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		<title>What QoS Looks Like</title>
		<link>http://www.ronek.com/blog/2011/01/what-qos-looks-like/</link>
		<comments>http://www.ronek.com/blog/2011/01/what-qos-looks-like/#comments</comments>
		<pubDate>Mon, 03 Jan 2011 17:05:56 +0000</pubDate>
		<dc:creator>eknaus</dc:creator>
				<category><![CDATA[Uncategorized]]></category>
		<category><![CDATA[ARP]]></category>
		<category><![CDATA[Assured FOrwarding]]></category>
		<category><![CDATA[Best Effort]]></category>
		<category><![CDATA[Differentiated Service Code Point]]></category>
		<category><![CDATA[Diffserv]]></category>
		<category><![CDATA[DSCP]]></category>
		<category><![CDATA[ECN]]></category>
		<category><![CDATA[Expedited Forwarding]]></category>
		<category><![CDATA[IP]]></category>
		<category><![CDATA[marking]]></category>
		<category><![CDATA[marking packets]]></category>
		<category><![CDATA[Pathview Cloud]]></category>
		<category><![CDATA[Pathview Premise]]></category>
		<category><![CDATA[QoS]]></category>
		<category><![CDATA[RTCP]]></category>
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		<category><![CDATA[SIP]]></category>
		<category><![CDATA[ToS]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[Wireshark]]></category>

		<guid isPermaLink="false">http://www.ronek.com/?p=372</guid>
		<description><![CDATA[How to use Wireshark and Pathview to verify QoS Performance Quality of Service, aka QoS, is one of those industry specific buzz words that everyone in the IT and telecom world is at least conceptually aware of.  When pressed for the details of the nuts and bolts, though, there is often an indignant “well, don’t [...]]]></description>
			<content:encoded><![CDATA[<p><strong><em>How to use <a href="http://www.wireshark.org" target="_blank">Wireshark</a> and <a href="http://www.apparentnetworks.com">Pathview</a> </em></strong></p>
<p><strong><em>to verify QoS Performance</em></strong></p>
<p>Quality of Service, aka QoS, is one of those industry specific buzz words that everyone in the IT and telecom world is at least conceptually aware of.  When pressed for the details of the nuts and bolts, though, there is often an indignant “well, don’t <strong><em>you</em></strong> know?”  response followed by immediate loss of eye contact and shrug of the shoulders.  In the Precambrian VoIP era (roughly 2001 and earlier), many of us were migrating from traditional “digital” voice technology and were sometimes asked how we could prioritize voice traffic over any other network traffic to which we would respond “Why, QoS it of course!” .</p>
<p>“Great!  Then bring your QoS thingy  out and let’s get these IP phones working …” was usually the customer’s response.  But as soon as the door was closed or the phone hung up, there would be that soul searching, gut wrenching moment of “why did I commit to that?” followed by “Who do I know that can do this?”.   In the beginning, for many of us, QoS, like the earth in Genesis  “was without form and void, and darkness was over the face of the deep.”</p>
<p><strong><em>IP Packet Cliff Notes </em></strong></p>
<p>Many IT guys know &#8211; and many more voice guys <strong><em>should</em></strong> know &#8211; that when you are capturing packets – usually with Wireshark – what you are seeing is a combination of traffic that is LAN specific and traffic that is on its way to another network.  Most of it is “IP” and some of it is not.  Non-IP packets, for example, would be ARPs that you see in the capture saying “Who has 192.168.1.1?  Tell 192.168.1.104” – commonly known as a ‘request’.  These sort of requests are all over the place in a busy network and you will see associated with them the replies  like “192.168.1.1 is at 00:23:ae:63:21:aa” (a MAC address) .  You will also see  the occasional “Gratuitous ARP” which is basically some host making sure that there is not another host on the network that has the same IP address it was just assigned (either manually or by a DHCP server).   What makes ARP a non-IP packet is that it does not have an IP field in its header and is therefore un-routable  off the local network.</p>
<div id="attachment_375" class="wp-caption aligncenter" style="width: 642px"><a href="http://www.ronek.com/files/2011/01/ARP_Packet_11-26-2010-9-00-15-PM.jpg" rel="lightbox[372]"><img class="size-large wp-image-375" src="http://www.ronek.com/files/2011/01/ARP_Packet_11-26-2010-9-00-15-PM-1024x341.jpg" alt="" width="632" height="341" /></a><p class="wp-caption-text">Typical ARP Packet</p></div>
<p>Comparing that with a typical TCP packet, you will see an Internet Protocol field along with all the goodies that come with it – including the holy grail (at least for this article) where you can actually tell whether the packet is given preferential treatment or not – the ToS (for Type of Service) field aka the  Differentiated Services Field.</p>
