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With business internet usage continuing to increase, contention for bandwidth between your data and VoIP activity can create problems.  Yes, the convergence buzzword has been replaced with contention!

Why is this happening?  Business needs and applications are evolving fast.  Even with bandwidth expansion availability from internet carriers, keeping up with the increasing demand is a challenge.  More businesses are using cloud computing applications, using the internet to move data to a host server rather than to servers on their local area networks.  On-line data backup is another such application.  General use of the internet continues to accelerate, both for personal and business purposes.  With the sky-rocket adoption rate of smart phones, the lines between business and personal have blurred, and office internet usage is following suit.  Social media is not only a personal outlet, but a business tool and resource.  Buying practices have shifted toward internet research and ecommerce, and marketing activity has followed.

I’m looking at my personal browser.  On a daily basis, my browser auto-opens tabs for three social media sites, five cloud applications, news, and Google.  Each application is actively worked throughout the day.  This is becoming the norm, and it requires a lot of bandwidth to perform efficiently.

What is the impact on VoIP applications?  Most businesses today use a VoIP phone service application for their phone service needs.  When a company uses a hosted VoIP service, voice packets travel over the same internet connection as the data packets.  Good VoIP phone service providers will provide instructions to program your router to prioritize voice packets over data to ensure that your calls remain clear and free from interruption.  However, if the internet connection is highly saturated with activity, even the best prioritization routines will be compromised.  Phone conversations can experience poor voice quality and phones can lose their registration and go off-line.

What can be done?  Where available, you should look at your internet provider’s capability to add bandwidth to the circuit or change to a provider who has the ability to grow with your needs.  Depending upon your internet usage and needs, you still may have occasional contention between your voice and data traffic.   However, an alternative may be to separate your voice and data network.

With a separate voice and data network, your calls and phones will no longer be impacted by your data usage.  Many VoIP phones have two ports on the back which allow for a computer to share an Ethernet cable to the network.  The phones and computers can still share an Ethernet connection to the servers, but they will simply be directed out disparate internet connections.  This can be accomplished by setting up a vLan on your switch or by simply assigning the phones separate IP addresses and route them out the new internet connection.  Your network manager or your hosted VoIP phone company should be able to assist you with this programming. 

Finally, voice calls do not require much bandwidth (80 kb/s non-compressed).   Dependent upon your concurrent call maximum, your alternative connection may not be required to be very large.

Capture some statistics on your network usage.  Compare this with the anecdotal complaints from your employees about internet speeds and VoIP call quality and then investigate your options for more bandwidth and/or disparate connections for voice and data.

About the author:  Mark Greim is the Vice President of Sales and Marketing at sipVine, a VoIP Phone Service Provider based in Kansas City, MO. Mark has extensive experience working for start-up or entrepreneurial organizations and a passion for affordable, reliable, and purposeful technology solutions in those environments. 

When considering a hosted VoIP phone service for your business, are you concerned about the voice quality of your phone conversations with your customers?  Do you believe that with increasing availability of expanded bandwidth that call quality concerns are diminished? 

Most people that I talk to agree that voice quality is important not only as a reflection of their business, but to ensure that operations run smoothly. And, while increasing internet bandwidth may be an option, it certainly does not diminish call quality concerns with VoIP. 

Hosted VoIP continues to be the product of choice for businesses looking to affordably increase their communication functionality and replace their old technology.  However, it is extremely important to select a provider who has a focus on call quality and the network infrastructure and appliances to deliver a crisp and clean voice conversation using VoIP.

sipVine, a Kansas City-based Hosted VoIP company, is an example of a provider who has such a focus on call quality for their customers.  I was asked recently if call quality was still a point of consideration before businesses switch to VoIP, and I needed to pause before answering.  Certainly, when sipVine began in 2005, not only was there a general uncertainty over the new technology, but also a vast opinion that the call quality could not be as good as an analog line.  Such opinions were formed from poor experiences with national consumer VoIP providers, and from business providers running VoIP over lower bandwidth connections at the customer locations.   However, such objections do seem to be in the past, as companies are no longer fearful of VoIP and it is the preferred technology as businesses replace current systems.

