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

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Status Menu

Call Statistics Screen

The Call Statistics screen displays information about the last call on the conference station.
describes the information displayed on the screen.
Note
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
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
Cisco Unified IP Conference Station 7937G Administration Guide for Cisco Unified Communications Manager 6.0
7-4
Chapter 7
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
"Monitoring the Conference Station Remotely."
Call Statistics Items
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
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.
Viewing Model Information, Status, and Statistics on the Conference Station
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.
Table 7-2
Chapter 8,
OL-11560-01 Rev. B0

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