How Much Bandwidth Do You Really Need?

February 17, 2013 — netequalizer

By Art Reisman – CTO – www.netequalizer.com

Art Reisman CTO www.netequalizer.com

When it comes to how much money to spend on the Internet, there seems to be this underlying feeling of guilt with everybody I talk to. From ISPs, to libraries or multinational corporations, they all have a feeling of bandwidth inadequacy. It is very similar to the guilt I used to feel back in College when I would skip my studies for some social activity (drinking). Only now it applies to bandwidth contention ratios. Everybody wants to know how they compare with the industry average in their sector. Are they spending on bandwidth appropriately, and if not, are they hurting their institution, will they become second-rate?

To ease the pain, I was hoping to put a together a nice chart on industry standard recommendations, validating that your bandwidth consumption was normal, and I just can’t bring myself to do it quite yet. There is this elephant in the room that we must contend with. So before I make up a nice chart on recommendations, a more relevant question is… how bad do you want your video service to be?

Your choices are:

  1. bad
  2. crappy
  3. downright awful

Although my answer may seem a bit sarcastic, there is a truth behind these choices. I sense that much of the guilt of our customers trying to provision bandwidth is based on the belief that somebody out there has enough bandwidth to reach some form of video Shangri-La; like playground children bragging about their father’s professions, claims of video ecstasy are somewhat exaggerated.

With the advent of video, it is unlikely any amount of bandwidth will ever outrun the demand; yes, there are some tricks with caching and cable on demand services, but that is a whole different article. The common trap with bandwidth upgrades is that there is a false sense of accomplishment experienced before actual video use picks up. If you go from a network where nobody is running video (because it just doesn’t work at all), and then you increase your bandwidth by a factor of 10, you will get a temporary reprieve where video seems reliable, but this will tempt your users to adopt it as part of their daily routine. In reality you are most likely not even close to meeting the potential end-game demand, and 3 months later you are likely facing another bandwidth upgrade with unhappy users.

To understand the video black hole, it helps to compare the potential demand curve pre and post video.

A  quality VOIP call, which used to be the measuring stick for decent Internet service runs about 54kbs. A quality  HD video stream can easily consume about 40 times that amount. 

Yes, there are vendors that claim video can be delivered at 250kbs or less, but they are assuming tiny little stop action screens.

Couple this tremendous increase in video stream size with a higher percentage of users that will ultimately want video, and you would need an upgrade of perhaps 60 times your pre-video bandwidth levels to meet the final demand. Some of our customers, with big budgets or government subsidized backbones, are getting close but, most go on a honeymoon with an upgrade of 10 times their bandwidth, only to end up asking the question, how much bandwidth do I really need?

So what is an acceptable contention ratio?

  • Typically in an urban area right now we are seeing anywhere from 200 to 400 users sharing 100 megabits.
  • In a rural area double that rati0 – 400 to 800 sharing 100 megabits.
  • In the smaller cities of Europe ratios drop to 100 people or less sharing 100 megabits.
  • And in remote areas served by satellite we see 40 to 50 sharing 2 megabits or less.
Posted in Bandwidth Management, Educational Articles, Research. Tags: , , , , , , , , , , , . 4 Comments »

Equalizing is the Silver Bullet for Quality of Service

November 30, 2012 — netequalizer

Silver Bullet (n.) – A simple and seemingly magical solution to a complex problem.

The amount of solutions available that have been developed to improve Quality of Service (QoS) for data traveling across a network (video, VoIP, etc.) are endless. Often, these tools appear to be simple, but seem to fall short in implementation:

Compression: Compressing files in transit helps reduce congestion by decreasing the amount of bandwidth a transfer requires. This appears to be a viable solution, but in practice, most of the large streams that tend to clog networks (high resolution media files, etc.) are already compressed. Thus, most networks won’t see much improvement in QoS when this method is used.

Layer 7 Inspection: Providing QoS to specific applications also sounds like a reasonable approach to the problem. However, most applications are increasingly utilizing encryption for transferring data, and thus determining the purpose of a network packet is a much harder problem. It also requires constant tweaking and updates to ensure the proper applications are given priority.

Type of Service: Each network packet has a flag as part of its payload that denotes its “type of service.” This flag was intended to help give QoS to packets based on their importance and purpose. This method, however, requires lots of custom router configurations and is not very reliable as far as who is able to set the flag, when, and why.

These solutions are analogous to the diet pill and weight loss products that inundate our lives on a daily basis. They are offering complex solutions to a simple problem:

Overweight? Buy this machine, watch these DVDs, take this pill.

When the real solution is:

Overweight? Eat better.

Simple solutions are what good engineering is all about, and it drives the entire philosophy behind Equalizing – the bandwidth control method implemented in our NetEqualizer. The truth is, you can accomplish 99% of your QoS needs on a fixed link SIMPLY by cranking down on the large streams of traffic. While the above approaches try to do this in various ways, nothing is easier and more hands-off than looking at the behavior of a connection relative to the available bandwidth, and subsequently throttling it as needed. No deep packet inspection, compression, or packet analysis required. No need to concern yourself with new Internet usage trends or the latest media file types. Just fair bandwidth, regardless of trunk size, for all of your users, at all times of day. When bandwidth is controlled, connection quality is allowed to be as good as possible for everyone!

