RonEK Communications

postheadericon Remember the 3 D’s when dealing with ISP’s

It’s no secret that I am a big fan of Pathview (both Premise & Cloud) and since becoming a partner last year it has really made a difference in our ability to assess, troubleshoot and monitor our customers’ network.  When first watching a few of the UTube infomercials about Pathview, one of the terms that caught my attention was ‘MTI’ – Mean Time to Innocence.  I thought immediately that the speaker must have had experience as a phone guy somewhere in his past and his soul, like mine, had been scarred for life!

Coming from a voice background, one of the first things you learn is that you are guilty of  all things until you prove – and often to the satisfaction of  completely clueless people (see figure 2) – that you or the equipment you installed was not at fault for something not working.  Furthermore,  just because you correctly identified the problem as not being yours does not a) make you popular or b) excuse you from having to fix it anyway.   Our customers have greatly benefited from Pathview and, one would think, that because of the highly technical nature of networks, ISP’s would welcome the assistance from someone armed with such an application.  Amazingly, this is not so.

Figure 1. Customer reported that the Internet seemed to suddenly slow to a crawl. ISP response - “We are testing good to the modem at 3Megs. Contact your IT person. It’s obviously your equipment.” After hours of arguing with them (Comcast), it turned out to be their faulty modem.

True story

Those of you familiar with Wireshark know it to be a very popular packet sniffer. If you have been looking at Wireshark recently you will know the name of Laura Chappell and her very thorough book “Wireshark Network Analysis” released last March.  In one of her case studies was a very amusing yet oh-so-true true situation.

It was a familiar story of a user having trouble connecting to the corporate office and the IT department pointing the finger towards the ISP.  The ISP calmly assured them that they were not blocking any traffic.  When confronted with a packet capture showing otherwise,  the ISP confirmed that they were – as they put it – blocking  “ports in question”.  Begrudgingly, the ISP allowed these port through but with a few caveats.  In an almost whimsical way, the author ended his report as follows;

“We now have a happy user, but I can’t help but wonder how many other customers of this ISP are encountering similar issues and wondering why it takes so many attempts to get connected to their corporate network.”

If you haven’t had an experience like this with an ISP, you are either very new, very lucky or VERY oblivious.  In the words of Morphius to Neo …. “Welcome to the real world.”

Here’s the fact, ISP’s have for years held – and lorded over – their two trump cards to end users and vendors alike, i.e. “We’re more technical than you could ever hope to be” followed closely by “Prove it …!” said usually with a subtle yet detectable sneer either on the phone or through an email.  Unfortunately, things like “it seems slow …” or “I don’t think we are getting the bandwidth we are paying for …”, or even, “my voice is definitely choppy at certain times of the day …” just doesn’t cut it with these guys.

Baselining – Getting Prepared for the 3 D’s

When a customer approaches us with their story of woe, the first thing we do is establish a baseline with Pathview using targets both inside their network and outside to one of our micro appliances.   The more elusive or intermittent the problem, the longer the base line needs to be and by that I mean minimum one day and as much as a week or even longer.  It is also recommended to see what their system looks like when things are being backed up.

Field note – if the customer can’t tell you when the system gets backed up, it is probably related to when they are having some type of network problem.

Figure 2. The weekend staff kept complaining about the Internet being slow even when only 4 of them were watching the ball games … on their pc’s.

I usually tell the customer that if the problem looks like the carrier, it will take a minimum of 4 calls to their Customer Care/Support center to get to someone who can actually do something.  If you are a consultant, this is something that you charge for (what?! you don’t think a lawyer would charge for his time to do this?).  Once contact with the carrier has been made, you will begin with the first of the 3 D’s.


The way ISP’s decide to address a customer’s problem depends on their technical level in the Customer Service Center which generally falls under one of the following two axioms:

Axiom 1 – The less technical they are, the more they will want to sell you additional bandwidth.

Slow Internet? More bandwidth.

Getting cut off altogether?  More bandwidth.

PC making strange noises? More bandwidth.

Think some of your ports are being blocked?  Get more bandwidth and, oh by the way,

there may be an extra charge for unblocked ports (whatever they are).

Axiom 2 – The more technical they are, the likelier they are to put all the blame on the user’s equipment.

Usually they will say something like “Well, I’m logged into your router now and I don’t see any packet loss at this time or in the last 48 hours …”.  One time with the NOC guy on the line, I told him that I was going to make a programming change in the router and actually unplugged the carrier’s equipment.

“How does it look now?”  I asked after about a minute.

