Got VoIP? Then Monitor These Key Network Metrics
How jitter, latency, and packet loss impact VoIP quality of service
June 16, 2015

Patrick Carey
Exoprise

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Chances are you either have already or are contemplating replacing your old POTS/PBX telephone system with some flavor of Voice of IP (VoIP). With so many solutions available, you may even be thinking about going beyond a simple phone system replacement and upgrading to a rich unified communications suite, complete with real-time voice, video, screen sharing, and collaborative document editing.

The good news is that you don’t even need to have a seven-figure IT budget to get all these features. Pay-as-you-go SaaS offerings like Skype for Business, GoToMeeting, join.me, BlueJeans, WebEx and others have put unified communications in reach of every organization. But before you cut the phone cord you need to understand how your network performance (or lack thereof) can make or break your user experience and productivity with these tools.

Quality of Service Metrics for Unified Communications

We’ve all suffered through live meetings or VoIP calls where the audio and/or video was bad. A few extra seconds in a webpage load might go unnoticed but when audio and video is choppy, delayed, or has gaps, it can be excruciating.

To ensure Quality of Service (QoS) for VoIP and unified communications you need monitor and manage three basic network metrics:

Latency – how much time it takes for data packets to get from one designated point to another.

Jitter – the variation in in latency over time.

Packet Loss – the percentage of packets that never reach their intended destination.

Let’s take a quick look at the effects, root causes, and remediation techniques for each metric.

Latency

Ever been on a conference call with somebody in an adjacent office or cubicle? You end up hearing everything they say twice, first directly from them and (hopefully) through your headset or speakers a short time later. That’s latency. There’s always some, but when the round trip latency goes above about 250ms we start to notice it – awkward pauses and people talking over each other. Some latency is out of your control (e.g. for calls that traverse the internet), but you should aim for internal network latency levels of less than 150ms.

Latency problems are usually the result in packet handling and queuing delays; essentially packets get bogged down at your network nodes, either because the network is not configured to expedite delivery of VoIP/UC traffic or there is insufficient bandwidth.

Jitter

Most VoIP/UC solutions employ packet buffers to compensate for some amount of jitter, so it’s often not as easy to detect problems on calls. High jitter rates (i.e. large fluctuations in latency) will often result in dropped packets and gaps and audio/video transmission. In general, you want to have internal network jitter of less than 100ms. Jitter beyond that level will result in additional transmission delay (from the buffer processing) and potentially packet loss. Jitter is also often a sign of networks not configured to prioritize VoIP/UC packets.

Packet Loss

VoIP is highly susceptible to packet loss. Even a 1% packet loss will noticeably affect most real time protocol (RTP) codecs. VoIP/UC clients will recover the best they can but the user experience will be very poor, resulting in lost productivity, aborted calls, and sometimes headsets that are smashed to bits in a fit of rage.

Packet loss is often the a symptom of latency/jitter issues, but is often the result of network misconfigurations, insufficient bandwidth, or simply network equipment that is not up to the task of handling the increased demands of real time communication and collaboration.

Getting Ready for VoIP

To get ahead of these problems, you need to implement a monitoring and remediation plan early, ideally before you deploy these IP communications solution widely. This plan should include the following:

Baseline Testing and Ongoing Monitoring – You can’t fix what you don’t see. It’s a good idea to run some diagnostics/synthetic point-to-point tests within your network to measure baseline latency, jitter, and packet loss metrics so you can see how well your network is configured today. However, it’s also important that you continue to monitor these metrics as you scale your VoIP/UC deployment so you don’t get blindsided as users come online.

Network Configuration Improvements – Your existing network equipment may support higher VoIP QOS but require configuration changes to do so. Configuring to prioritize VoIP traffic will improve both latency and jitter. Bandwidth reservation, policy-based network management, Type of Service, Class of Service, and Multi-Protocol Label Switching (MPLS) are techniques generally used for prioritizing VoIP traffic.

Network Equipment Upgrades – It may be time to replace older and less capable network equipment. High quality VoIP routers will have a significant impact on these issues and are a must if you are using VoIP as a complete replacement to your legacy phone system. In addition you should consider upgrading your internet connection. You’ll be needing more speed and bandwidth going forward.

With a solid monitor, reconfigure, and upgrade plan in place your users will have fewer call problems and you’ll have few mangled headsets to replace.

Patrick Carey is VP Product Management & Marketing at Exoprise.

Patrick leads product management and marketing for Exoprise (www.exoprise.com) a provider of Crowd Powered performance monitoring solutions for enterprise SaaS applications like Office 365, Salesforce.com, Box and others. He has over 20 years of experience delivering software solutions to both enterprises and service providers.
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