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Eleven Tips to Improve VoIP & Video on the Internet Using NetEqualizer and DiffServ/TOS Bits

October 30, 2011 — netequalizer

When talking to potential customers that do not have a NetEqualizer in place (yet), we often hear concerns from companies with recently installed VoIP systems that they are having trouble hearing incoming calls on their phones.  Typically, the root cause for this poor connection is that users are downloading files simultaneously with their VoIP calls.

Router technologies use a technology called DiffServ to enforce priority. Diffserv is reliable at preventing your outgoing Internet data users from interfering with your VoIP calls; however, most router technologies cannot prevent incoming Internet data traffic from overwhelming your incoming VoIP stream. This makes for the interesting dilemma on a call where they can hear you but you can’t hear them.

Fortunately, our bandwidth shaping technology, unlike a basic router, already uses techniques that allow an enterprise to prevent incoming data from overwhelming their VoIP/Skype calls.  We call this technology “Equalizing,” and we have recently enhanced our Equalizing algorithms (version 5.5 and above) such that specific priority for TOS/DiffServ bits will also be recognized.  DiffServ stands for “Differentiated Services Code Point (DSCP)” field and is analogous to the Type of Service (TOS) field.

The following FAQ addresses eleven common questions about our new TOS/DiffServ-aware technology:

1) Who can take advantage of this feature?
Anybody who needs to give priority to an incoming video or voice stream but does not know the source IP of the sender.

2) How do you control whether traffic coming into your network has a TOS/DiffServ bit enabled or not?
This is great mystery. Very little is written about this and how public Internet applications use the TOS bit. From experiments to-date, it seems that YouTube and VoIP providers are setting TOS bit(s) on their data streams.  This is the main reason why the initial NetEqualizer release 5.5 will be in beta test. It is an experimental release so our customers can turn on TOS/DiffServ priority and gather information on performance gains.

3) Who can set a TOS bit?
Almost any application that wants to can send out a stream with a TOS bit set; however, the typical home user does not have access to the TOS bit.

4) What are some of the Caveats with using the DiffServ/TOS Priority Feature?
In the initial beta release, we did not differentiate between types of TOS bits. There are several bits that can be set in this field by the sender that imply different types of quality. We decided to just treat this entire field as ON or OFF in our first release. Most networks that attempt multiple levels of priority are just not practical, as equipment lacks resolution in their processing to enforce different levels of priority. We decided to keep it simple; a stream either has priority or it doesn’t. Multiple levels of priority is more of an academic endeavor for wishful specifications.

5) How do you set the DiffServ/TOS Priority Feature from the NetEqualizer GUI?
Under “Modify Parameters” in the NetEqualizer set up screen:

TOS_ENABLED (on/off)

6) How do you know when a stream on your network has the DiffServ/TOS bit enabled?
From the “Active Connections” reporting screen on the NetEqualizer GUI, you will see a value of either on or off in the last column of the connection row.  “Off” indicates a TOS value of 0; “on” represents a TOS value greater than 0.

7) How does DiffServ/TOS bit priority compare with normal default equalizing?

To recap: A NetEqualizer bandwidth shaper naturally gives priority to VoIP and small web pages.

Now with the ability to provide priority specifically to streams with the TOS bit set, you can more tightly tune the NetEqualizer for VoIP priority, while at the same time provide priority for video.  The big variable will be just how much the TOS bit is used in public applications. On many of our field systems, we do have room to allow a little extra priority for the occasional video or Skype with video component. With the ability to honor TOS priority, your Internet link can grant priority to video without having to know the IP address of the sender or receiver.

8) What if an ISP allowed priority for a TOS bit and their users get wind of it?  Can they figure out a way to jump in front of the line giving ALL of their traffic priority?
We do not think this is likely at this time; the user would have to be aware of the practice of giving priority to TOS in a bandwidth controller to start, and they would then need a fairly sophisticated setup to change all of their applications to set this bit. A more realistic scenario is that video applications will by default already set this service.

9) With the lack of control over who can set a TOS bit, doesn’t this make this feature a little risky to turn on?
My analogy would be that we have a drug that promises to cure cancer and there might be some side effects (none of them will kill you, we promise), so give it a try and tell us what you find.

Note: An administrator has the ability to turn DiffServ/TOS priority on and off quickly, and take a look at the streams on the network. From our early tests over the Internet, we did see some public streams with this bit set, but it was only a small minority of them. We think the potential benefits far outweigh the risk.
Also, we will be working closely with all customers that participate in the Beta.  When Beta customers choose to turn  on DiffServ/TOS priority, we will be available to support them, and are happy to login and do some quick heuristics to assess results.  Our next release beyond the beta will make some sweeping optimizations.