<p><a href="../files/2011/01/TCP_Packet_11-27-2010-8-24-36-AM.jpg" rel="lightbox[372]"><img class="alignright" src="../files/2011/01/TCP_Packet_11-27-2010-8-24-36-AM-1024x378.jpg" alt="" width="496" height="378" /></a></p>
<p>There are, or better, there <strong><em>were</em></strong> two main flavors of ToS – IP Precedence and DSCP.  IP Precedence is the older of the two and consists of the first 3 “most significant” bits in the ToS field (i.e. reading the bits  from left to right).  From that, the router could determine what  ‘precedence’ this particular packet had (if any) in terms of how it got routed e.g. route it using a particular interface or use the path with the best known metric, etc. and not necessarily whether it got to go in front of the line or not.   Hence the term “IP Precedence” and not “IP Preference” .   When you see this, think DSCP as in just about every instance nowadays this term and method has superseded it.</p>
<p>A common QoS misconception that is shared by most non-packet oriented office members (think sales people who have just heard of the term and want to show off at the next sales meeting) is that you want to prioritize everything from a particular IP address.  On the surface that might seem okay until you realize the shear number of protocols, applications and chatter that just one semi-busy machine generates on the network.  There are times when you do this but not many.   Don’t be THAT GUY in the meeting with an owner/CEO and his IT staff that refers to QoS configuration in this manner – you will lose the job the minute they are able to talk about you behind your back.  And don’t pronounce it as “Kwoss” either, it’s Q-O-S.  Saying it any other way makes you sound like a hick telephone guy or a salesman trying to sound like he knows what he’s talking about.  It must be said smoothly, confidently and with a certain air of  “I know what this means, do you?”</p>
<p><a href="http://www.ronek.com/files/2011/01/UX5000_84-10_12-9-2010-6-17-09-PM.jpg" rel="lightbox[372]"><img class="size-full wp-image-383 alignleft" src="http://www.ronek.com/files/2011/01/UX5000_84-10_12-9-2010-6-17-09-PM.jpg" alt="" width="438" height="240" /></a></p>
<p>Most QoS is a combination of “marking” packets of a particular protocol at the source level in combination with a matching QoS outbound map/policy/rule  at the router level and/or the managed path level in the case of MPLS networks.  You <strong><em>can</em></strong> prioritize packets solely based on ports, IP addresses even entire subnets but that is done at the router level only – usually one with a lot of horse power too &#8211; and frankly, it is less common.</p>
<p>DSCP (aka “DS”, “DiffServ”) stands for Differentiated Service Code Point &#8211; not that anyone actually says that unless they are trying to impress you &#8211;  and it comes in several decimal, binary or hexadecimal  values.   In Wireshark, the DS field itself is the second byte in the IPv4 header packet (which you will still see as the ToS field in some text books) and consists of  two sections –<a href="http://www.ronek.com/files/2011/01/DSCP_Binary_12-4-2010-3-03-31-PM.jpg" rel="lightbox[372]"><img class="alignright size-full wp-image-377" src="http://www.ronek.com/files/2011/01/DSCP_Binary_12-4-2010-3-03-31-PM.jpg" alt="" width="242" height="103" /></a></p>
<ol>
<li>The      actual DS binary value which occupies the first six most significant bits .</li>
<li>The      ECN value (aka Explicit Congestion Notification) which is used to notify      end points that a router along the path is reaching its saturation point      and to “lighten up” until it has a chance to work through its issue.  This is a little used feature and most      of the time you will see it set to zero.</li>
</ol>
<p>Technically, you can have up to 64 (0 through 63, that’s 2^6) values or “classes” but I’ve never seen a small or medium size business that needed more than 5.  For most scenarios, you will be using one of the following 3 classes;</p>
<p>1)      <strong>Best Effort</strong> – which is like sending a message in a bottle hoping that it gets to the other side … eventually.  Most sites we have walked into that are not using VoIP usually are using this. The numeric value is always zero.</p>
<p>2)      <strong>Assured Forwarding</strong> – This has decimal values between 10 and 38 and is a step or two above just throwing your packets into the Internet ocean.   It’s not FedEx but it’s not carrier pigeon either if programmed right.   In most cases you will use this to prioritize different services  such as video, a Point of Sale application, security , utilities, email etc.  I will sometimes put just the signaling  packets of VoIP in this class with a decimal value of 26 or 28.  In other cases you will use it to control access to a shared circuit – e.g. you may have 2 or more tenants using the same Internet T-1 and you want to control their percentage of usage equally.</p>
