Tag Archives: VoIP analysis

Intermittent VoIP Issues Are a Big Deal to Customers

If during a VoIP call between two of your employees, the quality of the call begins to erode and jitter becomes a nuisance, your workers may simply shrug it off and wait until the problem passes. Even if a VoIP communication is dropped completely, the two employees are likely simply to reconnect and continue the conversation as if nothing much had happened.

Workers who spend five days a week together and have personal relationships with each other aren’t likely to get bent out of shape because of intermittent call difficulties. Customers, however, are a different story.

What effect do you think it would have on your relationship with a customer if after taking the time to reach out to your organization, his or her call was dropped 75 percent of the way through the interaction with one of your customer service agents? Probably the customer’s frustration in the face of having to call again—starting at the bottom of the queue—would prevent a callback from taking place and the customer’s issue from ever being resolved satisfactorily.

Your problems may actually go deeper than that, however, as recent research from New Voice Media found that 58 percent of Americans would never use a company again after a single negative customer experience.

So, how do you ensure top-notch call VoIP quality that helps keep your customers loyal? A network monitoring solution that includes tools for VoIP monitoring and analysis is a perfect place to start. Such a product can drill down specifically on VoIP calls to determine what may be causing intermittent issues and how different kinds of traffic are impacting each other. With this information, engineers can solve problems quickly as they pop up and make adjustments to avoid similar difficulties in the future.

So, put yourself in your customers’ shoes—what kind of call experiences are they getting? From there, you can decide your next course of action.

How to Overcome the Most Common VoIP Hurdles Network Engineers Face

Voice over Internet Protocol (VoIP) has become a total game-changer for businesses. In fact, the global VoIP services market – including residential and business VoIP service –totaled at upwards of $63 billion in 2012, according to research analyst Infonetics Research.

“The market for VoIP services has moved well beyond the early adopter stage to mainstream status in many developed countries,” says Diane Myers, principal analyst for VoIP, UC, and IMS at Infonetics Research.

Opposed to traditional phone systems, VoIP enables businesses to make and receive telephone calls over a broadband Internet connection. With VoIP, voice traffic is converted into data packets which are then transmitted over the public Internet or private IP network. The benefits of VoIP are unmatched – from reduced costs to increased flexibility to enhanced mobility.

However with businesses rolling out more end-user VoIP applications, network engineers need to be able to effectively analyze all network traffic, with a specific emphasis on VoIP, which presents a unique set of challenges. Let’s take a look at some of the obstacles:

  1. VoIP places pressure on network engineers to quickly resolve problems: Telephone calls are essential to businesses and today’s employees don’t have patience for interrupted, poor quality phone calls. When there are issues with the phone system, business can come to a screeching halt, which undoubtedly puts pressure on network engineers to quickly and effectively address problems.
  2. VoIP requires network engineers to rapidly become experts in telephone and voice technology: Protocol and network problems can tarnish VoIP services and, until recently, network engineers didn’t have the necessary network monitoring and analysis tools to find and fix faults on VoIP networks.
  3. VoIP makes it difficult for network engineers to diagnose problems on site: Most VoIP monitoring and troubleshooting products are hardware-based, expensive, and non-portable, which makes it difficult for engineers to diagnose problems on site. And when tools become too cost-prohibitive they are cut from the budget, leaving engineers without a solution to address VoIP complications.

To ease the pressure of network engineers having to maintain VoIP quality of service (QoS), WildPackets’ Network Analysis Solutions have the ability to monitor and analyze all network traffic, including voice and video over IP, allowing network engineers to determine when different types of traffic negatively affect each other. More specifically, OmniPeek Enterprise provides in-depth monitoring, analysis, and troubleshooting of both network and media traffic, eliminating the need for multiple analysis solutions. Network engineers are able to obtain a detailed analysis of latency, throughput, and network problems in a user-friendly display.

With quick access to this type of information, IT team members are able to quickly and effectively resolve issues, maintain a QoS experience, mitigate poor performance caused by competitions for network bandwidth, and monitor compliance with established network policies.