10) Lets suppose all video from YouTube has the TOS bit set, would it be counter-productive to turn it on?
The worst case scenario here is that it would render your bandwidth shaping ineffective, which is no worse than running your network without your bandwidth shaper.  The best case scenario is that you have a mix of large downloads, BitTorrents, etc. that do not have the TOS bit set,  and so turning this feature on will make your video and VoIP better.

11) Many of the points discussed are specific to priority for video.  What about priority for VoIP – does it help with that?
Yes, it can, but for the most part normal equalizing already gives priority to VoIP.  In our next release, we expect to know if the VoIP providers and video providers are following guidelines for using different TOS bits. We could then give priority to VoIP all of the time, and especially on very tight networks, we could lower the HOGMIN threshold to further differentiate VoIP traffic. This point is rather technical, and if you have read this far it might be a good idea to pick up the phone and talk over these concepts with one of our network engineers.

Related Article
Other Solutions

Posted in Bandwidth Management. Tags: , , . 1 Comment »

NetEqualizer Field Guide to Network Capacity Planning

June 10, 2010 — netequalizer

I recently reviewed an article that covered bandwidth allocations for various Internet applications. Although the information was accurate, it was very high level and did not cover the many variances that affect bandwidth consumption. Below, I’ll break many of these variances down, discussing not only how much bandwidth different applications consume, but the ranges of bandwidth consumption, including ping times and gaming, as well as how our own network optimization technology measures bandwidth consumption.

E-mail

Some bandwidth planning guides make simple assumptions and provide a single number for E-mail capacity planning, oftentimes overstating the average consumption. However, this usually doesn’t provide an accurate assessment. Let’s consider a couple of different types of E-mail.

E-mail — Text

Most E-mail text messages are at most a paragraph or two of text. On the scale of bandwidth consumption, this is negligible.

However, it is important to note that when we talk about the bandwidth consumption of different kinds of applications, there is an element of time to consider — How long will this application be running for? So, for example, you might send two kilobytes of E-mail over a link and it may roll out at the rate of one megabit. A 300-word, text-only E-mail can and will consume one megabit of bandwidth. The catch is that it generally lasts just a fraction of second at this rate. So, how would you capacity plan for heavy sustained E-mail usage on your network?

When computing bandwidth rates for classification with a commercial bandwidth controller such as a NetEqualizer, the industry practice is to average the bandwidth consumption for several seconds, and then calculate the rate in units of kilobytes per second (Kbs).

For example, when a two kilobyte file (a very small E-mail, for example) is sent over a link for a fraction of a second, you could say that this E-mail consumed two megabits of bandwidth. For the capacity planner, this would be a little misleading since the duration of the transaction was so short. If you take this transaction average over a couple of seconds, the transfer rate would be just one kbs, which for practical purposes, is equivalent to zero.

E-mail with Picture Attachments

A normal text E-mail of a few thousand bytes can quickly become 10 megabits of data with a few picture attachments. Although it may not look all the big on your screen, this type of E-mail can suck up some serious bandwidth when being transmitted. In fact, left unmolested, this type of transfer will take as much bandwidth as is available in transit. On a T1 circuit, a 10-megabit E-mail attachment may bring the line to a standstill for as long as six seconds or more. If you were talking on a Skype call while somebody at the same time shoots a picture E-mail to a friend, your Skype call is most likely going to break up for five seconds or so. It is for this reason that many network operators on shared networks deploy some form of bandwidth contorl or QoS as most would agree an E-mail attachment should not take priority over a live phone call.

E-mail with PDf Attachment

As a rule, PDF files are not as large as picture attachments when it comes to E-mail traffic. An average PDF file runs in the range of 200 thousand bytes whereas today’s higher resolution digital cameras create pictures of a few million bytes, or roughly 10 times larger. On a T1 circuit, the average bandwidth of the PDF file over a few seconds will be around 100kbs, which leaves plenty of room for other activities. The exception would be the 20-page manual which would be crashing your entire T1 for a few seconds just as the large picture attachments referred to above would do.

Gaming/World of Warcraft

There are quite a few blogs that talk about how well World of Warcraft runs on DSL, cable, etc., but most are missing the point about this game and games in general and their actual bandwidth requirements. Most gamers know that ping times are important, but what exactly is the correlation between network speed and ping time?

The problem with just measuring speed is that most speed tests start a stream of packets from a server of some kind to your home computer, perhaps a 20-megabit test file. The test starts (and a timer is started) and the file is sent. When the last byte arrives, a timer is stopped. The amount of data sent over the elapsed seconds yields the speed of the link. So far so good, but a fast speed in this type of test does not mean you have a fast ping time. Here is why.

Most people know that if you are talking to an astronaut on the moon there is a delay of several seconds with each transmission. So, even though the speed of the link is the speed of light for practical purposes, the data arrives several seconds later. Well, the same is true for the Internet. The data may be arriving at a rate of 10 megabits, but the time it takes in transit could be as high as 1 second. Hence, your ping time (your mouse click to fire your gun) does not show up at the controlling server until a full second has elapsed. In a quick draw gun battle, this could be fatal.

