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Configuring and Troubleshooting VoIP Monitoring
14
November 2013
The challenge then is to determine the best location to tap into this network to see the
audio packets. The most reliable method is to have the phone itself send the audio
streams to your software. This is possible with the Unified CM Recording and
Monitoring feature. This feature, which is available on Unified CM 6.0 and later, allows
the Unified CM to send a command to the built-in bridge (BIB) on the agent's phone to
have it duplicate the two audio streams and send them over the network to another
location. For recording, the two streams are sent to two ports on a recording server.
For live monitoring, the two streams are merged into one and sent as a new call to a
supervisor's phone, thus allowing the supervisor to listen to the call. The mechanics of
this feature are shown below.
audio packets. The most reliable method is to have the phone itself send the audio
streams to your software. This is possible with the Unified CM Recording and
Monitoring feature. This feature, which is available on Unified CM 6.0 and later, allows
the Unified CM to send a command to the built-in bridge (BIB) on the agent's phone to
have it duplicate the two audio streams and send them over the network to another
location. For recording, the two streams are sent to two ports on a recording server.
For live monitoring, the two streams are merged into one and sent as a new call to a
supervisor's phone, thus allowing the supervisor to listen to the call. The mechanics of
this feature are shown below.
The next most reliable location to tap into the network is at the IP phone itself. It is at
this point that we know the data will be flowing to or from the phone over a single
cable. The further away from the IP phone (toward the network cloud) we go, the more
complex is the solution to accessing the audio streams.
this point that we know the data will be flowing to or from the phone over a single
cable. The further away from the IP phone (toward the network cloud) we go, the more
complex is the solution to accessing the audio streams.
Most Cisco IP phones contain another network connection point where a computer
can be daisy-chained. This allows software running on the attached PC to see the
audio traffic. This method is referred to as “desktop monitoring” and is discussed in
more detail below.
can be daisy-chained. This allows software running on the attached PC to see the
audio traffic. This method is referred to as “desktop monitoring” and is discussed in
more detail below.
If the IP phone does not support daisy-chaining, or if policy dictates that this
configuration is not supported, the next access point away from the phone is the
switch to which the phone is connected.
configuration is not supported, the next access point away from the phone is the
switch to which the phone is connected.
NOTE: The phone must be directly connected to the switch. It cannot
be connected to a hub, router, or gateway.
be connected to a hub, router, or gateway.
The Cisco Catalyst line of switches supports a feature called Switched Port Analyzer
(SPAN), or port monitoring, that allows network traffic flowing through a particular
switch port or group of ports to be copied and sent to a destination port. Software
listening on this destination port can then get access to packets containing audio data
representing a phone call. This method of packet capture is known as server
monitoring.
(SPAN), or port monitoring, that allows network traffic flowing through a particular
switch port or group of ports to be copied and sent to a destination port. Software
listening on this destination port can then get access to packets containing audio data
representing a phone call. This method of packet capture is known as server
monitoring.
Figure 6.
Audio stream access points and solution complexity