“Looks the same, no packet loss.”
No surprise, he was looking at the wrong circuit all along.  That one was easy.

The hard ones are when they are actually looking at the right circuit and for some reason insist they are not seeing what you are seeing.  This, by the way, is what sold me on Pathview and led me to an almost X-File motto,

“The truth is out there … somewhere, but you will need Pathview to find it.”

(I should probably copyright that before Jim grabs it!).

In the chart below,  the customer had installed an IP based pbx and had just converted to SIP trunks.  They kept losing calls and their voice quality was very choppy – other than that, it was great …

Figure 3. Using Pathview Cloud looking towards the SIP provider

The SIP provider, Broadvox in this case, insisted that the problem was not at their end and to contact their local carrier.  The carrier, a wireless outfit, claimed that it could not be anything at their end so it had to be the customer’s equipment.

A quick look at their connection from our point of view (outside-looking-in to their pbx using Pathview Premise) and then to the SIP provider from the customer’s point of view (inside-looking-out using Pathview Cloud), shockingly revealed that the problem wasn’t the customer’s equipment at all but was instead the hop (in this case, the  wireless tower) just past the customer’s router.  Granted, the customer was in a remote area (northeast Texas) which is why they HAD to go with wireless for a while (but this was about to change).   The customer and I called the carrier and incredibly, got a hold of the operations manager on the first try.  Thus began the second “D” when dealing with the ISP.


He had us on the speakerphone and it seemed as though he was trying to demonstrate to someone(s) in the room how to deal with a complaining end user.  In rapid fire he began to drill us –

“How much packet loss? I will need to know the exact percentage and where it is occurring”

“When did this start?”

“Why are you just noticing it now?”

“How come we’re not seeing it?”

“Are you sure you have power?”

“Have you replaced your equipment?”

“Is it raining there?” ç That was an interesting one!

“We have VoIP sets and we’re not having problems.”

There were a few others whose relevance I questioned but it was clear that customer sensitivity was not in the field ops hand guide.  The only question he didn’t ask is whether we though we needed more bandwidth but then again he was obviously an Axiom 2 guy. He finally ended it with a ‘and-don’t-call-me-back-until-you-have-all-this-information’ “Okay?”.

For a moment it seemed as though he was leaning back in his chair while looking at his understudies with the air of  “ … and THAT’S how you handle that!”.  At the same time I heard my customer softly chuckle since he had already seen the Pathview charts and tests.  Needless to say, we were quickly placed on hold and picked up elsewhere – not on speakerphone  – and he asked us to email the information.  I don’t know if he actually looked at it or not but within moments a service call was scheduled with the results in figure 4.

Figure 4. “Okay, should be working fine now. The tech ran a few DSL Reports tests and then a traceroute for good measure. Looks great!” Are you kidding me?!

I was stunned.  Yes, it looked better but the customer was, well, let’s just say he was still experiencing problems.  This time the customer forwarded the reports to the ops manager and anyone whose email address he had.  On to the third “D”.


The response was that they did not see any problems – which also falls under “DENY” – but they promised to continue monitoring it and get back to us.  After a few days of non-returned emails and phone calls it was clear they were going to leave it as is and just wait us out.   Long story short, the customer switched Internet providers the next week and, while not perfect, things greatly improved.   I also heard that they raised the rent for the carrier’s repeater that was on their property.

Figure 5. “We don’t measure MOS in this department but that looks normal.” Is there someone I can go over this with? Hello? … Hello?

In the very old days of T-1 (for both voice and data),  and to a much lesser extend today,  the only way to really trouble shoot the circuit was to go on premise with a T-Berd and do a head-to-head test which meant disconnecting the circuit altogether from any customer equipment and start running tests directly back to the central office.  This was an after hours adventure known as a “vendor/telco meet” and had to be scheduled a few days in advance unless you happened to have your own T-Berd (which was not cheap) in which case you could almost do it on the fly.  The upshot was there was usually a conclusion one way or the other, the guilty were persecuted and sentenced while the innocent were absolved.

Figure 6. “We’re not seeing anything but go ahead and send us your graphs and we’ll forward them up the line.” Two days and 4 calls later I got a hold of a sharp router tech who found the problem in 15 minutes. “User provided graphs? No …. I’m not sure what they do with that stuff.”

But that was when the playing field consisted of AT&T, GTE/Verizon and then everyone else.  Nowadays, it’s all about how to repackage the same service that everyone else has and only worry about the larger customers.  The smaller the fish, the more they are going to have to put up with – “ ….just let them try to get out of our contract!”  Ironically, this is actually good for companies like us because the SMB’s of the world just simply don’t know who to turn to.

