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Copyright © 2010 Cisco Systems, Inc. This document is Cisco Public Information.
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Figure 1. Cisco NAM Deployment
NAM can be deployed in various locations depending on the use case being addressed. For example, when
deployed in a branch, NAM can monitor the quality of all calls entering and exiting that branch office and monitor any
dips in quality levels in that location. Alternatively, a NAM in the data center can monitor signaling messages sent
from a phone to the Cisco Unified Communications Manager cluster and collect detailed information about the
calling and called parties. A NAM located at the edge of the main campus can raise alarms about poor call quality
from the main campus to a particular remote office location.
How Voice Monitoring Works
VoIP phone calls are set up using a signaling protocol, such as Session Initial Protocol (SIP), using which the phone
endpoints exchange information. Once setup is complete, voice traffic is sent through a streaming protocol such as
RTP. NAM detects both signaling messages and voice traffic over RTP streams. It also has the ability to link a set of
call setup messages to the RTP streams associated with them.
NAM monitors signaling messages for Skinny Call Control Protocol (SCCP), SIP, H.323, and Media Gateway Control
Protocol (MGCP). The information collected from signaling messages includes currently active calls, call statistics,
call history, detailed information about the calling and called parties, codecs used, port numbers used, and other
relevant information. These statistics can be collected both on a real-time and historical basis.
In order to measure the quality of the call, NAM detects and monitors RTP streams. First, NAM examines the packet
header and identifies whether it is an RTP packet. If so, it checks whether the packet belongs to a new or existing
RTP stream. Once the RTP packet is detected and associated with a stream, it is sent to the MOS process for
quality analysis. The MOS process performs real time computations to measure voice quality metrics such as jitter,
actual packet loss, adjusted packet loss, seconds of concealment, and severe seconds of concealment. Using the
aforementioned metrics, the NAM computes the R-Factor MOS based on the ITU-T recommendation G.107. The
best, worst, and average values for these metrics are reported every minute through GUI.
An important aspect of this real-time reporting is that voice quality metrics are available to users even when the call
is active. There is no need to wait for the end of the call before such statistics are collected. This real-time visibility is
a critical part of the solution and facilitates rapid responses to problems.