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SIP module: Difference between revisions
→Per-Call statistics
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** Firmware 3.0: 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. 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. | ** Firmware 3.0: 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. 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. | ||
** Firmware 3.1: The "SIP displayed" and "SIP+RTP displayed" PCAP buttons next to the filter bar will capture all SIP calls visible on the table page. A filter that limits the visible SIP calls is taken into account. You can increase the elements of the table to have more calls captured. | ** Firmware 3.1: The "SIP displayed" and "SIP+RTP displayed" PCAP buttons next to the filter bar will capture all SIP calls visible on the table page. A filter that limits the visible SIP calls is taken into account. You can increase the elements of the table to have more calls captured. | ||
* The RTP IP and | * 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. | ||
* The RTCP IP and port for caller and the callee is shown, if RTCP packets were seen. RTCP packets contain information about quality parameters such as RTP packet loss and jitter. They are sent by the caller or callee and provide feedback of the RTP packets they have received. | |||
* Call initiation, Call established, Call end time and Call duration are displayed. | * Call initiation, Call established, Call end time and Call duration are displayed. | ||
* RTP packet loss per direction is shown. | * RTP packet loss per direction is shown. | ||
* RTP payload type (codec) per direction is shown. | * RTP payload type (codec) per direction is shown. | ||
* RTCP reported packet loss is shown. The indicated direction is where the loss occurred. | |||
* QoS tags per direction of both the SIP and RTP flow are shown. | * QoS tags per direction of both the SIP and RTP flow are shown. | ||
* Jitter of RTP stream per direction is calculated and shown. | * Jitter of RTP stream per direction is calculated and shown. | ||
* RTCP reported jitter is shown. The indicated direction is where the jitter was measured. | |||
* The maximum clock skew (negative or positive) is shown. (Version >= 3.1) | * The maximum clock skew (negative or positive) is shown. (Version >= 3.1) | ||
* Mean Opinion Score (MOS) per direction is evaluated and shown. | * Mean Opinion Score (MOS) per direction is evaluated and shown. | ||
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It is also possible to capture traffic for a specific SIP call by clicking on the corresponding button in the last column. | 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. | 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. | ||
The toggle button "Caller/callee (combined)" allows for switching the display to have both caller and callee counters in separate columns for easier sorting. The toggle button "Reduced view" allows switching the display to a condensed mode where one call only needs one row. With this mode it is possible to show up to 100 lines on one page. | |||
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. | 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. | ||