So, what affects ping times?

The most common cause would be a saturated network. This is when your network transmission rates of all data on your Internet link exceed the links rated capacity. Some links like a T1 just start dropping packets when full as there is no orderly line to send out waiting packets. In many cases, data that arrive to go out of your router when the link is filled just get tossed. This would be like killing off excess people waiting at a ticket window or something. Not very pleasant.

If your router is smart, it will try to buffer the excess packets and they will arrive late. Also, if the only thing running on your network is World of Warcraft, you can actually get by with 120kbs in many cases since the amount of data actually sent of over the network is not that large. Again, the ping time is more important and a 120kbs link unencumbered should have ping times faster than a human reflex.

There may also be some inherent delay in your Internet link beyond your control. For example, all satellite links, no matter how fast the data speed, have a minimum delay of around 300 milliseconds. Most urban operators do not need to use satellite links, but they all have some delay. Network delay will vary depending on the equipment your provider has in their network, and also how and where they connect up to other providers as well as the amount of hops your data will take. To test your current ping time, you can run a ping command from a standard Windows machine

Citrix

Applications vary widely in the amount of bandwidth consumed. Most mission critical applications using Citrix are fairly lightweight.

YouTube Video — Standard Video

A sustained YouTube video will consume about 500kbs on average over the video’s 10-minute duration. Most video players try to store the video up locally as fast as they can take it. This is important to know because if you are sizing a T1 to be shared by voice phones, theoretically,  if a user was watching a YouTube video, you would have 1 -megabit left over for the voice traffic. Right? Well, in reality, your video player will most likely take the full T1, or close to it, if it can while buffering YouTube.

YouTube — HD Video

On average, YouTube HD consumes close to 1 megabit.

See these other Youtube articles for more specifics about YouTube consumption

Netflix – Movies On Demand

Netflix is moving aggressively to a model where customers download movies over the Internet, versus having a DVD sent to them in the mail.  In a recent study, it was shown that 20% of bandwidth usage during peak in the U.S. is due to Netflix downloads. An average a two hour movie takes about 1.8 gigabits, if you want high-definition movies then its about 3 gigabits for two hours.   Other estimates are as high as 3-5 gigabits per movie.

On a T1 circuit, the average bandwidth of a high-definition Netflix movie (conversatively 3 gigabits/2 hours) over one second will be around 400kbs, which consumes more than 25% of the total circuit.

Skype/VoIP Calls

The amount of bandwidth you need to plan for a VoIP network is a hot topic. The bottom line is that VoIP calls range from 8kbs to 64kbs. Normally, the higher the quality the transmission, the higher the bit rate. For example, at 64kbs you can also transmit with the quality that one might experience on an older style AM radio. At 8kbs, you can understand a voice if the speaker is clear and pronunciates  their words clearly.  However, it is not likely you could understand somebody speaking quickly or slurring their words slightly.

Real-Time Music, Streaming Audio and Internet Radio

Streaming audio ranges from about 64kbs to 128kbs for higher fidelity.

File Transfer Protocol (FTP)/Microsoft Servicepack Downloads

Updates such as Microsoft service packs use file transfer protocol. Generally, this protocol will use as much bandwidth as it can find. There are several limiting factors for the actual speed an FTP will attain, though.

  1. The speed of your link — If the factors below (2 and 3) do not come into effect, an FTP transfer will take your entire link and crowd out VoIP calls and video.
  2. The speed of the senders server — There is no guarantee that the  sending serving is able to deliver data at the speed of your high speed link. Back in the days of dial-up 28.8kbs modems, this was never a factor. But, with some home internet links approaching 10 megabits, don’t be surprised if the sending server cannot keep up. During peak times, the sending server may be processing many requests at one time, and hence, even though it’s coming from a commercial site, it could actually be slower than your home network.
  3. The speed of the local receiving machine — Yes, even the computer you are receiving the file on has an upper limit. If you are on a high speed university network, the line speed of the network can easily exceed your computers ability to take up data.

While every network will ultimately be different, this field guide should provide you with an idea of the bandwidth demands your network will experience. After all, it’s much better to plan ahead rather than risking a bandwidth overload that causes your entire network to come to a hault.

Related Article a must read for anybody upgrading their Internet Pipe is our article on Contention Ratios

Created by APconnections, the NetEqualizer is a plug-and-play bandwidth control and WAN/Internet optimization appliance that is flexible and scalable. When the network is congested, NetEqualizer’s unique “behavior shaping” technology dynamically and automatically gives priority to latency sensitive applications, such as VoIP and email. Click here for a full price list.

