Cisco Systems OL-21011-01 manual Streaming Statistics

Chapter 8 Monitoring the Cisco Unified IP Phone Remotely

Streaming Statistics

•Core Dumps—Includes hyperlinks to individual dump files. The core dump files include data from a phone crash.

•Status Messages—Displays up to the 10 most recent status messages that the phone has generated since it was last powered up. You can also see this information from the Status Messages screen on the phone. Table 7-2describes the status messages that can appear.

•Debug Display—Displays debug messages that might be useful to Cisco TAC if you require assistance with troubleshooting.

Streaming Statistics

		A Cisco Unified IP Phone can stream information to and from up to three devices simultaneously. A
		phone streams information when it is on a a call or running a service that sends or receives audio or data.
		The streaming statistics areas on a phone’s web page provide information about the streams. Most calls
		use only one stream (Stream 1), but some calls use two or three streams. For example, a barged call uses
		Stream 1 and Stream 2.
		To display a Streaming Statistics area, access the web page for the phone as described in the Accessing
		the Web Page for a Phone, page 8-2, and then click the Stream 1, the Stream 2, the Stream 3, the
		Stream 4, or the Stream 5 hyperlink.
		Table 8-5describes the items in the Streaming Statistics areas.
Table 8-5		Streaming Statistics Area Items
	Item			Description
	Remote Address			IP address and UDP port of the destination of the stream.
	Local Address			IP address and UPD port of the phone.
	Start Time			Internal time stamp indicating when Cisco Unified Communications Manager requested
				that the phone start transmitting packets.
	Stream Status			Indication of whether streaming is active or not.
	Host Name			Unique, fixed name that is automatically assigned to the phone based on its MAC address.
	Sender Packets			Total number of RTP data packets transmitted by the phone since starting this connection.
				The value is 0 if the connection is set to receive only mode.
	Sender Octets			Total number of payload octets transmitted in RTP data packets by the phone since starting
				this connection. The value is 0 if the connection is set to receive only mode.
	Sender Codec			Type of audio encoding used for the transmitted stream.
	Sender Reports Sent1			Number of times the RTCP Sender Report have been sent.
	Sender Report Time Sent1			Internal time stamp indication when the last RTCP Sender Report was sent.
	Rcvr Lost Packets			Total number of RTP data packets that have been lost since starting receiving data on this
				connection. Defined as the number of expected packets less the number of packets actually
				received, where the number of received packets includes any that are late or duplicate. The
				value displays as 0 if the connection was set to send-only mode.
	Avg Jitter			Estimate of mean deviation of the RTP data packet inter-arrival time, measured in
				milliseconds. The value displays as 0 if the connection was set to send-only mode.
	Rcvr Codec			Type of audio encoding used for the received stream.
	Rcvr Reports Sent1			Number of times the RTCP Receiver Reports have been sent.
			Cisco Unified IP Phone Administration Guide for Cisco Unified Communications Manager 8.0 (SCCP and SIP)
	OL-21011-01				8-11