Software Defined Networking (SDN) is changing the way networks are managed and designed. In this article, we take a look at how an SDN approach can be of great benefit to VoIP and video telephony deployments in particular.
Many administrators resist implementing VoIP over Wi-Fi, often citing security and quality concerns. The truth is that if implemented correctly, voice can be safely and reliably deployed on a wireless network. This article examines some best practices for optimizing voice over Wi-Fi (VoWi-Fi).
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.
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.
Review of the EdgeMarc 2900a and 2900e Enterprise Session Border Controllers
Edgewater Networks has added to its line of enterprise session border controllers (eSBCs) with its EdgeMarc 2900a and 2900e models. These are designed to enable enterprises and service providers to future-proof their SIP trunking and Unified Communications (UC) deployments. With built-in mechanisms for both QoS and security, they provide intelligent mechanisms allowing for both hosted PBX services as well as SIP trunking applications.
They have several features that differ from other EdgeMarc SBCs.
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.