1.1 Overview
Voice over Internet Protocol (VoIP) technology is
an
effective and
efficient alternative to conventional phone calls. With most
enterprises embracing this technology, the VoIP network managers are
challenged with the task of ensuring a high quality service realtime and
also be able to assess the VoIP performance by looking at the performance
history. NetFlow Analyzer offers a comprehensive solution to managing your VoIP
network in addition to continuously monitoring your WAN and LAN for
performance. Intuitive dashboards, drill-down reports, hop graphs, and
meaningful alerts, helps a VoIP administrator in getting to the root of the
problem in no time. Measure and track the performance of voice quality
across Wide Area Networks (WANs) using NetFlow Analyzer.
NetFlow Analyzer
monitors your VoIP network and helps you achieve the following:
-
Proactive Monitoring : Visualize potential bottlenecks
proactively and eliminate choppy calls by identifying abnormal packet
delays by actively monitoring simulated VoIP packets round the clock.
-
Capacity Planning : Intuitive trend reports helps you
implement effective capacity planning measures resulting in an increase
of the VoIP quality and also better ROI.
-
Easy Troubleshooting with Netflow Reports : Provision to pin-down
bandwidth issues in the VoIP links with integrated Netflow Analyzer
Bandwidth usage graphs.
-
Meeting SLAs : And most importantly, keep up the SLA by
configuring meaningful threshold limits to deliver uninterrupted service.
NetFlow Analyzer continuously monitors the
key performance metrics of the VoIP network to determine its health. The
parameters measured include Jitter, Latency, Packet Loss etc.
Jitter: It is one of the key parameters to measure the voice
and video quality over an IP Network. Jitter indicates a variation in
delay between arriving packets (inter-packet delay variance). Users often
experience uneven gaps in speech pattern of the person talking on the
other end, and sometimes there are disturbing sounds over a conversation
coupled with loss of synchronization etc. This is reffered to as Jitter.
Latency: Another key parameter that is used to measure the
voice and video quality over the IP Network is the delay or latency. The
delay measured is the time taken for a caller's voice at the source site
to reach the other caller at the destination site. Network latency
contributes to delay in voice transmission, resulting in huge gaps between
the conversation and interruptions.
Packet Loss : Packet loss is a measure of the data lost
during transmission from one resource to another in a network. Packets are
discarded often due to network latency. Using NetFlow Analyzer, you can monitor
the packet loss and take corrective actions based on the information.
MOS : The jitter codec determines the quality of VoIP
traffic and each codec provides a certain quality of speech. The Mean
Opinion Score is a standard for measuring voice codecs and is measured in
the scale of 1 to 5 (poor quality to perfect quality). The quality of
transmitted speech is a subjective response of the listener.
2.1 How it works ?
NetFlow Analyzer primarily relies on
Cisco's
IP-SLA for monitoring the VoIP and the prerequisite therefore is, that
the device should be a Cisco Router and must have IPSLA agent enabled
on it. From IOS Version 12.3(14)T all Cisco routers support monitoring of
VoIP QoS metrics.
Cisco's IPSLA, an active monitoring feature of Cisco IOS software,
facilitates simulating and measuring the above mentioned parameters to
ensure that your SLAs are met.
Cisco IP SLA provides a UDP jitter operation where UDP packets are sent from
the source device to a destination device. This simulated traffic is used to
determine the jitter, the round-trip-time, packet loss and latency. This
data is gathered for multiple tests over a specified period to identify how
the network performs at different times in a day or over a few days. The
VoIP monitor gathers useful data that helps determine the performance of
your VoIP network, equipping you with the required information to perform
network performance assessment, troubleshooting, and continuous health
monitoring. Look at the following diagram to understand the workflow:
3.1 How to set up a VoIP
Monitor?
3.1.1 Prerequisites
When you want to test a link from your office to another location, you
need a Cisco router ( IOS version 12.4 or later ) at each end.
3.1.2 Steps to set up the monitor
Using NetFlow Analyzer, you can now
monitor the voice and video quality of a 'call path'. Call path is the WAN
link between the router in your main office and the one in the branch office
that you want to monitor.
