Cisco 7937G - Unified IP Conference Station VoIP Phone - Call Statistics Screen

7-4
Cisco Unified IP Conference Station 7937G Administration Guide for Cisco Unified Communications Manager 6.0
OL-11560-01 Rev. B0
Chapter 7 Viewing Model Information, Status, and Statistics on the Conference Station
Status Menu
Call Statistics Screen
The Call Statistics screen displays information about the last call on the conference station. Table 7-2
describes the information displayed on the screen.
Note You can remotely view the call statistics information by using a web browser to access the
Streaming Statistics web page. For more information about remote monitoring, see
Chapter 8,
“Monitoring the Conference Station 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, choose Applications >
Settings > Status > Call Statistics. To exit the Call Statistics screen, press Exit.
Ta b l e 7-2 Call Statistics Items
Item Description
Remote Address IP address and UDP port of the stream.
Local Address IP address and UDP port of the conference station.
Start Time Internal time stamp indicating when Cisco Unified Communications
Manager 6.0 requested that the conference station start transmitting
packets.
Codec Type Type of voice stream received or transmitted (RTP streaming audio): G.729,
G.711 u-law, G.711 A-law, G.722, G.722.1, or Lin16k.
Payload Size Size of voice packets, in milliseconds, in the receiving or transmitting voice
stream (RTP streaming audio).
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.
Rcvr Lost Packets Missing RTP packets (lost in transit).
Rcvr Octets Number of bytes of voice packets received since voice stream was opened.
Rx Expected Pkts The expected number of packets received for the local conference station.
Last Rx Seq No The sequence number of the last RTP packet received.
Most recent Rx SSRC The Synchronization Source field of the last RTP packet received.
Avg Jitter Estimated average RTP packet jitter (dynamic delay that a packet
encounters when going through the network) observed since the receiving
voice stream was opened.
Max Jitter Maximum jitter observed since the receiving voice stream was opened.