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User Guide for the Cisco Network Analysis Module (NAM) Traffic Analyzer, 5.0
OL-22617-01
Chapter 3 Monitoring and Analysis
Media
If you click on a call row in the table, in the RTP Streams for the Selected Call display at the bottom of
the page you will see all streams that are associated with the call. It will display the RTP streams that:
• have source address and port matched the call’s calling host address and calling port or called host
address and called port
• have destination address and port that matched the call’s calling host address and calling port or
called address and called port
Note There is a delay of two minutes of RTP streams statistics. As the result, there may not be any RTP stream
information of the call.
The RTP Streams of the Selected Call table shows the overall RTP streams statistics that are calculated
by the NAM. You can use this information to compare the views of the call endpoints and the NAM
regarding the call’s qualities. The columns of the RTP Stream are described in
Table 3-22.
Calling Host Address
RTP receiving address of the calling party detected by the NAM
from inspecting the call signaling protocol.
Calling Port
RTP receiving port of the calling party detected by NAM from
inspecting call signaling protocol.
Calling Alias
Calling party name detected by NAM from inspecting call signaling
protocol.
Called Host Address
IP address of the phone receiving the call.
Called Port
Port of the phone receiving the call.
Called Alias
Alias name, MGCP endpoint ID, or SIP URI of the called party
phone.
Calling Reported Jitter (ms)
Jitter value reported by calling party at the end of the call.
Calling Reported Packet Loss
(%)
Percentage of packet loss reported by calling party at the end of the
call.
Start Time
Time when the call was detected to start.
End Time
Time when the call was detected to end.
Duration
Duration of the call.
Note When the call signaling’s call tear down sequence is not
detected by the NAM, the NAM will assume:
- the call ended after 3 hours in low call volume per interval
- the call ended after 1 hour in high call volume per interval
(high call volume is defined as call table filled up during the
interval.)
Called Reported Jitter (ms)
Jitter value reported by called party at the end of the call.
Called Reported Pkt Loss (%)
Percentage of packet loss reported by called party at the end of the
call.
Table 3-21 Calls Table (continued)
Field Description