Chapter 3 Monitoring and Analysis

 

 

 

 

 

Media

Table 3-21

Calls Table (continued)

 

 

 

 

 

Field

 

Description

 

 

 

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.

 

 

 

 

 

 

 

 

 

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.

 

 

User Guide for the Cisco Network Analysis Module (NAM) Traffic Analyzer, 5.0

 

 

 

 

 

 

OL-22617-01

 

 

3-41

 

 

 

 

 

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Cisco Systems NAM From inspecting the call signaling protocol, Inspecting call signaling protocol, Protocol, Phone, Call