Today, we’re having a look at the best tools for monitoring VoIP networks. Voice over IP, or VoIP, is a relatively new technology that has been steadily gaining popularity these past several years. The technology, as its name implies allows the transmission of voice over IP networks. It gave rise to a whole new type of telephony systems that use the data network for connectivity. One big advantage of combining voice and data on the same network is that one no longer need two separate sets of cabling.
As a junior help desk attendant, I stumbled upon numerous instances of users complaining of their computers not connecting to the network only to find that they had connected it to the wrong jack. It was even worse with early laptops with built-in modems where users would use the phone jack when at home and the network jack when in the office. There was also the higher cost of keeping two separate sets of cabling. It thus quickly became obvious that carrying voice signal on data networks was advantageous.
But, contrary to data where chunks of data can arrive at their destination out of order and be reassembled into meaningful information, voice data needs to be transmitted in real time. It’s still broken down into chinks because this is how data networks work but those chinks must arrive in the right order and in a timely manner. In fact, VoIP requires that the network operate flawlessly and several aspects that are not so important with data transfers are of the utmost importance with voice traffic. For that reason, monitoring VoIP networks requires some specialized software that specifically watches all those important parameters.
We’ll start off today by discussing VoIP call quality. We’ll learn what factors can affect call quality. These are the parameters that should be monitored. We’ll also talk about MOS score, a system that the telephony industry has been using to quantify call quality. And finally, we’ll reveal what the best tools for monitoring VoIP networks are and give you a short review of each tool’s main features.
VoIP Call Quality
Before we go any further, let’s pause briefly and talk about call quality. The telephone was invented way back in the late nineteenth century and has constantly improved ever since. It’s gotten to a point where it’s taken for granted and every telephone user expects to experience a crystal clear conversation. And the fact that some phone conversations use VoIP rather than traditional telephony does not change that. The subjective experience that a phone user gets is usually referred to as call quality. It refers to how well speech can be heard and understood. A good quality call should have little or no static and, more importantly, no audio disruption.
Factors Affecting VoIP Networks And Call Quality
But with VoIP technologies, several factors can have an impact on call quality. Some of these factors are related to the technology choices while others have to do with operational characteristics of networks. Let’s see what the main quality-affecting factors are.
Codec stands for Coder-Decoder. This is the component that converts the analog speech into digital information and back for transmission via a data network. Different codecs can be used with different results. Some will offer better call quality at the expense of data size while others will keep the size down but sacrifice call quality. The highest quality codec commonly used with VoIP is the G.711 codec. Its high quality is achieved by not using any data compression, making it also the most bandwidth-hungry codec. This codec requires 90 Kbps of bandwidth for each conversation. G.729, G.726, and G.723 are also commonly used.
To a lesser extent, the available bandwidth also has an impact on call quality. For instance, chances are that using LAN switches with 1 Gbps ports will give you a better call quality than one using switches with 100 Mbps ports. Although a conversation uses less than 100 Kbps of bandwidth, it can, as you’ll soon discover as we explore other quality-impacting factors, suffer greatly from congestion. And since networks are rarely dedicated to VoIP, any overutilization will have an immediate impact on call quality.
In summary, networks should have enough bandwidth to support data AND voice traffic. Using QoS settings can allow one to reserve some bandwidth for voice and protect that traffic from other bandwidth hogs but the golden rule is still to ensure that there is enough bandwidth and that there is no congestion. This is why VoIP monitoring tools will often monitor bandwidth usage.
Jitter–or packet delay variation–refers to an irregularity in the delivery delay of data packets from their source to their destination. In an ideal network, transmitting each and every data packet from a given source to a given destination would take the exact same time. In reality, though, this is rarely the case and, for various reasons, there is often a variation in travel time between packets. This is what we call jitter.
Jitter affect call quality because VoIP data is transmitted in real-time and, although it will adapt to some degree of jitter, any major variation in transmission delay will cause the received speech to be chopped up. And chopped up audio is one of the most recognizable forms of bad call quality.
Latency refers to any delay that data suffers from on its path from source to destination. While data theoretically travel at the speed of light and should take a precisely defined time to go from one point to another, this is not true in real life. Several factors affect the transmission delays. Congestion and queuing are the most common ones but equipment overload can also be at fault.
While latency, if it is regular, has little impact on call quality per se, it can cause unnecessary delays which can be perceived as call quality issues. This is particularly true when latency reaches high values in the thousands of milliseconds. Because of that, latency is another parameter that should be closely monitored by VoIP monitoring tools.