<p><strong><em>3) </em></strong><strong>Expedited Forwarding</strong> -  When it comes to the RTP voice packets, this is the only class you want because it is the one that gets your information there ahead of everyone else.  On top of that, you will usually reserve some bandwidth, to the determent of all other packets, when it absolutely positively has to get there …. now.  Its decimal value is 46. <strong><em>Special note &#8211; if your application requires either hex or binary and you don’t do conversions well in your head, get an app for your iPhone or Droid that does this – their cheap and often free.</em></strong></p>
<div id="attachment_379" class="wp-caption aligncenter" style="width: 615px"><strong><em><strong><em><a href="http://www.ronek.com/files/2011/01/DSCP_EFa_12-9-2010-5-29-56-PM.jpg" rel="lightbox[372]"><img class="size-large wp-image-379" src="http://www.ronek.com/files/2011/01/DSCP_EFa_12-9-2010-5-29-56-PM-1024x252.jpg" alt="" width="605" height="252" /></a></em></strong></em></strong><p class="wp-caption-text">QoS from Wiresharks&#039; point of view</p></div>
<p><strong><em> </em></strong></p>
<p>The actual decimal value in the Differentiated Services field that you see in Wireshark will not seem to match what you thought you programmed until you look at it in the packet details pane.  The reason for this is because there is an “off set”  which, in this case, allows you to address two semi-separate needs (DSCP and ECN) in one byte – that happens sometimes in packets btw.</p>
<p><strong>Getting from Here to There in Style …. And Knowing It.</strong></p>
<p>Keep in mind that the voice packets are usually going to be “marked” at the end points by what ever application is creating the data traffic – in this case the  “VoIP” app.  These points include the actual IP-PBX or Voice Server, the physical IP set or even the “softphone”.  Routers within an MPLS cloud will sometimes alter the marking by either changing them to “Best Effort” &#8211; which defeats our purpose – or some other value either intentionally or inadvertently.   Also, QoS of any sort only kicks in when the <strong><span style="text-decoration: underline">outgoing</span></strong> bandwidth from your WAN is saturated.</p>
<div id="attachment_380" class="wp-caption alignright" style="width: 485px"><a href="http://www.ronek.com/files/2011/01/High_Utilization_12-31-2010-3-51-59-PM.jpg" rel="lightbox[372]"><img class="size-full wp-image-380" src="http://www.ronek.com/files/2011/01/High_Utilization_12-31-2010-3-51-59-PM.jpg" alt="" width="475" height="287" /></a><p class="wp-caption-text">Customer &quot;The Internet is slow sometimes but that should not affect anything you&#039;re doing, right?&quot;  IT Dept - &quot;We have everything locked down pretty tight.  I doubt we&#039;re using more than 1 meg.&quot;</p></div>
<p>From a VoIP standpoint, there are effectively 2 protocol groups you care about;</p>
<p>1)      SIP, SDP, H.323, RTSP and H225 (there are more but these are the most used ones) which controls the non-voice part of a conversation such as the set up and teardown of  calls (as a voice guy, think ‘signaling’) and almost anything that is not actually carrying packets with audio in it.  This does not HAVE to have the top priority as it is not nearly as time sensitive as the second group but in cases when network traffic is really congested, you should give it some type of priority.</p>
<p>2)      RTP and RTCP   which are referred to as the “Transport”.  RTP is the protocol that actually has voice audio in it and you will want this to get the highest priority possible.  RTCP is the control link for a particular RTP session.  It has a few flavors and is another form of signaling but you want to keep this with the same QoS treatment as RTP.</p>
<p>A typical question you will hear when the MPLS carrier or generic router guy starts setting up QoS for you is  “How are you marking your packets?”.  To which you would inform him that the packets will be marked such and such to which they will usually say “No problem” and then disappear with nary a call or email.  So the question becomes a) Did they do it? and b) How do I verify that they did it?  After all, it’s you and “that phone system your company sold us” that will be held accountable.  This is where two important rules come into play with regard to doing anything on someone’s network the first time around.</p>
