Cisco 7971G-GE Administration Guide - Page 145

Call Statistics Screen, Settings, Status, Call Statistics

Page 145 highlights

Chapter 7 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone Status Menu Table 7-5 Expansion Module Statistics (continued) Item TX Retransmit TX Buffer Full 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 Call Statistics Screen The Call Statistics screen displays counters statistics and voice-quality metrics in these ways: • 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. Note 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 Chapter 8, "Monitoring the Cisco Unified IP Phone Remotely." 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 Step 2 Step 3 Press the Settings button. Select Status. Select Call Statistics. Table 7-6 explains the items displayed in the Call Statistics screen: Table 7-6 Call Statistics Items Item Rcvr Codec Sender Codec Rcvr Size Description Type of voice stream received (RTP streaming audio from codec): G.729, 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). OL-15299-01 Cisco Unified IP Phone 7970G/7971G-GE Administration Guide for Cisco Unified Communications Manager 7.0 7-13

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7-13
Cisco Unified IP Phone 7970G/7971G-GE Administration Guide for Cisco Unified Communications Manager 7.0
OL-15299-01
Chapter 7
Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone
Status Menu
Call Statistics Screen
The Call Statistics screen displays counters statistics and voice-quality metrics in these ways:
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.
Note
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
Chapter 8, “Monitoring the
Cisco Unified IP Phone Remotely.”
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
.
Step 3
Select
Call Statistics
.
Table 7-6
explains the items displayed in the Call Statistics screen:
TX Retransmit
Number of packets that have been retransmitted to the expansion
module
TX Buffer Full
Number of packets discarded because the expansion module was not
able to accept new messages
Table 7-5
Expansion Module Statistics (continued)
Item
Description
Table 7-6
Call Statistics Items
Item
Description
Rcvr Codec
Type of voice stream received (RTP streaming audio from codec):
G.729, G.711 u-law, G.711 A-law, or Lin16k.
Sender Codec
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.
Rcvr Size
Size of voice packets, in milliseconds, in the receiving voice stream
(RTP streaming audio).