Parting Shots

When we started using Pathview, aside from the occasional ticker tape parade,  I was looking forward to the time and aggravation we would save ourselves and the customer.  What we got was all of the above along with the revelation that ISP’s ;

1)      Don’t look at packets they way they are actually used

2)      Don’t want your help when trouble shooting

3)      Often don’t have the software  tools, training or even the inclination to  look beyond the single port of a router.

4)      Would rather put you or your customer through the 3 D’s than to actually fix the problem.  How that happened is for another blog.

Copyright Eric Knaus 2010

postheadericon Are You VoIP Ready? – Glossary

Bandwidth Saturation The point in which all available bandwidth on an Internet

connection is used up.

Bandwidth The amount of data passing through a connection over a given time.
It is usually measured in bps (bits-per-second) or Mbps (Megabits
per Second). As a general rule, get as much as you can afford – and
the make sure you are getting it.
Content Filtering On the Internet, content filtering (also known as information

filtering) is the use of a program to screen and exclude from access

or availability Web pages or e-mail that is deemed objectionable.

Content filtering is used by corporations and governments as part of

Internet firewall computers and also by home computer owners,

especially by parents to screen the content their children have access

to from a computer

Dropped Packets Packets (i.e. small data “packages”) are occasionally dropped, or

lost, on the network for various reasons. For instance, two nodes

may be communicating at widely disparate transfer rates. TCP

packets are resent, UDP s are not.

Hop In a packet-switching network, a hop is the trip a data packet takes

from one router or intermediate point to another in the network. On

the Internet (or a network that uses TCP/IP), the number of hops a

packet has taken toward its destination (called the “hop count”) is

kept in the packet header.

ICMP Internet Control Message Protocol is a message control and

error-reporting protocol between a host server and a gateway to the

Internet. ICMP uses Internet Protocol (IP) datagrams, but the

messages are processed by the IP software and are not directly

apparent to the application user.

ISP An ISP (Internet Service Provider) aka Carrier, aka Provider, is a

company that collects a monthly or yearly fee in exchange for

providing the subscriber with Internet access.

Jitter The difference in latency from one packet to the next measured in


LAN A Local Area Network is a computer network that spans a

relatively small area. Most LANs are confined to a single building or

group of buildings. However, one LAN can be connected to other

LANs over any distance via telephone lines and radio waves. A

system of LANs connected in this way is called a wide-area network


Latency In a network, latency, a synonym for delay, is an expression of how

much time it takes for a packet of data to get from one designated

point to another. Typically, latency is measured by sending a packet

that is returned to the sender. The round-trip time – measured in

milliseconds – is considered the latency.

NIU A Network Interface Unit (sometimes called a network interface

device) is a device that serves as a common interface for various

other devices within a LAN , or as an interface to allow networked

computers to connect to an outside network.

Ping Loosely, ping means “to get the attention of” or “to check for the

presence of” another party online. Ping operates by sending a packet
to a designated address and waiting for a response. The computer
acronym (for Packet Internet or Inter-Network Groper) was
contrived to match the submariners’ term for the sound of a returned
sonar pulse.

Point to Point Point-to-point telecommunications generally refers to a connection

restricted to two endpoints, usually host computers.

PSTN Public Switched Telephone Network is the world’s collection of

interconnected voice-oriented public telephone networks, both

commercial and government-owned

QoS Quality of Service. Describes the ability of a e.g. router to prioritize

certain packets

SIP Session Initiation Protocol is an application-layer control

(signaling) protocol for creating, modifying, and terminating

sessions with one or more participants. It can be used to create

two-party, multiparty, or multicast sessions that include Internet

telephone calls, multimedia distribution, and multimedia


TDM Short for Time Division Multiplexing, a type of multiplexing that

combines data streams by assigning each stream a different time slot

in a set. TDM telephone sets (often referred to as digital  sets)

differ from IP sets in that they do not go on a LAN s infrastucture

are compatible with analogue wiring schemes and can work on cable

runs oup to 1,600 feet.

VPN (pronounced as separate letters) Short for Virtual Private Network,

is a private network that uses a public network (usually the Internet)

to connect remote sites or users together. VPNs use “virtual”

connections routed through the Internet from a company’s private

network , a remote site or employee.

