Cisco 7975G Administration Guide - Page 138
Call Statistics Screen, Settings, Status, Call Statistics
![]() |
UPC - 882658140457
View all Cisco 7975G manuals
Add to My Manuals
Save this manual to your list of manuals |
Page 138 highlights
Status Menu Chapter 7 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone 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 that are 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. The Call Statistics screen displays these items: 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.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). 7-12 Cisco Unified IP Phone 7975G Administration Guide for Cisco Unified Communications Manager 6.1 OL-14640-01
![](/manual_guide/products/cisco-7975g-administration-guide-f446b09/138.png)