Attributes For Call Statistics In Sip Messages - Cisco 8800 Series Manual

Cisco IP Phone Configuration
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
Attribute
Dur
Pkt
Oct
LatePkt
LostPkt
AvgJit
VoRxCodec
VoPktSizeMs
maxJitter
VoOneWayDelayMs
MOScq
maxBurstPktLost
avgBurstPktLost
networkType
hwType
Description
Duration of media session/call
Number of RTP packets received
Number of RTP packets octets received
Number of RTP packets received and discarded as late due to
outside of buffer window
Number of RTP packets lost
Average Jitter over session duration
Stream/session codec negotiated
Packet size in milliseconds
Max Jitter detected
Latency/one way delay
Mean opinion score conversational quality for the session, per
RFC https://tools.ietf.org/html/rfc3611
Maximum number of sequential packets lost
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.
Type of network the device is on (if possible).
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.
Cisco IP Phone 8800 Series Multiplatform Phone Administration Guide for Release 11.3(1) and Later
Attributes for Call Statistics in SIP Messages
Mandatory
Yes
Yes
No
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
No
Yes
Yes
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