WIC WAN Interface Card is installed in a router and is the component

that a Internet T-1 will physically plug in to.

postheadericon Are You VoIP Ready? – The Road to China: Content Filtering to the Max

ChinaNET is managed by the Data Communications Bureau of the Ministry for Posts and Telecommunications, and provides Internet service in all 31 provincial capitals in mainland China. It is one of the two major commercial networks approved by the State Council, the other being ChinaBGN. For this reason, Figure 6 is one of my favorite sites to watch, not because it has great VoIP possibilities – because it does not – but because you can capture the business heart beat of a nation along with the ideology of a government just by viewing this graph over a week s
time. The target site is in a town just south of Shanghai called Hangzhou. The part I find most interesting is that you can tell the moment you hit mainland China (hop 13) because the latency skyrockets from 62ms to 449ms. This is a classic example of Content Filtering  that ChinaNet does in order to keep certain things out of their country. Fortunately, ICMP packets are not one of them, so once we get past the censors, you can see that even within the mainland, there is an overall increase in latency to the final destination – this hints at content filtering within the borders as well. Overall, you can see what the average Chinese Internet user experiences in terms of latency over a weeks time. The graph shows a 7 day cycle and within each day cycle you can see a consistent dip just a little past half way which I found out later was when they – you guessed it – took the equivalent of lunch. This also shows that the basic internal data-transport infrastructure is under a severe load and the chances of VoIP running well WITHIN China are slim if you have to go more than a handful of hops. This might be another reason the Chinese government blocks most of the incoming Internet traffic – the network just could not handle it!
As an aside, when I first started watching this site about a year ago, I was able to see ChinaNet  in the DNSName column. About 7 months ago they removed any identification other than the IP address.
The client originally asked me to see if they could have a telephone connected via VoIP from California to this site and the answer was an emphatic No! . We considered a satellite solution but found out that there were restrictions on this as well. Besides, satellite in general,
has very high latency (too high for decent voice, in my opinion) and is susceptible to bad weather. So as of this article, they are simply resorting to email and regular PSTN connections to communicate.

postheadericon Are You VoIP Ready? – What Stable Connections Look Like

Figure 5 is a good example of a connection that should work for your VoIP application. In this screen shot, showing a 24 hour segment, you can see that there are only a total of 5 samples that go into the yellow area. The vast majority are in the lower middle of the green band with the average latency at 73ms and jitter of 4ms. As such, I would tell the customer that this circuit meets my criteria of VoIP readiness. The only thing I do not know for sure is how much bandwidth they have. Sometimes the customer will know and other times they will think  they know. If it is a T-1 connection, then you can be fairly certain that you are getting 1.54M up AND down and base your voice session calculations off of that. If it is a DSL or cable connection, you will most likely experience swings in latency as usage (voice and data) goes up.

postheadericon Are You VoIP Ready? – The “X” Factor

One X factor you will need to consider when looking at a VoIP solution is your network s vulnerability to viruses, worms and Trojans. The first thing I caution customers about when they want to go all out and purchase a pure  IP telephony solution is that as a general rule, you want to keep your local voice and local data traffic separate. In practice, this means if you already have a voice infrastructure (i.e. jacks specifically for telephone
sets that home run to a main telephone room or the IT Head End  room without connecting to the LAN wiring), put your voice on that with TDM sets instead of abandoning the wires that are in place.
Voice and data packets going out to other offices that are linked to you through a Point to Point or VPN connection will inevitably share time on the same Internet circuit. However, TDM sets are not affected by anything on the LAN until they have to connect to some off site device through VoIP. Regular local traffic such as voice mail retrieval, intercom calls, paging to the warehouse or calling a supplier over the PSTN will occur with or without a data switch or server in place.
If you do decide to go all IP, then do it when you can make a fresh wiring start and run a separate cable for voice and data sets. Also, keep the voice devices on their own subnet separated from the other LANs by a decent router. RonEK is not a Cisco reseller but we like their routers and recommend them in cases like this.
Many IT people would take issue with this approach but I have personally witnessed and been victim to what can happen as shown in Figure 4. As you can see, everything looks great until the main server on the LAN was hit by a very aggressive virus. For about 2 hours it created such havoc on the LAN in the form of broadcast storms that all network traffic was reduced to a crawl and VoIP was stopped cold.

postheadericon Are You VoIP Ready? – QoS (Quality of Service)

