You can also configure this parameter in the configuration file (cfg.xml) by entering a string in this format:
<Call_Statistics ua="na">Yes</Call_Statistics>
The allowed values are Yes|No. The defaut value is No.
Step 3 Click Submit All Changes.
Attributes for Call Statistics in SIP Messages
Table 32: Audio: RTP-RxStat Payload
MandatoryDescriptionAttribute
YesDuration of media session/callDur
YesNumber of RTP packets receivedPkt
NoNumber of RTP packets octets receivedOct
YesNumber of RTP packets received and discarded as late due to
outside of buffer window
LatePkt
YesNumber of RTP packets lostLostPkt
YesAverage Jitter over session durationAvgJit
YesStream/session codec negotiatedVoRxCodec
YesPacket size in millisecondsVoPktSizeMs
YesMax Jitter detectedmaxJitter
YesLatency/one way delayVoOneWayDelayMs
YesMean opinion score conversational quality for the session, per
RFC https://tools.ietf.org/html/rfc3611
MOScq
NoMaximum number of sequential packets lostmaxBurstPktLost
NoAverage number of sequential packets lost in a burst. The number
can be used in conjunction with overall loss to compare the
impact of loss on the call quality.
avgBurstPktLost
YesType of network the device is on (if possible).networkType
YesHardware client that the session/media is running on. More
relevant for soft clients but still useful for hard phones. For
example, Model number CP-8865.
hwType
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
217
Cisco IP Phone Configuration
Attributes for Call Statistics in SIP Messages
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