<p><strong>First Rule</strong> – Get a baseline of the health and use of their circuit(s) BEFORE you start tinkering.  This does not necessarily mean capturing packets with Wireshark (at least, not at first) as you are not so interested in the details of WHAT is going through (web traffic, emails, hosted services etc.) as much as you are interested in HOW it is going through.  Are the circuits you are working with clean and by that I mean is there any continuity loss?  High utilization? Jitter? High latency? Packet loss of  either voice or data packets – i.e. anything that will affect a real time application like voice..  Fyi,  normally data packets have larger payloads than voice packets and are sometimes treated differently as they travel across the network.   Programming QoS on a circuit that is not fit for the doghouse (even if you’re not in it at the time) will be a big waste of unbillable time.<a href="http://www.ronek.com/files/2011/01/W_O_QoS_12-31-2010-3-50-03-PM.jpg" rel="lightbox[372]"><img class="size-full wp-image-385 alignleft" src="http://www.ronek.com/files/2011/01/W_O_QoS_12-31-2010-3-50-03-PM.jpg" alt="" width="428" height="345" /></a></p>
<p><strong>Second Rule</strong> – Do not rely on the customer, their IT department or the carrier to give you anything close to accuracy or the truth as to the condition of their circuit(s).  They are usually clueless, misinformed or indifferent as to its importance.  Invest in a piece of software that can reliably measure all of the above.    The key word here is “reliably” because there is plenty of software (free and not free) that will claim to do it but  with some caveats.  If your reputation is being the “go to” guy,  don’t morph into the “go too cheap” guy.   If the app you’re using misdiagnoses, comes up with lots of false positives or simply misses the problem altogether, get something else … quick!    I am partial to Pathview (developed by <a title="Link to Apparent Networks" href="http://www.apparentnetworks.com" target="_blank">Apparent Networks</a>, btw we&#8217;re partners with them and can hook you up!) for that reason.  With this type of software, you can keep everyone honest – you, the customer and most importantly, the carrier/ISP.</p>
<p>Nothing gets you off the hook faster than being able to prove that a customer’s network or (more likely) Internet connection was screwed up BEFORE you walked into the building than a nice colorful graph or set of tests taken ahead of time.  I have run into situations where the IT department <strong><em>knew</em></strong> that the network was messed up but wanted to keep it that way just enough to allow them to increase their budget for more “ IT things” (not Pathview or anything like it by the way).   It was one of those “silent battles” that occasionally come up – just go in knowing that sometimes not everyone wants you to fix the problem.<a href="http://www.ronek.com/files/2011/01/W_QoS_12-31-2010-3-50-34-PM.jpg" rel="lightbox[372]"><img class="alignright size-full wp-image-374" src="http://www.ronek.com/files/2011/01/W_QoS_12-31-2010-3-50-34-PM.jpg" alt="" width="371" height="387" /></a></p>
<p>Once both Rules have been followed, you will then have something to go to work with – and eventually pick up your check.    What you should see is things like jitter, MOS, packet loss and latency drop for those packets that are prioritized (see the graphs showing QoS enabled and not enabled).  QoS is not perfect but in critical situations – when done right – it’s pretty good.</p>
<p><strong>Some Final Pointers</strong></p>
<p>Implementing QoS is like any other network project.  Get the big picture first with baselines because it will definitely help color in the details later.  Links to remote sites – especially end users working at their homes as more and more are doing nowadays – are almost always the trouble points so use an app that is good for that and does not take all your hard drive.<a href="../files/2011/01/Bad_MOS_12-31-2010-4-42-30-PM.jpg" rel="lightbox[372]"><img class="alignleft" src="../files/2011/01/Bad_MOS_12-31-2010-4-42-30-PM.jpg" alt="" width="672" height="274" /></a></p>
<p>Going in with a packet sniffer like Wireshark first – unless you are really familiar with the site or the problem – is arduous and may even distract you from the primary objective.  Besides, packet analysis is intense and time consuming &#8211; if you have that skill you are entitled to get top dollar but do it efficiently.   If you don’t have that skill and want it, check out <a href="http://www.wiresharkbook.com/" target="_blank"><strong><em>Wireshark Network Analysis</em></strong></a> by <a href="http://www.chappellu.com" target="_blank">Laura Chappell</a> – you will be a much sought after, well informed (and well compensated) network geek when you are through.</p>
