Pinpoint VoIP call quality issues with smart diagnostics

NetIQ's Hafid Saba explains how to troubleshoot VoIP call quality in this exclusive tip.

When problems arise with VoIP phone calls, where do you get help with troubleshooting? This is one of the many challenges IT departments face when managing an IP Telephony-environment.

Poor call quality is not something that end users like dealing with and their explanation of the issue is often unhelpful (‘echoes’, ‘noise’, and ‘delays in the conversation’). As a voice or network engineer, this only adds to the confusion about how to start diagnosing the issue. Is it the IP Telephony system itself? Perhaps the network is having issues? Maybe the user is simply imagining an echo?

Often your VoIP calls are only as good as your weakest link. It only takes one router with an incorrect QoS configuration, or the dreaded half-duplex mode still enabled, to hinder an entire phone call.

Soft phones are another problematic factor. These are easy to deploy and cost much less than giving out fancy VoIP phones, however they’re susceptible to the same problems and can also be affected by the performance of the desktop or laptop they’re installed on.

The choice of codec used for audio compression levels is also important and will depend on the bandwidth available on the network. Common codecs include G.711u/a and G.729, the latter being compressed to an eighth of the size of a normal packet, which considerably reduces the margin for issues affecting call quality. This type of codec is ideal when using VoIP over small links.

Whether poor call quality is related to one of the above, or any number of other factors, NetIQ’s Vivinet Diagnostics application can be used to isolate and resolve problems. It conducts tests based on synthetic data created between two locations on the internal network. This is a good place to start, by identifying if the problem is internal or perhaps part of the WAN link (if there is one).

Vivinet Diagnostics achieves the best results when diagnosing IP Telephony solutions from Cisco, Nortel, Avaya or Microsoft, but the example discussed in this article will take a vendor-agnostic approach. By using software endpoints (probes on the network) to generate a synthetic call using the same codec and QoS tags that may be in use in the production voice system, it’s possible to execute a manual test to determine what degradation occurs in the phone call and where to start troubleshooting the issue.

It is strongly recommended that the PCs running the endpoint software sit as close as possible to the same VLAN, to give the most accurate depiction of all the hops that voice traffic traverse to get to their destination.

When looking at the main Diagnostics windows (below), you can define your call test and the IP addresses of the endpoints. However, if you provide the logon information for a Cisco Call Manager or Nortel CS1K, you can also target real phones.

Figure 1

Once you’ve chosen the appropriate codec and QoS tag (if used) running the diagnostic will automatically populate the trace of the call, as depicted in the following screenshot. Note: you will need the SNMP read strings to interrogate routers and switches.

Figure 2

Once the diagnosis is complete, the red Xs and warning signs mark potential issues on the network. Clicking on these objects and links reveals detailed information about the state of that device or connection.

Figure 3

Next we see information collected on the network devices and links.

Figure 4

Figure 5

The real guts of any call quality issue can be identified via detailed report views, which take all the issues determined by Diagnostics during its analysis (based on built-in knowledge) and present them to you in a prioritised list highlighting the most common issues first.

This information also shows the results of the synthetic call based on industry standard targets for call quality, as detailed below. This allows you to see not only the quality of the test call, but also all the possible issues that could have caused problems.

Metrics
Acceptable
Good
MOS (Mean Opinion Score)
>3.6
>4.08
R-Value
>70
>80
Delay
<400ms
<150ms
Jitter
<150ms
<60ms
Packet loss
<1%
<0.5%

The Diagnostics also take into consideration the fact that RTP (voice streams) traffic moves in one direction only, so two traces for each direction must be made to determine if there are any asynchronous routing issues.

The following screen is a summary of the problems identified in the report view.

This report could easily be considered the starting point for investigating where problems lie on your network. Let’s take a closer look at some of the common issues affecting VoIP calls today:

  1. Network impairments
    • Echoes in a call, words cutting out, delay in the conversation... these are all examples of how network impairments affect the call. Packet loss, end-to-end delay, and high congestion on ports are examples of where we see poor call quality.
    • Incorrect priority queuing for RTP streams
    • Jitter buffer loss running high
    • Half-duplex configuration
    • Congestion on the ports
    • Asynchronous routing
    • High link utilisation
  2. Soft phone considerations
    • Operating system performance
    • Resource constraints on the desktop
    • Insufficient bandwidth when connecting via VPN over slow links (public wi-fi locations, etc)
    • Poor packetisation delay
  3. WAN links
    • Incorrect QoS configurations
    • High congestion

These are only a sample of the issues that can affect phone call quality - many more factors come into play. However, finding a point to begin focusing on can help save considerable time and effort when solving in-depth problems. When it comes to VoIP, the time it takes to resolve issues is crucial, because you’re dealing with an application critical to the everyday function of just about every enterprise – voice communication.

Read more on Voice networking and VoIP

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