Normally, when packets are serialized out the router to the Internet, they are sent in a first come first serve  fashion. If your router is equipped with QOS, packets from your PBX or SIP server* can be prioritized ahead of the other non-voice packets thereby keeping the flow of
voice traffic relatively smooth. Most carriers that offer a combined package of voice and data services do just that. RonEK is a partner with several ISPs and one of the first questions on the vendor check list is the IP address of the PBX. Most of the higher end carriers will provide an
end to end managed circuit  which means that they can control the connection from your office to their space in the central office. From there, packets are routed out on their backbone or someone else s backbone depending on the carrier s capacity and the final destination.
Keep in mind that QOS prioritizes packets going out to the other end. Once a packet leaves your premise or your provider s backbone, it is no longer prioritized and subject to the winds and tides of the Internet just like all the other packets. So, when designing a multi-site
network, try to stick with one provider and, if possible, try to do it on an MPLS platform. Usually if you stay within a provider s backbone from end to end, the prioritization will be maintained throughout the connection. Also, there is no way to prioritize packets coming to you until they actually get to you. Many times customers think that if they simply implement QOS that all their voice issues will go away not realizing that they only addressed half of the potential problem. Your connection and QOS is just part of the overall voice session that YOU control. The rest is in the hands of the intermediary (often there is more than one) that controls the path of the packets and then finally, the ISP and equipment at the final destination.
* More about SIP servers in coming articles.

postheadericon Are You VoIP Ready? – Latency

Every IT staff member knows that when you ping something, in addition to confirming that a device is connected to the network, the reply will give you the round trip time in milliseconds from the device you are pinging. A ping is a type of ICMP packet (along with the commonly used trace route command) that you can use to determine just how much delay your packets will experience from point A to point B and back. All of the graphs you see in this article are latency based and bandwidth availability – or lack thereof – has to be extrapolated from that.
Realistically, there is no way to know how much bandwidth some one has by simply looking at their connection from the outside  unless you are
sitting in the Central office looking directly at the connection. That said, there are several things that latency will tell you. I usually try to get a week s worth of data before I m comfortable with the circuit – if there is a problem, it will likely show up within that sample.
The magic latency number I like to see when testing for VoIP usability between sites is 80ms or lower. Another term that is directly related to latency is jitter . Jitter is caused when packets leaving a source in a certain order and spacing, arrive at the destination in the same order
(usually) but with different spacing. It is essentially the difference in latency time from one packet to the next. When jitter is high – anything over 15% variance between samples – it usually points to bandwidth problem. To get an idea of the impact of high jitter, imagine the sound of an
audio CD that is played while alternating between pause and fast-forward. The garbled sound is characteristic of jitter.
If you are going to network offices within the same city, you should see ping times of around 30ms or less and jitter under 5ms. The further across the country you go, the higher the latency tends to get. As of this article, a typical ping time from Houston to Los Angeles is
between 54ms and 67ms. Surprisingly, latency from Houston to Hong Kong is only 62ms to 87ms. I was in Switzerland not too long ago and the ping time was only 75ms from Bern to Los Angeles. My point is that, while geographical distance is an issue, it is not going to be the
determining factor of whether your VoIP project is going to work or not. Things that will affect whether your voice packets will get there in reasonable time or not is QOS (which stands for Quality of Service), the ISP or Carrier, bandwidth, hardware and hardware configuration.
To get an idea of what mis-configured hardware looks like, take a look at Figure 3. The IT manager had just moved into a new facility and users were complaining to him about slow Internet speed. As you can see, the latency is pretty good but the amount of dropped packets was
very high. The poor guy spent the better part of 3 weeks arguing with the carrier about the problem. Their contention was that the source of the problem was at his end as they showed everything good when they tested up to the NIU. They also ran diagnostics on the router – that
they supplied – and that also produced nothing. Technically, they were right except for one important setting that would not show up on any diagnostic. The Clock Source  setting for the router was configured as Internal  – i.e. it was referencing itself as a clock source – instead of
clocking off the network (sometimes referred to as Recovered  mode). For the most part, it worked but any time the router or carrier s clock drifted slightly, packets would be dropped.

