Factors that Determine the Best Quality of Service VoIP

Voice over Internet Protocol (VoIP) has a much shorter history but VoIP benefits from the vast amount of technical and financial investment driving the rapid evolution of the internet. Assuring the quality of service VoIP (or video) connection over the internet, because each connection is “built” from a potentially enormous number of devices and connecting cables, appears to be a much more difficult goal.

While a block diagram describing the quality of service VoIP connections and flow of data traffic for even a small organization can be intimidating, in a business setting VoIP service has little difficulty matching the call performance we've come to expect from the PSTN. With a better understanding of the quality of service VoIP, you can manage any disruptions to your company’s communication. With effective bandwidth management, you are able to have fewer dropped calls and frustrating calls.

You can measure the quality of service VoIP using the Mean Opinion Score (MOS). MOS of quality of service VoIP is measured by measuring the bandwidth, latency, jitters, packet loss, and codecs with a minimum score of 4.3-5 being required to make VoIP calls.

Bandwidth

Measured in Mb/second and this shows the upload and download speed of your internet connection.

Latency

The lower the latency score, the better quality of service VoIP. Latency measures how fast you are connected to the other party in a call.

Jitter

Jitters are the time gaps between data packets being transmitted during a VoIP call. A lower jitter score means a higher quality of service VoIP calls. Fiber connections offer the best jitter rate.

Packet Loss

The virtual cases carrying data of your call are packets. You can experience data pack loss with lower quality of service VoIP as a result of the jitter buffer being overwhelmed. The best quality of service VoIP providers have virtually no packet losses.

Voice Compression

VoIP calls are compressed to improve the effectiveness of transmission. The lower the size of the data packets, the better quality of service VoIP transmission. VoIP services use varied compression ratios where the lower the ratio, the better quality of service VoIP you can experience.

Codecs

VoIP coders and decoders convert the analog signal into data packets and back again to voice to place calls via the internet. Using different sample rates, codecs are able to determine the compression and transmission rate required for quality of service VoIP. A lower sampling rate means a lower quality of service VoIP. However, a lower sampling rate can reduce contention and prevent degradation of the call as well.

If you are the decision maker for your organization's planned VoIP system, there are a couple of basics to keep in mind to assure your users never come looking for you with malice in their eyes regarding the quality of service VoIP.

Bandwidth

All (or essentially all) of the vocal sounds made by one party need to arrive “instantly”, in the correct order, and appropriately spaced, at the ear of the other party. The “pipe” between them cannot restrict the flow of information.

Having a large enough “pipe” to assure every bit of quality of service VoIP data poured across it meets zero resistance due to other traffic (congestion) is generally a good idea. This philosophy has been termed “over-provisioning” and approaches VoIP bandwidth specifications from the perspective of “If some is good, more's better, and too much is just enough.”

For a simple network, over-provisioning quality of service VoIP may be effective and relatively inexpensive, but when the block diagram of devices and connections begins to look like the map of the Tokyo Metro, more serious quality of service VoIP consideration is called for.

QoS Management

As with all things internet, the number of Quality of Service VoIP (QoS) products is impressive. From freeware to complex proprietary hardware/software devices, the choices all provide some combination of the following services:

They “watch” the internet traffic flowing through their device for volume (bandwidth) and type. They record and report on parameters, including the ones most important to the quality of service VoIP; lost data, latency, and jitter. Traffic should be “watched” across many devices, depending on the complexity of the network, so Quality of Service VoIP software is not located on only one device (except in very simple systems).

At the very least, your ISP has a QoS package they use to assure they are delivering the level of quality of service VoIP promised your organization (and they can also use it to suggest additional services when you begin to come up against the current capacity limitations).

The quality of service VoIP information gathered is available to your IT folks to configure the appropriate devices (such as VoIP routers, gateways, ATAs, and bridges) to prioritize types of traffic and allocate bandwidth. Typically voice and video traffic are given the highest priority. Voice and

video are also frequently allocated a minimum level of bandwidth to further assure the quality of service VoIP. Optimizing the flow of network traffic for VoIP can deliver the best quality of service VoIP to your users and deliver the best return on your bandwidth investment.

As an organization grows, the quality of service VoIP system will alert the IT staff to increase bandwidth where it is needed. With a well-implemented quality of service VoIP system and a large enough “pipe”, the users will enjoy the same instant, clear connections worldwide with the VoIP system, as they have with the PSTN.

To help you select the best quality of service VoIP provider on the market, refer to our Phone Systems Buyer’s Guide to learn what you need to be on the lookout for. You can compare the different quality of service VoIP providers to determine the best fit for your choosing a VoIP to grow your small business.