Streaming Statistics Area - Cisco 8831 Administration Manual

Remote monitoring

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
The following table describes the items in the Streaming Statistics areas.
Table 33: Streaming Statistics area 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
section, and then click the Streaming Statistics hyperlink.
Description
IP address and UDP port of the stream.
IP address and UDP port of the conference phone.
Internal time stamp indicating when Cisco Unified Communications Manager
requested that the conference phone 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, 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.
This number is not necessarily identical to the number of RTP voice
Note
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 phone.
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 Phone 8831 Administration Guide for Cisco Unified Communications Manager 9.0

Streaming Statistics area

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