Cisco Systems 8961 manual Call Statistics Screen, Select Status, Select Call Statistics, 10-11

Chapter 10 Viewing Model Information, Status, and Statistics on the Cisco Unified IP Phone

Status Menu

Call Statistics Screen

You can access the Call Statistics screen (see Table 10-5) on the phone to display counters, statistics, and voice-quality metrics of the most recent call.

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 11, “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 latest voice stream, follow these steps:

Procedure

Step 1 Press the Applications button.

Step 2 Select Administrator Settings.

Step 3 Select Status.

Step 4 Select Call Statistics.

The Call Statistics screen displays these items:

Table 10-5 Call Statistics Items for the Cisco Unified Phone

Item	Description
Rcvr Codec	Type of voice stream received (RTP streaming audio from codec):
	G.729, G.722, G.711 u-law, G.711 A-law, and iLBC.
Sender Codec	Type of voice stream transmitted (RTP streaming audio from codec):
	G.729, G.722, G.711 u-law, G.711 A-law, and iLBC.
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.

	Cisco Unified IP Phone 8961, 9951, and 9971 Administration Guide for Cisco Unified Communications Manager 8.5 (SIP)
OL-20861-01		10-11