 | By clicking on the "Details" button on the far right, a detailed summary page for the particular call will be displayed. | ||
* The RTP caller and callee | * 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 call identifiers. This allows for filtering all legs of one call, even if several SIP flows were involved. | ||
* Call initiation, established, end time and duration are displayed | When such filter is set, the "Filter SIP" and "Filter SIP+RTP/RTCP" PCAP buttons next to the filter bar, will capture all associated legs of a SIP call. | ||
* RTP packet loss per direction is shown | Please note, that only the associated calls displayed on 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 (10) legs can be captured. | ||
* RTP payload type per direction is shown | * The RTP IP and Port for caller and the callee is shown, if the call is/was successful and an RTP stream could be initiated. | ||
* QoS tags per direction of both SIP and RTP flow are shown | * Call initiation, Call established, Call end time and Call duration are displayed. | ||
* Jitter of RTP stream per direction | * RTP packet loss per direction is shown. | ||
* Mean Opinion Score (MOS) per direction | * RTP payload type (codec) per direction is shown. | ||
* Audio level (min, max and RMS) per direction if G.711 codec is used. The values are extracted from the samples of the RTP packets. The max value will show peaks whereas the RMS will show an average loudness level and | * QoS tags per direction of both the SIP and RTP flow are shown. | ||
* Expected and | * Jitter of RTP stream per direction is calculated and shown. | ||
* Status of origin SIP invite request that initiated the call | * Mean Opinion Score (MOS) per direction is evaluated and shown. | ||
* The graph shows information of the RTP stream per direction. By using the | * Audio level (min, max and RMS) per direction if G.711 codec is used. The values are extracted from the samples of the RTP packets. The max value will show peaks, whereas the RMS will show an average loudness level and a better impression of the whole call. The values are displayed in dbFS. | ||
* Expected and actual 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 graphselector<sup>1</sup> packet rate, dropped packets, overhead (duplicated) packets, Jitter, MOS and max audio level over time can be displayed. | |||
[[File:Graphselector.jpg|none]] | |||
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. | ||
First click the “Zoom” button below the PCAP button, to select the time range of a call. 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/RTCP" allows extracting both the SIP and RTP packets of that specific call. | |||
For audible validation of call quality (e.g. distortion not caused by the network), it is possible to extract the audio of RTP streams as an MP3. Currently there are two buttons, each containing one direction (A/B) of the audio stream. | |||
The MP3 extraction is only supported for the following codecs: | |||
* G.711 A-law (ALAW) | |||
* G.711 μ-law (ULAW) | |||
* G.722 | |||
* G.729 | |||
On the bottom of the table the "CSV download" button allows exporting all pages of the table in CSV format. | |||
'''Addendum - MOS:''' | |||
The displayed mean opinion score (MOS) of a call is calculated from various metrics within an RTP stream. | |||
Following scores are used: | Following scores are used: | ||
| Line 67: | Line 80: | ||
* G.729 | * G.729 | ||
* GSM | * GSM | ||
=== Call detail page === | === Call detail page === | ||