Cisco CP-7937G Administration Guide - Page 78

Call Statistics Screen, Applications, Settings, Status, Call Statistics

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Status Menu Chapter 7 Viewing Model Information, Status, and Statistics on the Conference Station 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. Table 7-2 Call Statistics Items Item Remote Address Local Address Start Time Codec Type Payload Size Rcvr Packets Rcvr Lost Packets Rcvr Octets Rx Expected Pkts Last Rx Seq No Most recent Rx SSRC Avg Jitter Max Jitter Description IP address and UDP port of the stream. IP address and UDP port of the conference station. Internal time stamp indicating when Cisco Unified Communications Manager 6.0 requested that the conference station start transmitting packets. 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. Size of voice packets, in milliseconds, in the receiving or transmitting voice stream (RTP streaming audio). 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. Missing RTP packets (lost in transit). Number of bytes of voice packets received since voice stream was opened. The expected number of packets received for the local conference station. The sequence number of the last RTP packet received. The Synchronization Source field of the last RTP packet received. Estimated average RTP packet jitter (dynamic delay that a packet encounters when going through the network) observed since the receiving voice stream was opened. Maximum jitter observed since the receiving voice stream was opened. Cisco Unified IP Conference Station 7937G Administration Guide for Cisco Unified Communications Manager 6.0 7-4 OL-11560-01 Rev. B0

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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
.
Table 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.