Packet loss is most likely the worst enemy of VoIP. While data traffic will often recover from packet loss through a process where the receiving end will detect the missing data and request it to be sent again, this is not possible with VoIP traffic because it’s all happening in real time. Any lost packet will translate into lost audio. If it happens occasionally, it might go unnoticed but a high level of packet loss will have an enormously detrimental effect on call quality. Concretely, users will be missing some part for the conversation.
MOS Score — An “Objective” Evaluation Of Call Quality
Call quality is a highly subjective concept. For instance, one user can find some degree of audio degradation acceptable while another might insist on crystal-clear audio. In order to add some degree of objectivity to an otherwise subjective concept, the telephony industry has come up with the concept of a Mean Opinion Score or MOS.
Mean Opinion Score gives VoIP testing a number value as an indication of the perceived quality of received voice after being transmitted and compressed using codecs. This measurement is the result of underlying network attributes that act upon data flow and is useful in predicting call quality and in determining issues that can affect VoIP quality.
MOS values range from 1 to 5 with values of 4 or higher indicating a generally satisfying call quality. Values between 3 and 4 generally indicate some level of insatisfaction among users while values below 3 indicate a mostly unsatisfactory call quality. Some of the best VoIP monitoring tools are able to calculate the MOS value.
The Best Tools For Monitoring VoIP Networks
Now that we have a better understanding of how monitoring VoIP networks differ from generic network monitoring, we’re ready to have a look at some of the best tools we could find. Some tools on our list are dedicated VoIP tools while others are multi-purpose network monitoring tools that either have built-in VoIP monitoring functionalities or that have add-on modules offering the functionality.
SolarWinds is a well-known name amongst network administrators. The company has been making some of the best network administration tools for the past 20 years. Its flagship product, the Network Performance Monitor is an SNMP monitoring platform that consistently scores among the best on the market. The company is also famous for its many free tools. Made to address specific needs of network administrators, they include a TFTP server or a subnet calculator, just to mention a few.
The SolarWinds VoIP and Network Quality Manager is a dedicated VoIP monitoring tool that is packed with great features. This tool can be used to monitor VoIP call quality metrics, including jitter, latency, packet loss, and MOS. It can also be used to troubleshoot VoIP call performance by correlating call issues with WAN performance. The system also offers real-time WAN monitoring is using Cisco IP SLA technology. Its visual VoIP call path trace feature lets you see and pinpoint call problems along the entire network path.
Setting up the SolarWinds VoIP and Network Quality Manager is easy and can be accomplished with just a few mouse clicks. The system automatically discovers Cisco IP SLA-enabled network devices, and typically deploys in less than an hour. And once it’s up and running, it provides a very deep insight into your VoIP networking environment.
This tool provides real-time monitoring of site-to-site WAN performance and it also has alerting features to notify you of any abnormal situation. It can help ensure that WAN circuits are performing as expected by utilizing Cisco IP SLA metrics, synthetic traffic testing, and custom performance thresholds and alerts. It also has visual VoIP call patch trace, an invaluable troubleshooting tool.
The SolarWinds VoIP and Network Quality Manager won’t only monitor your WAN circuits, it can also display the utilization and performance metrics of your VoIP gateways and PRI trunks. It can help with capacity planning by allowing you to measure voice quality in advance of new VoIP deployments.
Price for the SolarWinds VoIP and Network Quality Manager start $1 615 for up to 5 IP SLA source devices and 300 IP phones. Other licensing levels–including a device-unlimited license–are also available. And like with most SolarWinds tools, a free 30-day trial is available should you want to test the product before committing to purchasing it.
- FREE TRIAL: SOLARWINDS VOIP AND NETWORK QUALITY MANAGER
- Official download site: https://www.solarwinds.com/voip-network-quality-manager
The Paessler Router Traffic Grapher, or PRTG, is a well-known network monitoring system. It does much more than just monitor bandwidth usage, though. Through the use of sensors, PRTG can be used to monitor many different parameters of networks and systems. The tool can monitor any system, device, traffic, and application in your IT infrastructure. Two specific sensors are available for Monitoring VoIP networks. The QoS sensor measures parameters such as UDP packet loss, jitter, Ethernet latency, etc. As you’ll recall, these are important parameters for VoIP networks. For IP-SLA enabled Cisco devices, there is an IP-SLA sensor, which reads similar metrics from Cisco devices. Both methods show you the quality of your VoIP connection and enable you to define what level of latency, jitter, etc. are acceptable. Whenever the threshold is exceeded, you can be notified and take appropriate measures to address the situation. Notifications can be sent via email or SMS or pushed to a mobile device using the free client app available for Android, iOS and Windows Phone.