With more businesses adopting VoIP technology, network engineers must have a solution that enables them to effectively and quickly monitor, analyze, and troubleshoot problems in the real-time, ensuring that businesses get the type of voice quality they’ve come to expect.

To learn more about VoIP monitoring and WildPackets’ OmniPeek Enterprise solution, click here.

Tackling a 3-headed beast – Latency, Jitter, and Packet Loss – for VoIP and Video

VoIP monitoring and troubleshooting are now table stakes for any respectable network analysis system, and video monitoring is quickly becoming so as well as enterprise networks get bogged down under the weight of streaming video, often for extracurricular reasons. Fortunately VoIP and video over IP share an important characteristic – both are real-time protocols with unique network performance requirements. When your quest is problem-free VoIP and Video experience for your end users, you must tackle and slay Latency, Jitter, and Packet Loss.

You’re probably already familiar with latency. Ever watched the news and listened to a story being broadcasted from the other side of the world? As the signals get bounced from satellite to satellite and, ultimately, to your TV’s receiver, the delay might cause the news anchor and correspondent to interact out of sync. A person who is listening expects to hear the other’s voice in certain amount of time; when this doesn’t happen people start talking over each other. This can happen on your network as well. These delays cause the conversation to come off as unnatural and callers feel like they must “push to talk” or say, “over” to control the conversation and let the other person know that they have finished their thought.

The bright side is that packet-based analysis can measure latency, letting you know whether or not excessive latency is an issue on your network. This way, a network administrator can identify the problem in real-time and solve it before latency issues get worse. How long is too long? That’s somewhat dependent on each situation, but certainly latencies of 100 msec or longer are cause for concern, especially if they are common. In extreme cases, prolonged latencies can cause serious degradation to VoIP calls, including missing words or phrases.

Jitter doesn’t have the same simple, real world example as latency, but its effects are just as real. Jitter causes static and other audio anomalies, like stuttering, uneven audio and abnormal speech rhythm, in VoIP calls. Jitter is caused when the data packets that make up the VoIP call are not delivered at regular intervals to the receiver. Regular delivery of IP packets is required for the final digital to analog conversion at the receiver to work correctly. A typical receiver expects packets to be delivered every 20 msec, no more and no less. When the packets start to deviate from this expected delivery sequence, jitter happens.

Jitter can be even more detrimental in multimedia systems. With jitter, videos become jerky or irregular and very difficult to watch. If jitter levels become too high, packet loss can result, with a resulting loss of data.

Packet Loss
And speaking of packet loss, it is just as the name implies. Some packets, meaning some critical media data, never make it to the receiver. Packet loss causes missing sounds, syllables, words or phrases. DSP algorithms may compensate for up to about 30 ms of missing data, but anything more and the algorithms can’t compensate for the data loss.

Real-time protocols like VoIP and video are much more susceptible to packet loss than traditional network data, since there is very little cushion to wait for missing data or to put out-of-order data back into the right sequence. After about 150 msec or so, any data that is missing or out of order is essentially lost forever, since there is no way to properly reconstruct and maintain the real-time data stream, and this creates gaps (packet loss) in the data.

Gaps of more than 30 msec are noticeable to listeners. An average person speaks at a rate of 200 words per minute. That translates to about 3.33 words/sec = 300 ms per word. For G.711, we would need to lose 15 consecutive RTP packets to lose a whole word. Dropping 15 packets/sec for G.711 would be a loss rate of 30%, but losing only a few packets can still be very noticeable. As a general rule of thumb, loss of more than 2 consecutive packets will be heard. Loss rates > 2% will have a strong impact on quality. Losses of 5-10% make calls all but intolerable. Another good rule of thumb is that bursty periods of packet loss are worse than more dispersed loss.

These problems are very common and as more companies move their communication systems onto digital networks, they will only happen more frequently. It is important to have network monitoring and troubleshooting solutions that provide full visibility into all the types of data streaming on your network.