Other products that classify bandwidth

Posted in Bandwidth Monitoring, Educational Articles, Network Capacity. Tags: , , , , , , , , , , , , , , , , , , , , , , . 2 Comments »

Using NetEqualizer to Ensure Clean, Clear QoS for VOIP Calls

April 27, 2010 — netequalizer

A Little Bit of History

Many VoIP installations  are designed with an initial architecture that assumes inter-office  phone calls will reside within the confines of the company LAN. Internal LANs  are almost always 100 megabit and consist of multiple paths between end points. The basic corporate LAN design usually provides more than enough bandwidth to route all inter-office VoIP calls without congestion.

As enterprises  become more dispersed geographically, care must be taken when extending  VoIP calls beyond the main office.  Once a VoIP call leaves the confines of your local network and traverses  over  the public Internet link, it will have to compete for space with any data traffic that might also be destined  for the Internet. Without careful planning, your Enterprise will most likely start dropping VoIP calls during  busy traffic times.

The most common way of dealing with priority or VoIP  is to set what is called the TOS bit.  The TOS bit acts like a little flag inside each Internet packet of the VoIP stream. An Internet router can  rearrange the packets destined for the Internet, and give priority to the outgoing VoIP packets by looking at the TOS bit. The downside of this method is that it does not help with VoIP calls originating  from the outside coming into your network.  For example, somebody receiving a VoIP call in the main office from a VPN user working at home, may experience some distortion on the incoming VoIP  call.  This is usually caused when somebody else in the office is doing a large download during the VoIP call.  Routers typically cannot set priority on incoming data, hence the inbound data download can dominate all the bandwidth, rendering the VoIP call inaudible.

How NetEqualizer Solves VoIP Congestion Issues

The NetEqualizer  solves the problem of  VoIP traffic competing with regular data traffic by using a simple  method. A NetEqualizer provides priority for both incoming and outgoing VoIP traffic . It does not use TOS bits.  It is VoIP and Network agnostic.  Sounds like the old Saturday Night Live commercial where Chevy Chase hawks a floor cleaner that is also an ice cream topping.

Here is how it works…

It turns out that VoIP streams require no more than 100kbs per  call,  usually quite a bit less.  Large downloads, on the other hand, will grab the entire Internet Trunk if they can get it.  The NetEqualizer has been designed to favor streams of less than 100kbs over larger data streams. When a large download is competing with a VoIP call for precious resources, the NetEqualizer will create some artificial latency on the download stream causing it to back off and slow down. No need to rely on TOS bits in this scenario, problem solved.

Conceptually, that is all there is to it.  Obviously, the NetEqualizer engineering team has refined and tuned  this technique over the years.  In general, the NetEqualizer Default Rules need very little set-up, and a unit can be inline in a matter of minutes.

The scenarios where NetEqualizer is appropriate for ensuring that your VoIP system runs smoothly are:

  1. You are running an Enterprise VoIP service with remote offices that connect to your main PBX over VPN links
  2. You are an ISP and your customers use a VoIP service over limited bandwidth connectivity

Recommended Reading

Other vendor White Papers on the subject:  River Bed

Other suggested reading:  http://www.bandwidth.com/wiki/article/QoS_(Quality_of_Service)

Posted in Bandwidth Management, Educational Articles. Tags: , , , , , , , , . 3 Comments »

Speeding up Your T1, DS3, or Cable Internet Connection with an Optimizing Appliance

April 30, 2009 — netequalizer

By Art Reisman, CTO, APconnections (www.netequalizer.com)

Whether you are a home user or a large multinational corporation, you likely want to get the most out of your Internet connection. In previous articles, we have  briefly covered using Equalizing (Fairness)  as a tool to speed up your connection without purchasing additional bandwidth. In the following sections, we’ll break down  exactly how this is accomplished in layman’s terms.

First , what is an optimizing appliance?

An optimizing appliance is a piece of networking equipment that has one Ethernet input and one Ethernet output. It is normally located between the router that terminates your Internet connection and the users on your network. From this location, all Internet traffic must pass through the device. When activated, the optimizing appliance can rearrange traffic loads for optimal service, thus preventing the need for costly new bandwidth upgrades.

Next, we’ll summarize equalizing and behavior-based shaping.

Overall, equalizing is a simple concept. It is the art form of looking at the usage patterns on the network, and when things get congested, robbing from the rich to give to the poor. In other words, heavy users are limited in the amount of badwidth to which they have access in order to ensure that ALL users on the network can utilize the network effectively. Rather than writing hundreds of rules to specify allocations to specific traffic as in traditional application shaping, you can simply assume that large downloads are bad, short quick traffic is good, and be done with it.

How is Fairness implemented?

If you have multiple users sharing your Internet trunk and somebody mentions “fairness,” it probably conjures up the image of each user waiting in line for their turn. And while a device that enforces fairness in this way would certainly be better than doing nothing, Equalizing goes a few steps further than this.

We don’t just divide the bandwidth equally like a “brain dead” controller. Equalizing is a system of dynamic priorities that reward smaller users at the expense of heavy users. It is very very dynamic, and there is no pre-set limit on any user. In fact, the NetEqualizer does not keep track of users at all. Instead, we monitor user streams. So, a user may be getting one stream (FTP Download) slowed down while at the same time having another stream untouched(e-mail).

