Monitor VoIP using NetFlow Analyzer





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:


        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 1Export 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.


  1. Open a CLI session on the destination router and enable the EXEC mode as follows: 
                
    Router>enable

  2. Start the global configuration mode:

    Router#configure terminal

  3. Enable the IP SLA responder:

    Router(config)#ip sla responder

  4. 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:

  1. Go to Home > VoIP Monitors >Configure VoIP Monitor > Create New, and enter a name for the monitor.
  2. Select the source router from the list of routers discovered in NetFlow Analyzer, and select the relevant interface.
  3. 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.
  4. 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:


  1. From Modules > VoIP Monitors > Settings > go to 'Call Settings'
  2. 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:


  1. Go to Modules > VoIP Monitors > Settings > go to ' Threshold Template'

  2. 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

 
  1. Why do i need to set SNMP write community on the Source Router ?
  2. Why should the IP SLA Responder be enabled on the destination device ?
  3. Why are the VoIP metrics shown as zero or 'Not available' in NetFlow Analyzer?
  4. What are all the VoIP QoS metrics measured by NetFlow Analyzer ?
  5. What is VoIP codec?
  6. How much bandwidth does each monitor occupy ?


1. Why do i need to set SNMP write community on the Source Router ?

Both, the SNMP read and write community string needs to be set on the source router. The write community is used to configure the IPSLA on the device while the read community is used by NetFlow Analyzer to gather performance data from the router.

2. Why should the IP SLA Responder be enabled on the destination device ?

Enabling the IP SLAs Responder provides the details of packet loss statistics on the device sending IP SLAs operations. IP SLAs Responder is enabled on the target router (rtr responder) before configuring a Jitter operation.

3. Why are the VoIP metrics shown as zero or 'Not available' in NetFlow Analyzer?

You will see zero or 'not available' values when data is not collected for the monitored metrics. This can be either due to incorrect SNMP read community configured, or of the Responder is not enabled on the destination device. Make sure that the correct SNMP read community is configured and the SLA Responder is enabled.

4. What are the critical parameters monitored to determine the VoIP QoS performance?

The monitored parameters include Latency, Jitter, Packet Loss, and MOS. The parameters are described below for reference:

Jitter : Jitter is defined as a variation in the delay of received packets. Users often experience disturbing sounds over a conversation coupled with loss of synchronization at times and is reffered to as jitter. High levels of jitter can result in some packets getting discarded and thereby impact the call quality. Ensuring a jitter-free transmission to provide qualitative service depends on identifying the bottle-neck responsible for the jitter, and acting on it to eliminate it. NetFlow Analyzer's VoIP monitoring feature helps you find the problem and ensures maximum QoS on your VoIP network.

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.

One way Latency: Latency (delay) is the time taken for a packet to reach the destination device. When monitoring latency over VoIP, 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.

Round Trip Time: Round Trip Time is the time taken for a packet to reach the destination and again comes back to the source device. The total time it takes for the round trip is measured in milliseconds.

MOS: The Mean Opinion Score is the key quality indicator of VoIP traffic quality. And is measured in the scale of 1 to 5 (poor to excellent quality).


5. What is VoIP codec?


Codecs (Coder/Decoder) serve to encode voice/video data for transmission across IP networks. The compression capability of a codec facilitates saving network bandwidth and it is therefore appropriate that you choose the correct codec for your IP network. Here is a quick reference to the codecs with the corresponding packets size and bandwidth usage:
 

Codec & Bit Rate (Kbps)

Operation Frequency

Default number of packets

Voice Payload Size

Bandwidth
MP or FRF.12
(Kbps)

Bandwidth
w/cRTP MP or FRF.12
(Kbps)

Bandwidth
Ethernet
(Kbps)

G.711a/u
(64 kbps)

60 msecs by default. You can specify in the range of 0 - 604800 msecs.

1000

160 + 12 RTP bytes

82.8 kbps

67.6

87.2

G.729
(8 kbps)

1000

20 + 12 RTP bytes

26.8 kbps

11.6

31.2



6. How much bandwidth does each monitor occupy ?

The bandwidth occupied depends on the codec selected. Look at the above table for reference.





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