<p style="text-align: left">If you are consulting or selling managed services,  it is important to straighten out your sales staff that contrary to what they read in the “IT Sales Action Hero” comic book, QoS is not magic and can not traverse into a remote network to prioritize THEIR data traffic.<a href="http://www.ronek.com/files/2011/01/QoS_SQR_1-3-2011-12-20-23-AM.jpg" rel="lightbox[372]"><br />
</a></p>
<p>Finally, be sure to let your customer and sales staff know that in the event that the entire circuit is down, nothing will get through – not even QoS’ed packets.  As obvious as this sounds, I actually had someone ask me that.  It’s hard to make this stuff up …..</p>
<p><a href="../files/2011/01/QoS_SQR_1-3-2011-12-20-23-AM.jpg" rel="lightbox[372]"><img src="../files/2011/01/QoS_SQR_1-3-2011-12-20-23-AM-1024x147.jpg" alt="" width="631" height="147" /></a></p>
<p>Copyright Eric Knaus, WCNA  Jan 2011</p>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="wireshark dscp">wireshark dscp</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="arp packet example">arp packet example</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="wireshark ip precedence">wireshark ip precedence</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="nec aspire qos">nec aspire qos</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="arp packet format">arp packet format</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="dscp and its function ?? in wireshark">dscp and its function ?? in wireshark</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="dscp voip wireshark">dscp voip wireshark</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="DiffServ Codepoint field in IP packet">DiffServ Codepoint field in IP packet</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="differentiated services code point 2011">differentiated services code point 2011</a></li><li><a href="http://www.ronek.com/blog/2011/01/what-qos-looks-like/" title="dscp values in hex">dscp values in hex</a></li></ul>]]></content:encoded>
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		<title>Remember the 3 D&#8217;s when dealing with ISP&#8217;s</title>
		<link>http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/</link>
		<comments>http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/#comments</comments>
		<pubDate>Thu, 02 Sep 2010 23:52:08 +0000</pubDate>
		<dc:creator>eknaus</dc:creator>
				<category><![CDATA[Uncategorized]]></category>
		<category><![CDATA[bandwidth]]></category>
		<category><![CDATA[carrier]]></category>
		<category><![CDATA[circuits]]></category>
		<category><![CDATA[connection]]></category>
		<category><![CDATA[data]]></category>
		<category><![CDATA[internet]]></category>
		<category><![CDATA[ISP]]></category>
		<category><![CDATA[network]]></category>
		<category><![CDATA[packets]]></category>
		<category><![CDATA[routers]]></category>
		<category><![CDATA[SIP]]></category>
		<category><![CDATA[T-1]]></category>
		<category><![CDATA[voice]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[voip ready]]></category>

		<guid isPermaLink="false">http://www.ronek.com/?p=319</guid>
		<description><![CDATA[It’s no secret that I am a big fan of Pathview (both Premise &#38; 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 [...]]]></description>
			<content:encoded><![CDATA[<p>It’s no secret that I am a big fan of Pathview (both Premise &amp; 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’ – <strong><em>Mean Time to Innocence</em></strong>.  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!</p>
<p>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.</p>
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_1_8-26-2010-9-23-23-AM.png" rel="lightbox[319]"><img class="size-large wp-image-321 " title="Fig_1_8-26-2010 9-23-23 AM" src="http://www.ronek.com/files/2010/09/Fig_1_8-26-2010-9-23-23-AM-1024x259.png" alt="" width="608" height="259" /></a></dt>
<dd><span style="color: #c0c0c0"><strong>Figure 1. <em>Customer reported that the Internet seemed to suddenly slow to a crawl.  ISP response -</em> &#8220;We are testing good to the modem at 3Megs.  Contact your IT person.  It’s obviously your equipment.&#8221;  <em>After hours of arguing with them (Comcast), it turned out to be their faulty modem.</em></strong></span></dd>
</dl>
</h3>
<p style="text-align: center">
<h4><strong>True story</strong></h4>
<p>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 <a title="Chappell Seminars" href="http://www.chappellseminars.com" target="_self">Laura Chappell</a> and her very thorough book<a title="Wiireshark Network Analysis" href="http://www.wiresharkbook.com/" target="_self"> <strong><em>“Wireshark Network Analysis”</em></strong> </a>released last March.  In one of her case studies was a very amusing yet oh-so-true true situation.</p>
<p>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;</p>