That said, call quality is still very much something that needs to be managed, and managed correctly.  The same day after being asked the question on the relevance of call quality in the process of choosing VoIP, we closed two sales from companies using national VoIP providers and were experiencing horrific call quality.  We encounter this situation routinely.  Most national providers suggest that the business expands their bandwidth or purchase an additional internet circuit to pass the phone traffic.  This action is an unnecessary use of money and network resources, and still doesn’t guarantee improved call quality.

At sipVine, we take a two-pronged approach to call quality management:

Quality of Service (QoS) Software and Network Design - Our technology team believes strongly that the best way to manage VoIP call quality is to have an on-network "traffic cop.” Resident on the router or through an edge appliance on the network, we manage call quality by prioritizing voice traffic over the data traffic.  To the naked eye, you do not see any network slow down by such prioritization routines.  However, to the ear, you certainly can notice when your data traffic interrupts the voice packet delivery.  This software allows businesses to utilize the same internet connection for voice and data and, many times, this action will negate the need to expand bandwidth just to run VoIP.  Most importantly, simply expanding bandwidth does not remove the potential for data packets to interrupt voice packets! 

sipVine also has a unique network design that creates an optimum mix of internet and public telephone network utilization.  Although it is more expensive to operate, we believe it is a small price to pay to produce a quality call environment.

Call Quality Troubleshooting - Even with QoS software and an optimal network topology, there are cases where the customer experiences degradation of voice quality, as manifested by one-way audio or choppy communication.  sipVine technicians routinely help identify the root causes, whether it be a customer network related issue or an internet circuit related issue.  Only on rare occasions do we see where normal data and voice utilization are saturating the bandwidth provided!  In every case where we've rescued a customer from a national VoIP provider, the initial recommendation was always to expand bandwidth. 

Examples of network challenges we've encountered were certain workstations being hit with a virus and saturating either the upload or download side of the available bandwidth.  This would lead to one-way audio.  Our technicians were able to, remotely, identify the workstation(s) driving the issue.  Other examples are when customers have established new network backup or storage routines during business hours.  After pointing out the impact it was having to overall network performance and voice quality, these customers usually changed the timing of their routines to after business hours. 

Finally, many times there's an issue with the internet circuit itself.  Perhaps there's a higher amount of latency than acceptable, or that the ping times that measure the round trip latency are not consistent.  For example, if they typically run at 15-25 milliseconds and the current experience shows 25ms, 25ms, 100ms, 90ms, 20ms, etc., there is going to be a problem in voice quality.  sipVine technicians have helped our customers effectively communicate with their internet provider by providing such reports on circuit performance so that it may be quickly addressed.

So, back to the question that was posed to me.  The answer is unequivocally YES, call quality does matter when considering VoIP!  If you’re considering outsourcing your business phone service, our advice is to work with a provider who makes quality a priority within their own network, and will actually help diagnose any problems with networks or circuits that may be driving a problem.

About the author:  Mark Greim is the Vice President of Sales and Marketing at sipVine, a provider of a variety of VoIP phone solutions and services. Mark has extensive experience working for start-up or entrepreneurial organizations and has a passion for affordable, reliable, and purposeful technology solutions in those environments. 

There is no nice way to put this, but certain national Hosted VoIP providers are giving the industry a bad name.   There are countless examples of businesses who are experiencing poor voice quality with these providers and receive the standard recommendation to increase their internet bandwidth.  This solution costs more money and isn’t always the best advice.

The fact of the matter is that VoIP calls do not take much bandwidth, less than 80kb per call.  However, because voice and data packets are sharing the same connection, there is always an opportunity for voice packets to be interrupted by data packets.  Expanding bandwidth doesn’t always take care of the problem.