Step 1 :
Export NetFlow from the
router in your LAN to NetFlow Analyzer. And make sure the snmp read and write
community are configured properly, for that router.
Step 2: Enable SLA responder on the
destination device you wish yo monitor, Steps are detailed below.
-
Open a CLI session on the destination router and enable the EXEC mode as
follows:
Router>enable
-
Start the global configuration mode:
Router#configure terminal
-
Enable the IP SLA responder:
Router(config)#ip sla responder
-
Repeat the above steps for all the destination routers on which you want
to monitor VoIP performance.
Step 3: Selecting the Call Path ( Source and Destination
Router ) to be monitored:
-
Go to Home > VoIP Monitors >Configure VoIP Monitor > Create
New, and enter a name for the monitor.
-
Select the source router from the list of routers discovered in
NetFlow Analyzer, and select the relevant interface.
-
Specify the destination router either by using the 'Search' option to
pick from the discovered routers, or use the 'Add'specify the IP address
of the destination router and submit the details.
-
You will see the summary of the monitor you are about to configure. Now
click 'Apply to device' to sumbit the details to the device. This will
take few seconds to configure.
Refresh the page after few seconds to see the new monitor. The data will
be collected every hour, from the time you have configured.
To edit any of
the configuration details, go to the respective template, make the changes
and save the details. When you create a new monitor, the updated values take
effect. When the configuration is complete, the router starts collecting the
data at the specified frequency 60 seconds ( default value). NetFlow Analyzer
updates this statistics (collected data) every hour and the reports are
generated after one hour of configuration. Go through the FAQs section
to understand QoS parameters.
4.1 How to customize
Threshold Limits ?
4.1.1 Defining Call Settings:
Define a template with the required VoIP settings to be used for monitoring
performance. The VoIP template comes with pre-populated default values.
Incase you would like to effect some changes to the values before initiating
monitoring, make the changes as follows:
-
From Modules > VoIP Monitors > Settings > go to 'Call Settings'
-
Configure the following parameters:
Destination Port - Specify the VoIP UDP port to which VoIP Monitor
sends simulated traffic to generate performance metrics. The default port
number is set as 16384. You can specify a port in the range of 16384 32766.
Simulated VoIP Codec - The VoIP jitter codec decides the type of
traffic that VoIP Monitor simulates over your network.
Operation Frequency - The operation frequency is the frequency with
which QoS metrics are collected by the IP SLA agent on your network to
determine performance.
Operation Timeout - The operation timeout is time to wait for the
response from the responder / destination device in msecs.
Type of service - The Type of Service octet allows you to set
precedence levels for VoIP traffic of the IP SLA operations.
MOS Advantage Factor - The advantage factor is a measure, on a scale
of 0 to 20, of the willingness of your VoIP network users to trade call
quality for convenience
4.1.2 Defining Thresholds for the monitored parameters:
You can define a threshold template so that the VoIP
performance parameters can be better suit your company SLA's (Service Level
Agreements). Alerts are triggered based on the thresholds configured so that
you can take corrective actions in time. Here are the steps to define a
threshold template:
-
Go to Modules > VoIP Monitors > Settings > go to ' Threshold
Template'
-
Configure the following values:
MOS Threshold : Configure the MOS threshold by specifying the upper
and lower MOS range values in the range of 1 to 5.
Jitter Threshold : Configure the jitter threshold in msecs with
upper and lower threshold limites. The range is from 0 to 6000 msecs.
Latency Threshold : Specify the delay allowed in msecs again in the
range of 0 to 6000.
Packet Loss : Specify the number of packets that can be lost in
transit.
5.1 FAQs
-
Why
do i need to set SNMP write community on the Source Router ?
-
Why
should the IP SLA Responder be enabled on the destination device ?
-
Why
are the VoIP metrics shown as zero or 'Not available' in NetFlow Analyzer?
-
What
are all the VoIP QoS metrics measured by NetFlow Analyzer ?
-
What is VoIP
codec?
-
How
much bandwidth does each monitor occupy ?