Another key element in behavior-based shaping is connections. Equalizing takes care of instances of congestion caused by single-source bandwidth hogs. However, the other main cause of Internet gridlock (as well as bringing down routers and access points) is p2p and its propensity to open hundreds or perhaps thousands of connections to different sources on the Internet. Over the years, the NetEqualizer engineers have developed very specific algorithms to spot connection abuse and avert its side effects.

What is the result?

The end result is that applications such as Web surfing, IM, short downloads, and voice all naturally receive higher priority, while large downloads and p2p receive lower priority. Also, situations where we cut back large streams is  generally for a short duration. As an added advantage, this behavior-based shaping does not need to be updated constantly as applications change.

Trusting a heuristic solution such as NetEqualizer is not always an easy step. Oftentimes, customers are concerned with accidentally throttling important traffic that might not fit the NetEqualizer model, such as video. Although there are exceptions, it is rare for the network operator not to know about these potential issues in advance, and there are generally relatively few to consider. In fact, the only exception that we run into is video, and the NetEqualizer has a low level routine that easily allows you to give overriding priority to a specific server on your network, hence solving the problem. The NetEqualizer also has a special feature whereby you can exempt and give priority to any IP address specifically in the event that a large stream such as video must be given priority.

Through the implementation of Equalizing technology, network administrators are able to get the most out of their network. Users of the NetEqualizer are often surprised to find that their network problems were not a result of a lack of bandwidth, but rather a lack of bandwidth control.

See who else is using this technology.

Created by APconnections, the NetEqualizer is a plug-and-play bandwidth control and WAN/Internet optimization appliance that is flexible and scalable. When the network is congested, NetEqualizer’s unique “behavior shaping” technology dynamically and automatically gives priority to latency sensitive applications, such as VoIP and email. Click here for a full price list.

Posted in Bandwidth Management, Educational Articles, Network Speed. Tags: , , , , , , , , , , , , . 2 Comments »

Hotel Property Managers Should Consider Generic Bandwidth Control Solutions

March 1, 2009 — netequalizer

Editors Note: The following Hotelsmag.com article caught my attention this morning. The hotel industry is now seriously starting to understand that they need some form of bandwidth control.   However, many hotel solutions for bandwidth control are custom marketed, which perhaps puts their economy-of-scale at a competitive disadvantage. Yet, the NetEqualizer bandwidth controller, as well as our competitors, cross many market verticals, offering hotels an effective solution without the niche-market costs. For example, in addition to the numerous other industries in which the NetEqualizer is being used, some of our hotel customers include: The Holiday Inn Capital Hill, a prominent Washington DC hotel; The Portola Plaza Hotel and Conference Center in Monterrey, California; and the Hotel St. Regis in New York City.

For more information about the NetEqualizer, or to check out our live demo, visit www.netequalizer.com.

Heavy Users Tax Hotel Systems:Hoteliers and IT Staff Must Adapt to a New Reality of Extreme Bandwidth Demands

By Stephanie Overby, Special to Hotels — Hotels, 3/1/2009

The tweens taking up the seventh floor are instant-messaging while listening to Internet radio and downloading a pirated version of “Twilight” to watch later. The 200-person meeting in the ballroom has a full interactive multimedia presentation going for the next hour. And you do not want to know what the businessman in room 1208 is streaming on BitTorrent, but it is probably not a productivity booster.

To keep reading, click here.

Posted in Commentary and Editorials, News. Tags: , , , , , , , , , , , , , , , , . Leave a Comment »

Virtual PBX revisited

January 15, 2009 — netequalizer

Editors Note:

This article written for VOIP magazine back in 2004 is worth revisiting.

Back in 2004 when I first wrote this article for the most part there was nothing commercially available  now, Jan 2009, the market is crowded with offers claiming to be virtual PBX’s . At APconnections, we currently use an offering from Aptela.com.  A true virtual PBX. Make sure you look under the hood at anything you evaluate.  All  the 800 service numbers call themselves virtual PBX’s; however, in our opinion, simply having a call answer service in the sky  is not a PBX. Read on for a detailed definition.

Before reposting we searched for the original but were unable to find it online.

—————————————————————————————————-

Art Reisman

By Art Reisman, CTO, APconnections makers of NetEqualizer Internet Optimization Equipment

Outsourcing Communications with a Virtual PBX

CTO http://www.apconnections.net http://www.netequalizer.com

A new breed of applications emerging from the intersection of VoIP and broadband may soon make the traditional premise-based PBX a thing of the past. Virtual PBX, hosted and delivered by today’s telcos and cable operators, is quickly becoming an option for businesses looking to outsource portions of their communications network. Rather than purchase and maintain an expensive piece of equipment, you can now sign up for a pay-as-you-go service with all of the functionality of an on-site PBX but with none of the expense.