One of sipVine’s biggest differentiators in the market is that we believe that call quality is best controlled on the customer’s network.  We provide a router or a transparent bridge that has Quality of Service programming built in, ensuring your voice calls have priority.  This allows your business to right size the amount of bandwidth to match the needs of the business and not overbuild just for VoIP. 

Certainly, there are occasions where the amount of data activity is over-saturating the internet bandwidth.  In those situations, QoS will not be able to work as well as it should and the bandwidth will need to be addressed.  However, we are bringing businesses on to our platform on daily bases that have experienced this poor advice from a national hosted VoIP provider.  In addition to providing a localized QoS device, we have helped our customers troubleshoot issues with their internet circuit to find the root cause of the call quality concerns.  This has saved them from blindly adding bandwidth, thus saving them dollars on a monthly basis.

To learn more, check out our whitepaper, Voice Quality: The Ultimate Differentiator in VoIP Providers.

About the author:  Mark Greim is the Vice President of Sales and Marketing at sipVine, a provider of a variety of VoIP phone solutions and services. Mark has extensive experience working for start-up or entrepreneurial organizations and has a passion for affordable, reliable, and purposeful technology solutions in those environments. 

This is one of the most common questions we receive from our prospective customers.  VoIP calls really do not require much bandwidth at all, and there are adjustments that can be made to condense the requirement even further.

The first mental hurdle to overcome is that your concurrent voice calls are no longer restricted to the number of analog “lines” you have provisioned for your company.  With VoIP, your only restriction is the amount of bandwidth you have.  Even if you are flooded with concurrent calls, many VoIP systems will capture the call and deliver it to your company voicemail, thus your customers will never hear a busy signal.  However, the above scenario makes the question of bandwidth requirements very appropriate.

A single VoIP call requires 80k of bandwidth in a non-compressed environment.  Thus, 10 concurrent calls active on your system would require 800K of bandwidth.  However, it is common practice to utilize voice compression to reduce this requirement. A compression codec will lower the requirement to as little as 25k of bandwidth per call.   Utilizing compression can help conserve bandwidth without a material sacrifice of voice quality, especially if your VoIP provider provides a Quality of Service software with their service.

There are some things you can do to test your company’s readiness for VoIP phone service, however a good VoIP service provider should consult with you prior to your installation on these very topics:

Run a speed test – run a free speed test of your bandwidth.  Record the download and upload speeds, and try again later in the day and repeat every few days.  You’re looking for your download speeds to be robust enough to handle your voice and data volume and for the speed results consistent as well. 

Run an extended ping test – you should ping the potential service provider to check the latency between you and their server.  You’re looking for the result to be under 110 milliseconds at a minimum.   Ideally, a good circuit should produce latency in the 20 milliseconds or less range.  Furthermore, it should be relatively stable.  Results bouncing around (like 20, 20, 150, 90, 20, 20, 150, 20, etc) will produce packet drops and jittery phone calls.

Check for packet loss – a stable internet connection should experience no packet loss. VoIP service is dependent upon a no to very low packet loss.  If packet loss is approaching 2%, contact your internet provider to service the line.

Determine your concurrent call volume – this may be via observation or through an understanding at which point you receive busy signals on your analog system.  If you have five incoming lines, and you or your customers have experienced busy signals, then your concurrent call maximum is greater than five.

Do the math – Maximum concurrent call volume multiplied by 80k must be less than the lower of the upload or download speeds recorded on your speed test.  We usually recommend that the maximum voice call packet transfer doesn’t absorb more than 40-50% of your bandwidth availability.  However, this is entirely dependent upon your data requirements.  Additionally, compressed codec could be deployed before considering more bandwidth.

The amount of bandwidth you have isn't the only variable in delivering quality phone conversations with VoIP.  Many providers "solve" voice quality issues by oversizing bandwidth.  However, there are other, more reliable solutions.   Give sipVine a call today for a consultation.  We're happy to help.