7-14
Cisco Unified IP Phone 7965G and 7945G Administration Guide for Cisco Unified Communications Manager 6.0
OL-12650-01
Chapter 7 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone
Status Menu
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.
The Call Statistics screen displays these items:
Table 7-6 Call Statistics Items
Item Description
Rcvr Codec 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.
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).
Sender Size Size of voice packets, in milliseconds, in the
transmitting voice stream.
Rcvr Packets 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.
Sender Packets 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.