To some, this idea may sound like a return to the past and, in a sense, it is. AT&T began delivering PBX functionality through its Centrex services in the 1970s. However, upon closer investigation, it is clear that the functionality delivered and the economics of the two approaches are very different.

The Private Branch Exchange: A Brief Primer

A PBX or private branch exchange allows an organization to maintain a small number of outside lines when compared to the number of actual telephones and users within an organization. Users of the PBX share these outside lines for making telephone calls outside the organization (external to the PBX).

Onsite PBX became popular and matured in the 1980s when the cost of remote connectivity was extremely high and the customer control of hosted PBX-like services of the time (Centrex) was limited, if it was even offered. In 1980, providing advanced, remote PBX services to a building with 100 employees would have required AT&T to run 100 individual copper lines from the local exchange to each telephone at the site.

As more and more businesses opted to install a PBX onsite, competition for customer dollars drove ever more extensive “business-class” features into these devices, further differentiating the premise-based PBX from the hosted products offered by telephone companies. Over time, PBX offerings gradually standardized into the product set that today we have come to expect when we pick up any business phone: voice-mail, auto attendant, call queuing, conferencing, call transfer, and more.

Flash forward from 1980 to 2005. Today, 100 direct phone lines can be transported from one location to another over many miles with no more than one wire. Remote access to control a PBX outside of your building is also trivial to implement with a simple Web portal. Technological advances coupled with feature stability and the broad appeal of PBX “applications” makes them a prime candidate for hosting.

A business starting today can have a full-featured hosted PBX with a single high-speed Internet connection. These virtualized services would require no additional equipment to purchase or maintain.

Defining Virtual PBX

Businesses looking to purchase such a service today can expect to find significant differences in the features and functionality available among offerings being marketed under the, often interchangeable, terms hosted or virtual PBX. To alleviate confusion and provide a starting point in your quest to outsource your communications network, the perfect, hosted PBX service would have the following features:

Auto-detectionThe PBX must dynamically detect remote stations from any place in the world and provide dial tone (As opposed to having a user dial in to obtain service. See the sidebar, Start with a Dial Tone).
Start with a Dial Tone
There are products on the market that remotely host a set of PBX services and require the user to dial in with a standard phone so the PBX can identify the caller. This is a viable approach to providing a hosted PBX with established stability. However, it does have a few restrictions not applicable to a pure hosted PBX.

  • When using the PBX services, the caller ties up a local phone line and blocks calls directly made to that line.
  • Obtaining a dial tone for an outbound call can only be done by first connecting to the PBX, or as a final alternative just using the standard phone line to dial out without going through the PBX, which takes away all of the cost and convenience benefits of the PBX.

    • A truly hosted PBX solution must provide a dial tone without first dialing in.

    Service Provisioning New service provisioning must be self-service with no expensive customer premise equipment required. For example, a customer with a credit card and access to a provider’s Web page should be able to initiate worldwide service in a matter of minutes.

    Standards Support Off-the-shelf SIP phones must be supported by the hosted service. A virtual PBX should not lock customers into using specific equipment or proprietary protocols.

    Affordable Start-up costs should be minimal and usage-based, allowing a small business to seamlessly grow and add stations as needed, without ever needing a disruptive upgrade or requiring a large capital investment.

    Level Rates Outbound and inbound toll rates should be provided at wholesale prices globally by the service provider. The customer can be assured of one published competitive price for outgoing calls and incoming calls.

    Administration Each business using the service should have access to a private portal allowing them to administer features and options. The organization’s account and services should be secure and accessible to a designated administrator 24/7.

    Bundled Applications The service must offer a minimum set of applications common to an onsite PBX. The most common of which include: transfer, conference, forward, find me, follow me, voice mail, auto attendant, basic call reporting, and inbound and outbound caller ID.

    Technology Considerations

    While the benefits to a hosted PBX solution are immediately obvious–elimination of equipment hard costs and the specialized knowledge required to keep it up and running–there are drawbacks to consider when adopting an emerging technology.

    The first point to consider is that the technology behind hosted PBX services has not yet developed to the point of large-scale enterprise deployments. Currently, the organizations that will see the most benefit from a hosted solution are small- to medium-sized businesses.

    Quality of service, the shadow that follows every voice over IP application, is the overriding technology hurdle that consumers need to be aware of when considering a hosted PBX solution. Latency can also be an issue; the different routes that IP data takes across the Internet can cause speech breaks and dropped calls.

    QoS and latency are key considerations when discussing bandwidth requirements and network architecture with potential vendors. Being undersold on bandwidth when moving to an IP communications network can create problems above and beyond being oversold.

    Selecting a Vendor

    The low barrier to entry for vendors looking to offer hosted PBX services has created a number of options for consumers and driven down costs, but customers need to be aware that not all service providers are equal.

    Existing Infrastructure Deploying a world-wide hosted PBX service as outlined above requires a significant infrastructure investment to handle the centralized switching needed to move millions of simultaneous call around the world. When investigating service providers, look for a vendor that has the knowledge to grow not only with your business but also with the broad adoption of the technology as a whole. Having a tested, existing infrastructure in place for business-class communications is key.

