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Cisco 7937G - Unified IP Conference Station VoIP Phone - Device Logging; Streaming Statistics

Cisco 7937G - Unified IP Conference Station VoIP Phone
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8-8
Cisco Unified IP Conference Station 7937G Administration Guide for Cisco Unified Communications Manager 6.0
OL-11560-01 Rev. B0
Chapter 8 Monitoring the Conference Station Remotely
Device Logging
Device Logging
The Device Logging area on a conference station’s web page provides information you can use to help
monitor and troubleshoot the conference station. It includes debug and error messages received on the
conference station that might be useful to Cisco TAC if you require assistance with troubleshooting.
To display device logs, access the web page for the conference station as described in the
Accessing the Web Page for a Conference Station” section on page 8-2, and then click the
Device Logging hyperlink. In the File Download dialog box, click Open to view the device logs, or click
Save to save the logs to a specific location.
Streaming Statistics
A conference station can stream information to and from up to three devices simultaneously. A
conference station streams information when it is on a a call or running a service that sends or receives
audio or data.
The Streaming Statistics area on a conference station’s web page provides 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 the Streaming Statistics area, access the web page for the conference station as described in
the
Accessing the Web Page for a Conference Station” section on page 8-2, and then click the
Streaming Statistics hyperlink.
Table 8-4 describes the items in the Streaming Statistics areas.
Ta b l e 8-4 Streaming Statistics Area 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 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.

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