postheadericon Are You VoIP Ready? – Bandwidth

Bandwidth -
When measuring your needs and potential use for VoIP you need to understand the type of compression (aka codec) your device (eg. your PBX) is going to use. The three most common types are g.711, g.723 and g.729 and it essentially refers to the number of samples of your voice
the device is going to send over the wires to the far end.
G.711 uses the highest sampling rate and consequently uses the highest amount of bandwidth - typically 80k to 90k per session (or per conversation). This means that over a T-1 connection (1.54m of bandwidth) you will be able to sustain 16 simultaneous conversations or voice sessions. Most VoIP services like Vonage state in their contract that the user understands that they will have to have at least 90k to sustain a VoIP conversation (implying a g.711 compression). In theory, you get the best voice quality (you will see this often referred to as Toll Quality ) with this codec and for the most part that is true. If you intend to run faxes or dial-up modems over this connection, g.711 is about the only way to go (there are some exceptions to this but not many).
The main down side to this compression, of course, is the bandwidth requirement. If you have off site users (the CSR agents working out of their homes, for example) using a DSL or cable connection, this could be an issue because you need the 90k for inbound AND outbound audio.
For these type of circuits, the downstream speed (i.e. packets coming to you from the Internet) is usually much greater than upstream speed (packets you send to the Internet). I often hear from end users that they get 3M up and down from their provider for $40/mo but upon closer
inspection, their contract actually says something like … you may experience speeds UP TO 3M …  which usually translates into something like 700k down and maybe 256k up on average – if that.
G.723 and g.729 codec sample rates are much lower and therefore use much less bandwidth. Typically, g.729 will consume about 40k per session and g.723 will be as low 23k. Most devices will offer g.729 along with g.711 and some offer all three – there are others (g.726,
g.728 etc) but you will only have know about those if you are going to study for your CCNA. Between g.723 and g.729, I personally prefer the latter – the sound quality is very close to g.711 and is great if you have a half way decent Internet connection and do not need to run a fax or
modem across it. G.723 sounds a little thin  to me and is usually accompanied with low volume. This is the compression that people commonly experience when they are calling support centers in India or the Philippines.
About 5 years ago, codecs like g.729 and g.723 use to add about 20ms to 30ms of latency which would have been a problem if your latency was already high – say 110ms or higher. Now it is almost a non issue with some of the Cisco routers that most of the carriers are using to offer
Figure 3 shows an elusive hardware configuration issue. The packets were doing well up to the last hop which was the customer s gateway where they were experiencing 10% packet loss at the point of the sample (next to the red triangle on the time line). The hardware was fine but as it turned out, the router had it s clock source set to itself instead of getting it from the ISP. As you can see, once the problem was corrected, the packet loss stopped altogether.
convergent solutions of voice (with built in QOS), data and MPLS. Symptoms of a bandwidth problem, unfortunately, are reported to the IT department the same as latency problems (which are of caused by bandwith problems) and from my field experience, they usually are. As the users will put it – It does not work …  and they won t care what the cause is and usually won t try to distinguish. The voice will be choppy, unintelligible or simply gone. Low volume is usually not a bandwidth issue but more often the result of the type of compression being used or configured. In the case of bandwidth saturation, as illustrated in Figure 2, your voice packets are likely to get delayed or lost altogether unless they are prioritized. If the saturation is really severe, you will probably experience poor voice quality even with QOS. In such cases, get more bandwidth!

postheadericon Are You VoIP Ready?

If you are considering VoIP for your company – or more to the point, if YOU are responsible for the implementation of VoIP for your company – here are the basics you will need to understand.

Voice over IP performance is a function of:

a) Bandwidth

b) Latency

Both components have to be within a certain tolerance  in order to be usable and there are a multitude of factors that can adversely affect either. Many IT professionals who are first time VoIP-ers  often think that given enough bandwidth, you can do anything – including voice. This is only half true. A lack of bandwidth can cause latency, but an abundance of bandwidth is not a guarantee of a clear conversation. Of the two, latency is not only a show stopper, it is also the hardest one to find and correct because there are many causes that are often times not within your control. Figure 1 is a great example of a company with a bonded 6M T-1 connection with such erratic latency – and jitter  – that it is virtually unusable for

voice applications. In many cases like this, the most natural scapegoat is the carrier, however, in this case, hop 14 is the WAN side of the customer s router and hop 15 is a device behind it which implies a problem with the hardware. Here, the problem turned out to be one of the 4 T-1′s were in a permanent Admin Down  mode and needed to have the WIC replaced.

WARNING TO THOSE JUST GETTING INTO VoIP – Voice is real time  – using primarily UDP packets – and therefore much more affected by things like dropped packets, jitter and high latency than regular  data. If there are issues on your Internet connections with these items, you will probably not realize it unless you specifically – and continuously – test for it. Most carriers only guarantee bandwidth and NOT latency. Trust me, the CEO, CFO, Customer Service manager and Sales manager will get upset if the email server or Internet is a little slow, but they will absolutely FREAK if their phones don’t work.