TeleDynamics Think Tank

Understanding how codecs work

One of the most abstract attributes of Voice over IP (VoIP) is the codec. We may wonder which codec to use in a given situation. Of course, there are general industry guidelines that help us make such decisions, but it can be unnerving to blindly follow recommendations without knowing why. For example, why should G.729 be preferred over the G.711 codec for WAN links? Why should I use G.722 when I don’t have an issue with bandwidth? What are the advantages of iLBC and Opus over the G series of codecs and when should I use them?

New year, new risks. Let’s talk security.

In today’s digital economy, mobility is not only an expectation, but in many cases, a requirement for workers to successfully fulfill their responsibilities. Security is a key preoccupation for network administrators as they implement voice and data technologies for remote workers.

In this article, we review various remote employee voice service scenarios, look at what their potential risks are, and examine the best voice encryption and VPN solution for each case. 

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.

Product Review

Panasonic has taken advantage of the incredible flexibility that IP telephony provides by introducing its new Mobile Softphone. It integrates with Panasonic IP telephony systems to provide versatile and adaptable unified communication services, anytime, anywhere.

IP Telephony features are not always “plug-and-play.” Rather, they must be configured to function properly. This is equally true when dealing with quality of service (QoS) on a network that transmits both data and voice (i.e., a converged network).

Companies commonly find that when they install their VoIP system on a preexisting data network, it works great at first. Days or weeks later, however, users complain of poor voice quality and intermittent disconnections. The network has not changed, so what’s going on?

In this article, we’ll see why QoS is a fundamental part of your network design for voice, and examine five configurations that should always be employed to achieve high-quality voice on a converged network.

Product Review

Thanks to VoIP technology, it’s the software, not the hardware, that takes a starring role in a business telephony system. And the sophistication of the software can turn even an entry-level endpoint into a full-service telephony device with the robust feature set that in the past was only available on more expensive phones. For micro- and small businesses that want high-end functionality on an entry-level endpoint with a small price tag, Grandstream has you covered with the GXP1610 and GXP1615 1-line IP phones.

Your customer using a legacy phone system decides to switch to voice over internet protocol (VoIP). They install an IP PBX and buy some IP phones. Great, they’re all set, right? Wrong. Without configuring their data network for quality of service (QoS), they will experience a severe deterioration in voice quality and may regret making the decision to switch. Yes, the IP PBX and the IP endpoints will already be configured for QoS. But what about other parts of the network like the pre-existing routers, switches and firewall? All it takes is one missing link for the whole system to be compromised.

QoS is a big topic. In this article, we’ll look at two main approaches to QoS: IntServ and DiffServ, their strengths and limitations, and when to use which one.

Over the past few weeks we’ve looked at different ways to structure multisite VoIP deployments. The truth is, each business is unique and sometimes a customized architecture is needed. In this article we look at a hybrid solution. A hybrid call processing deployment allows you to craft a solution that is perfectly tailored to a specific customer’s needs.

Last week, we discussed centralized call processing and its benefits and potential pitfalls. In this article, we’ll examine another type of telephony infrastructure design for multisite enterprises: distributed call processing.

Enterprises with multiple sites have a variety of options when it comes to their IP telephony infrastructure. Which setup will best meet their needs depends on the answers to questions such as:

• Where should the call processing take place?
• Where should calls to the PSTN exit the network?
• Where should incoming calls connect to the corporate network?

These decisions depend on the nature of the underlying network infrastructure that will carry the IP telephony services, as well as other considerations such as the location of the remote sites, the dial plan, and cost.

In this article, we discuss the centralized call processing model and how to avoid common complications with its deployment.