Call Statistics Screen - Cisco 7942G 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
Step 1 Press the Settings button.
Step 2 Select Status.
Select Call Statistics.
Step 3
The Call Statistics screen displays these items.
Table 7-6
Item
Rcvr Codec
Sender Codec
Rcvr Size
Cisco Unified IP Phone 7962G and 7942G Administration Guide for Cisco Unified Communications Manager 6.1(3)
OL-17679-01
Expansion Module Statistics (continued)
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
Cisco Unified IP Phone Remotely."
Call Statistics Items
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
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
Chapter 8, "Monitoring the
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