Call Statistics Screen - Cisco CP-7961G Administration Manual

Chapter 7 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone
Table 7-5
Item
TX Retransmit
TX Buffer Full

Call Statistics Screen

You can access the Call Statistics screen on the phone to display counters, statistics, and voice-quality
metrics in the following ways:
•
•
Note
A single call can have multiple voice streams, but data is captured for only the last voice stream. A voice
stream is a packet stream between two endpoints. If one endpoint is put on hold, the voice stream stops
even though the call is still connected. When the call resumes, a new voice packet stream begins, and the
new call data overwrites the former call data.
To display the Call Statistics screen for information about the last voice stream, follow these steps:
Procedure
Press the Settings button.
Step 1
Select Status.
Step 2
Select Call Statistics.
Step 3
Table 7-6
Table 7-6
Item
Rcvr Codec
Sender Codec
Rcvr Size
OL-21011-01
Expansion Module Statistics (continued)
Description
Number of packets that have been retransmitted to the expansion module
Number of packets discarded because the expansion module was not able to
accept new messages
During call—You can view the call information by rapidly pressing the ? button twice.
After the call—You can view the call information captured during the last call by displaying the Call
Statistics screen.
You can also remotely view the call statistics information by using a web browser to access the
Streaming Statistics web page. This web page contains additional RTCP statistics not available
on the phone. For more information about remote monitoring, see
Unified IP Phone Remotely, page 8-1
describes the items displayed on the Call Statistics screen:
Call Statistics Items
Cisco Unified IP Phone Administration Guide for Cisco Unified Communications Manager 8.0 (SCCP and SIP)
Description
Type of voice stream received (RTP streaming audio from codec):
G.729, G.728/iLBC, G.711 u-law, G.711 A-law, or Lin16k.
Type of voice stream transmitted (RTP streaming audio from codec):
G.729, G.728/iLBC, G.711 u-law, G.711 A-law, or Lin16k.
Size of voice packets, in milliseconds, in the receiving voice stream
(RTP streaming audio).
Status Menu
Monitoring the Cisco
7-13