    Service Provider Network One method of alleviating IP voice quality issues on a regional basis is by staying within a large service provider network. For example, if an organization uses a Qwest T3 trunk service at its headquarters and an employee travels to neighboring cities with Qwest DSL service in their hotels, it is unlikely that quality problems will be experienced at the carrier level. Choosing a vendor that understands how your organization will use the service should be an important part of your selection process.

    Conclusion

    While adoption is not yet widespread, hosted services are here and will only get better with time. As companies continue to seek the benefits of outsourcing the elements of their enterprise–from business processes to core technologies—adoption will continue to grow, making hosted PBX is a technology to keep your eye on in 2005.

    Note the author uses a solution from Aptela and has found their support to be top notch and was the main reason for switching about 4 years ago.

    Posted in Educational Articles. Tags: , , , , , . 5 Comments »

    QoS on the Internet — Can Class of Service Be Guaranteed?

    July 24, 2008 — netequalizer

    Most quality of service (QoS) schemes today are implemented to give priority to voice or video data running in common over a data circuit. The trick used to ensure that certain types of data receive priority over others makes use of a type of service (TOS) bit. Simply put, this is just a special flag inside of an Internet packet that can be a 1 or a 0, with a 1 implying priority while a 0 implies normal treatment.

    In order for the TOS bit scheme to work correctly, all routers along a path need to be aware of it. In a self-contained corporate network, an organization usually controls all routers along the data path and makes sure that this recognition occurs. For example, a multinational organization with a VoIP system most likely purchases dedicated links through a global provider like ATT. In this scenario, the company can configure all of their routers to give priority to QoS tagged traffic, and this will prevent something like a print server file from degrading an interoffice VoIP call.

    However, this can be a very expensive process and may not be available to smaller businesses and organizations that do not have their own dedicated links. In any place where many customers share an Internet link which is not the nailed up point-to-point that you’d find within a corporate network, there is contention for resources. In these cases, guaranteeing class of service is more difficult. So, this begs the question, “How can you set a QoS bit and prioritize traffic on such a link?”

    In general, the answer is that you can’t.

    The reason is quite simple. Your provider to the Internet cloud — Time Warner, Comcast, Qwest, etc. — most likely does not look at or support TOS bits. You can set them if you want, but they will probably be ignored. There are exceptions to this rule, however, but your voice traffic traveling over the Internet cloud will in all likelihood get the same treatment as all other traffic.

    The good news is that most providers have plenty of bandwidth on their backbones and your third party voice service such as Skype will be fine. I personally use a PBX in the sky called Aptela from my home office. It works fine until my son starts watching YouTube videos and then all of a sudden my calls get choppy.

    The bottle neck for this type of outage is not your provider’s backbone, but rather the limited link coming into your office or your home. The easiest way to ensure that your Skype call does not crash is to self-regulate the use of other bandwidth intensive Internet services.

    Considering all of this, NetEqualizer customers often ask, “How does the NetEqualizer/AirEqualizer do priority QOS?”

    It is a very unique technology, but the answer is also very simple. First, you need to clear your head about the way QoS is typically done in the Cisco™ model using bit tagging and such.

    In its default mode, the NetEqualizer/AirEqualizer treats all of your standard traffic as one big pool. When your network is busy, it constantly readjusts bandwidth allocation for users automatically. It does this by temporarily limiting the amount of bandwidth a large download (such as that often found with p2p file sharing) might be using in order to ensure greater response times for e-mail, chat, Web browsing, VoIP, and other everyday online activities.

    So, essentially, the NetEqualizer/AirEqualizer is already providing one level of QoS in the default setup. However, users have the option of giving certain applications priority over others.

    For example, when you tell the NetEqualizer/AirEqualizer to give specific priority to your video server, it automatically squeezes all the other users into a smaller pool and leaves the video server traffic alone. In essence, this reserves bandwidth for the video server at a higher priority than all of the generic users. When the video stream is not active, the generic data users are allowed to utilize more bandwidth, including that which had been preserved for video. Once the settings are in place, all of this is done automatically and in real time. The same could be done with VoIP and other priority applications.

    In most cases, the only users that even realize this process is taking place are those who are running the non-prioritized applications that have typically slowed your network. For everyone else, it’s business as usual. So, as mentioned, QoS over the NetEqualizer/AirEqualizer is ultimately a very simple process, but also very effective. And, it’s all done without controversial bit tagging and deep packet inspection!

    Posted in Commentary and Editorials. Tags: , , , , , , , , . 1 Comment »

    VoIP Call Quality Hindrances, Meet NetEqualizer

    April 25, 2007 — netequalizer

    Erik Linask, associate editor of  Internet Telephony Magazine, discusses NetEqualizer’s VoIP QoS capabilities in this April article. Read it here.

    Posted in News. Tags: , , . Leave a Comment »
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