TeleDynamics Think Tank

How to deal with jitter, packet loss, latency and signaling delay

Some of the most difficult issues to troubleshoot on a network are failures resulting in a degradation of service, a slowdown in speed, or intermittent connectivity, as opposed to a complete disruption in service.

When it comes to troubleshooting issues involving VoIP, this becomes even more acute. Intermittent, low-quality, and metallic-sounding voice, as well as erratic connectivity between VoIP end devices, are all network problems falling into this difficult-to-diagnose category.

Almost all such malfunctions are due to a faulty implementation (or lack thereof) of quality of service mechanisms. In this article, we will look at some of the most common failures of this type, what they sound like, and what should be done to verify and rectify the problems.

If you’re on the voice side of the telecommunications industry, it is inevitable that you will be dealing with data networking. As Voice over IP (VoIP) technologies are overtaking more traditional telephony systems such as ISDN and analog, the voice engineer is required to know data networking fundamentals in order to implement, maintain and troubleshoot modern voice systems.

Back in 1999, when VoIP was still in its infancy, Industry Expert Bruce Robertson told students in the Sheridan College Telecommunications Management post graduate program, “If you go into the telecom industry on the data side, you will probably stay on the data side for all of your career. If you go in on the voice side of things, you will inevitably become involved with the data side, as well.”

This statement has certainly borne true. So, if you’re an expert in ISDN, SS7 or other conventional telephony technologies, it’s essential to also be familiar with data networking fundamentals to remain competitive and relevant in the telecom industry. In this article we look at a nifty tool called a packet sniffer that can be used for troubleshooting many types of VoIP problems and should be in the toolbox of any telephone installer.

Unlike conventional telephony infrastructure, which was developed and tailored to be used exclusively for voice, today’s telecom networks accommodate a multitude of different data traffic types, of which VoIP is just one of many. Because of this, it can be challenging to adapt a network to accommodate voice.

One of the most common challenges involves a technology called Network Address Translation (NAT), which for data networks has been a godsend, but if not configured carefully, could cause problems for voice applications. In particular, NAT is a common cause of one-way and no-way audio on VoIP calls.

In this article, we dive into how NAT can impair voice sessions, cite some common symptoms that indicate NAT may be at the root of your call audio problems, and address how to resolve the issues. 

How to troubleshoot problems arising from blocked ports, ACLs, firewalls, NAT, and more

For networking professionals, one of the most difficult things to deal with is troubleshooting VoIP issues, because the troubleshooting process for VoIP is not always intuitive. Answers to the questions, “What could be wrong?” and “What should I check first?” are not always readily apparent.

To aid your troubleshooting and allow you to take some meaningful actions before reaching out to your telco or vendor help desk, we’ve listed the most common VoIP problems you may face, as well as their causes and solutions. 

Are you hearing both sides?

By Kate Clavet, Marketing Specialist, Grandstream Networks

The very technology that makes VoIP so efficient can also indirectly cause one-way audio issues.

VoIP stands for Voice over Internet Protocol (VoIP), indicating that the communication travels over the internet in little packets using RTP (Real-Time Transport Protocol). Since the packets are traveling through the internet, they must pass through firewalls and network address translation (NAT). Due to the tricky nature of firewalls, a common complaint of customers and installers alike is that either the caller or callee cannot hear the other party. Although this is a common inquiry and topic for support tickets, it is also rather easy to fix.

If you deal with Voice over IP (VoIP), you must have come across this scenario at one time or another: A user complains that when they answer their phone, the caller can’t hear them, even though they can hear the calling party. Or, it may be that neither party can hear the other and there is just silence on the line.

This is the classic case of one-way or no-way audio, where a voice call is successfully completed, but either the voice packets only successfully travel in one direction, or neither end successfully receives voice packets. It may be difficult to understand why this happens, especially since the phone does ring, both physically for the called party and via the ring-back tone for the calling party. It seems counterintuitive that the transmission of voice packets could be unsuccessful if the call was successfully set up.

This is a scenario that comes up a lot on our tech support calls at TeleDynamics. Here we list four of the most common culprits of this issue and suggestions for how to tackle them.