Paessler claims that you could start monitoring with PRTG within a couple of minutes of starting the installation. The tool’s auto-discovery system will scan network segments and automatically recognize a wide range of devices and systems. It will then create sensors from predefined device templates. Specific VoIP sensors will then need to be set up, making the installation a bit longer but this is still one of the fastest tools to set up.
PRTG is available in a free, full-featured version which is limited to 100 sensors. Withing PRTG, any monitored parameter counts as one sensor. For example, monitoring the bandwidth on each port of a 48-port switch will count as 48 sensors. To monitor more than 100 sensors, you’ll need to purchase a license. You’d also need one QoS or one IP-SLA sensor for each VoIP networking devices you want to monitor. Price increases with the number of sensors and starts at $1 600 for 500 sensors up to $14 500 for unlimited sensors. A free device-unlimited 30-day trial version is available.
3. ManageEngine OpManager
The ManageEngine OpManager is another one of the best-known network monitoring tools. It will monitor the vital signs of your servers (physical and virtual) as well as your network equipment and alert you as soon as something is out of specs. The tool features an intuitive user interface that will let you easily find the information you need. The product also features an excellent reporting engine along with some pre-built as well as custom reports. To complete the package, this system’s alerting features are also very complete.
And when it comes to VoIP monitoring, the ManageEngine OpManager‘s VoIP monitor option seamlessly integrates with OpManager to proactively monitor and report on your infrastructure’s capacity to handle VoIP calls. The tool uses Cisco IP SLA to continuously monitor critical Quality of Service parameters of VoIP networks. The monitored VoIP parameters include packet loss, delay, jitter, the Mean Opinion Score (MOS) and Round Trip Time (RTT).
The ManageEngine OpManager is priced based on the number of monitored devices. Prices range from $715 for 25 devices to $14 995 for 1000 devices. The VoIP monitoring option adds $125 per device that requires it. Like with most full-featured commercial monitoring tools, a free 30-day trial is available.
VoIPmonitor is an open source network packet sniffer with a commercial frontend for monitoring most VoIP protocols. The tool, which runs on Linux, is designed to analyze the quality of VoIP calls based on network parameters such as delay variation (jitter) and packet loss according to the ITU-T G.107 E-model which predicts quality using the MOS scale. Call information, together with relevant statistics, are saved to a MySQL database. Optionally each call can be saved to a pcap file (a file capture format that can be opened with other analysis tools such as Wireshark) with either only SIP protocol or SIP, RTP, RTCP, T.38, and udptl protocols. VoIPmonitor can also decode speech and play it over its WEB GUI as well as save it to disk as a .WAV file. It natively supports the G.711 alaw and ulaw codecs and commercial plugins add support for G.722, G.729a, G.723, iLBC, Speex, GSM, Silk, iSAC, and OPUS. VoIPmonitor is also able to convert T.38 FAX to PDF.
The VoIPmonitor GUI front end is available either as a locally hosted server at prices ranging from $42/month for 10 channels to $917/month for 6000 channels or as a cloud-based service with prices varying from $20/month for 3 channels to $200/month for 200 channels. Both versions are available in a free and unlimited 30-day trial.
VQmon/EP is different from other VoIP monitoring tools in that it is integrated into your devices. It claims to be the most widely used technology for monitoring the quality and performance of live VoIP calls. The system is integrated into a range of IP phones sold by Avaya, Mitel, Polycom, Cisco, and several other manufacturers. It provides built-in support for the industry-standard SIP QoE (RFC 6035) and RTCP XR (RFC 3611) reporting protocols, allowing network administrators to monitor call quality everywhere within their network without the use of probes. VoIP Spear
VQmon/EP can detect packet loss and jitter buffer discard events. It can also extract key information from DSP software and produce real-time call quality scores and diagnostic data. This tool generates listening and conversational quality MOS scores and R factors as well as a wide range of diagnostic data. Furthermore, VQmon/EP features real-time call quality thresholds, supporting either alert generation or automatic configuration.