Cisco 6921 Administration Manual page 117

Chapter 7 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone
To display the Call Statistics screen for information about the latest voice stream, follow these steps:
Procedure
Step 1 Press the Applications button.
Select Admin Settings.
Step 2
Select Status.
Step 3
Select Call Statistics.
Step 4
The Call Statistics screen displays these items:
Table 7-4
Item
Rcvr Codec
Sender Codec
Rcvr Size
Sender Size
Rcvr Packets
Sender Packets
Avg Jitter
Max Jitter
Rcvr Discarded
Rcvr Lost Packets
Voice Quality Metrics
Cumulative
Conceal Ratio
Cisco Unified IP Phone 6921, 6941, and 6961 Administration Guide for Cisco Unified Communications Manager 8.0 (SCCP)
OL-20851-01
Call Statistics Items for the Cisco Unified Phone 6900 Series
Description
Type of voice stream received (RTP streaming audio from codec): G.729, G.711
u-law, G.711 A-law.
Type of voice stream transmitted (RTP streaming audio from codec): G.729, G.711
u-law, G.711 A-law.
Size of voice packets, in milliseconds, in the receiving voice stream (RTP
streaming audio).
Size of voice packets, in milliseconds, in the transmitting voice stream.
Number of RTP voice packets received since voice stream was opened.
Note This number is not necessarily identical to the number of RTP voice
packets received since the call began because the call might have been
placed on hold.
Number of RTP voice packets transmitted since voice stream was opened.
Note This number is not necessarily identical to the number of RTP voice
packets transmitted since the call began because the call might have been
placed on hold.
Estimated average RTP packet jitter (dynamic delay that a packet encounters
when going through the network) observed since the receiving voice stream was
opened.
Maximum jitter observed since the receiving voice stream was opened.
Number of RTP packets in the receiving voice stream that have been discarded
(bad packets, too late, and so on).
Note The phone will discard payload type 19 comfort noise packets that are
generated by Cisco Gateways, which will increment this counter.
Missing RTP packets (lost in transit).
Total number of concealment frames divided by total number of speech frames
received from start of the voice stream.
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