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Streaming Statistics area
A conference phone can stream information to and from up to three devices simultaneously. A conference
phone streams information when it is on a call or running a service that sends or receives audio or data.
The Streaming Statistics area on a conference phone 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 phone as described in the
Access web page
section, and then click the
Streaming Statistics
hyperlink.
The following table describes the items in the Streaming Statistics areas.
Table 33: Streaming Statistics area items
Description
Item
IP address and UDP port of the stream.
Remote Address
IP address and UDP port of the conference phone.
Local Address
Internal time stamp indicating when Cisco Unified Communications Manager
requested that the conference phone start transmitting packets.
Start Time
Type of voice stream received or transmitted (RTP streaming audio): G.729,
G.711 u-law, G.711 A-law, G.722, or Lin16k.
Codec Type
Size of voice packets, in milliseconds, in the receiving or transmitting voice stream
(RTP streaming audio).
Payload Size
Number of RTP voice packets received since voice stream was opened.
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.
Note
Rcvr Packets
Missing RTP packets (lost in transit).
Rcvr Lost Packets
Number of bytes of voice packets received since voice stream was opened.
Rcvr Octets
The expected number of packets received for the local conference phone.
Rx Expected Pkts
The sequence number of the last RTP packet received.
Last Rx Seq No
The Synchronization Source field of the last RTP packet received.
Most recent Rx SSRC
Estimated average RTP packet jitter (dynamic delay that a packet encounters
when going through the network) observed since the receiving voice stream was
opened.
Avg Jitter
Maximum jitter observed since the receiving voice stream was opened.
Max Jitter
Cisco Unified IP Conference Phone 8831 Administration Guide for Cisco Unified Communications Manager 9.0
121
Remote monitoring
Streaming Statistics area