An SD-WAN can be up to 2.5 times less expensive than traditional WAN architecture.

One of the most challenging situations for networking professionals over the years has been how to interconnect remote sites and branch offices so users at all locations can enjoy the same quality of network and telephony services in a secure and timely manner. One way to do this is to duplicate all services at each location. However, this can become administratively unwieldly, not to mention prohibitively expensive.

The solution is the use of wide area networks (WANs), where centralized services like voice and data reside at headquarters and can be provided to the remote users as if they were all physically located inside the headquarters’ LAN (local area network). In this article, we look at the different ways to deploy WAN, as well as the problems solved and business benefits to be gained by using software-defined WANs (SD-WANs).

Last week, we touched on some of the technological advances that have allowed VoIP telephony to surpass the PSTN in both quality and reliability.

Yet, even though VoIP has been around for two decades, there are still some misconceptions about what it really is. A lot of people think of VoIP as a technology that allows placing phone calls over the internet. Although this is true, such a narrow definition encompasses only a fraction of the functionality of VoIP.

The unfortunate consequence of this misperception is that VoIP is often associated with low quality and best-effort services that offer low cost or free voice calls over the internet. Those familiar with these services have experienced the frequent disconnects, jitter and stuttering on the line characteristic of this type of service. As a result, VoIP has erroneously been associated with an unreliable user experience.

The truth is, VoIP is much more (and better) than just voice over the internet.

The most recent FCC Voice Telephone Services report, published in November 2016, states that two thirds of businesses in the United States are still using switched access telephone lines (PSTN or POTS). The reasons they haven’t yet migrated to VoIP (voice over internet protocol) are varied. One of the most prevalent myths about VoIP is that the quality is not as good or reliable as that of traditional telephone lines. The truth, however, is that VoIP technology has evolved to a point where the audio quality is generally as good if not better than analog lines.

Here we list some of the technological developments that have allowed VoIP to surpass even PSTN voice quality, some of which may surprise you!

Voice over IP networks can be challenging to implement efficiently and securely. This is in part because voice packets require specialized management and must be treated differently than normal data packets. Quality of Service (QoS), security and flow control are just some areas in which the required handling of voice traffic differs from conventional data traffic.

The good thing is that with the proper expertise and care, many of these issues can be successfully dealt with within the corporate network. Such networks are under the complete control of the network administration team and thus can be fully customized for the needs of the voice being transmitted.

However, what happens when voice packets are routed beyond the corporate network, either for voice calls to the PSTN or mobile network, or for employees who use internal voice network services remotely? Here we look at various network edge mechanisms that can be used to monitor and manage data traveling in and out of the LAN.

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.

By John Macario, SVP of Product Management and Marketing at Edgewater Networks

The beginning of a new year is a time when many business managers reflect on learnings from the previous year and begin working on their goals for the current year. Business continuity and data security are, of course, top priorities and securing the network is a vital part of this. 

For network professionals, there is no word more fearsome than “breach.” A security breakdown can mean lost data, stolen secrets, expensive lawsuits, and more. According to IBM, the average cost of a data breach is now US$4 million. This is why nearly every organization that knows what's good for it now invests in antivirus, data loss prevention (DLP), and of course, firewalls.

These days, however, we need to protect more than just data. Enterprise communications have grown into a multi-billion dollar market, particularly due to the widespread adoption of VoIP and unified communications (UC) services. This presents a new challenge that firms will have to face in order to gain a winning edge: voice security. IP phone services present a whole new attack area to potential hackers, a frontier that firewalls simply cannot protect by themselves. The essence of this challenge lies in the fact that we're talking about two entirely different types of information transmission that, in turn, require different security solutions.

By Stefan Eriksson, Marketing and Communications Manager of Konftel

Remote meetings with poor quality audio are often exhausting. Misunderstandings become more likely because it’s difficult to hear nuances and other subtleties in conversation. So you want to aim for the best sound quality possible during teleconferences. The following is a quick explanation of different technical requirements for audio quality.

• A purely mobile solution gives you great flexibility and mobility, but sometimes at the expense of sound quality. Many mobile operators now offer HD Voice in their networks, which delivers HD audio if the phone supports the technology.
• Traditional analog telephony delivers an acceptable sound quality, but with limitations in frequency range. Sometimes known as telephone quality or narrowband.
• VoIP, i.e., digital telephony via the data network (voice over IP), allows for extended frequency range, but with some compression. IP makes it possible to achieve superior audio quality, also called HD audio or wideband.
• Remember that all local networks and accessories, e.g. Wi-Fi, DECT (wireless telephony) or Bluetooth^®, affect the transmission capacity and can have an adverse impact on the sound quality.
• All of Konftel’s phones and conference units offer HD audio in contexts where PBXs and networks support it.

About 90% of businesses in the United States have fewer than 20 employees, according to the Small Business & Entrepreneurship Council. In the past, companies of this size often couldn’t benefit from PBX functionality like IVR (interactive voice response), hunt groups, hold music, and so forth, because of the high cost of installing and maintaining a PBX.

But today, things are different. Thanks to hosted VoIP service, micro-enterprises can have a phone system with even more functionality than a legacy PBX, even if they only have one user! All they need is a broadband Internet connection. With VoIP, micro-businesses can be more efficient and enjoy the same functionality that big corporations have relied on for decades.

Here we look at how a micro-enterprise can get started with VoIP and some of the benefits it can reap as a result.

Ensuring business continuity in case of unexpected service loss is an essential component of any IP telephone system. In a previous blog post we mentioned various failover mechanisms that can be deployed to address different scenarios. Here, we’ll take a closer look at two in particular: backup SIP trunk or ITSP, and failover servers. You may be surprised to discover how easy and affordable they are to set up – especially when compared with the costly alternative of service interruption.

We consider session border controllers (SBCs) to be a best practice in most enterprise contexts because of the improved security, control and overall VoIP system functionality they lend.

So what exactly are SBCs and why do you need one?