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
Mandatory
Description
Attribute
Yes
Duration of media session/call
Dur
Yes
Number of RTP packets received
Pkt
No
Number of RTP packets octets received
Oct
Yes
Number of RTP packets received and discarded as late due to
outside of buffer window
LatePkt
Yes
Number of RTP packets lost
LostPkt
Yes
Average Jitter over session duration
AvgJit
Yes
Stream/session codec negotiated
VoRxCodec
Yes
Packet size in milliseconds
VoPktSizeMs
Yes
Max Jitter detected
maxJitter
Yes
Latency/one way delay
VoOneWayDelayMs
Yes
Mean opinion score conversational quality for the session, per
RFC
https://tools.ietf.org/html/rfc3611
MOScq
No
Maximum number of sequential packets lost
maxBurstPktLost
No
Average 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
Yes
Type of network the device is on (if possible).
networkType
Yes
Hardware 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
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Cisco IP Phone Configuration
Attributes for Call Statistics in SIP Messages
Summary of Contents for 8800 Series
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