<p><em>“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.”</em></p>
<p>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.”</p>
<p>Here’s the fact, ISP’s have for years held &#8211; 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.</p>
<h4><strong>Baselining – Getting Prepared for the 3 D’s </strong></h4>
<p>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.</p>
<p><strong><em>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.</em></strong></p>
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_2_8-26-2010-5-36-46-PM.png" rel="lightbox[319]"><img class="size-full wp-image-322 " title="Fig_2_8-26-2010 5-36-46 PM" src="http://www.ronek.com/files/2010/09/Fig_2_8-26-2010-5-36-46-PM.png" alt="" width="646" height="554" /></a></dt>
<dd><strong>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.</strong></dd>
</dl>
</h3>
<p style="text-align: center">
<p>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.</p>
<h4><strong><span style="text-decoration: underline">DENY</span></strong></h4>
<p>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:</p>
<p><strong><em>Axiom 1</em></strong> &#8211; The less technical they are, the more they will want to sell you additional bandwidth.</p>
<p style="padding-left: 90px">Slow Internet? More bandwidth.</p>
<p style="padding-left: 90px">Getting cut off altogether?  More bandwidth.</p>
<p style="padding-left: 90px">PC making strange noises? More bandwidth.</p>
<p style="padding-left: 90px">Think some of your ports are being blocked?  Get more bandwidth and, oh by the way,</p>
<p style="padding-left: 90px">there may be an extra charge for unblocked ports (whatever they are).</p>
<p><strong><em>Axiom 2</em></strong> – The more technical they are, the likelier they are to put all the blame on the user’s equipment.</p>
<p>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.</p>
<p>“How does it look now?”  I asked after about a minute.</p>
<p>“Looks the same, no packet loss.”<br />
No surprise, he was looking at the wrong circuit all along.  That one was easy.</p>
<p>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,</p>
<p><strong><em>“The truth is out there … somewhere, but you will need Pathview to find it.”</em></strong></p>
<p>(I should probably copyright that before Jim grabs it!).</p>
<p>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 …</p>
<p style="text-align: center">
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_3_8-26-2010-8-55-49-AM.png" rel="lightbox[319]"><img class="size-large wp-image-323  " title="Fig_3_8-26-2010 8-55-49 AM" src="http://www.ronek.com/files/2010/09/Fig_3_8-26-2010-8-55-49-AM-1024x199.png" alt="" width="660" height="199" /></a></dt>
<dd>Figure 3. Using Pathview Cloud looking towards the SIP provider</dd>
</dl>
</h3>
<p>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.</p>
<p>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.</p>
<h4><strong><span style="text-decoration: underline">DEFLECT </span></strong></h4>
<p>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 –</p>
<p>“How much packet loss? I will need to know the exact percentage and where it is occurring”</p>
<p>“When did this start?”</p>
<p>“Why are you just noticing it now?”</p>
<p>“How come we’re not seeing it?”</p>
<p>“Are you sure you have power?”</p>
<p>“Have you replaced your equipment?”</p>
<p>“Is it raining there?” ç That was an interesting one!</p>
<p>“We have VoIP sets and we’re not having problems.”</p>
<p>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?”.</p>
<p>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.</p>
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_4_8-26-2010-7-56-05-AM.png" rel="lightbox[319]"><img class="size-full wp-image-324" title="Fig_4_8-26-2010 7-56-05 AM" src="http://www.ronek.com/files/2010/09/Fig_4_8-26-2010-7-56-05-AM.png" alt="" width="643" height="456" /></a></dt>
<dd>Figure 4. &#8220;Okay, should be working fine now.  The tech ran a few DSL Reports tests and then a traceroute for good measure.  Looks great!&#8221;  <strong><em>Are you kidding me?!</em></strong></dd>
</dl>
</h3>
<p>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”.</p>
<h4><strong><span style="text-decoration: underline">DELAY</span></strong></h4>
<p>The response was that they did not see any problems – which also falls under “DENY” &#8211; 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.</p>
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_5_8-26-2010-3-53-17-PM.png" rel="lightbox[319]"><img class="size-full wp-image-325" title="Fig_5_8-26-2010 3-53-17 PM" src="http://www.ronek.com/files/2010/09/Fig_5_8-26-2010-3-53-17-PM.png" alt="" width="673" height="286" /></a></dt>
<dd>Figure 5. &#8220;We don&#8217;t measure MOS  in this department but that looks normal.&#8221;  <em>Is there someone I can go over this with?  Hello? … Hello?</em></dd>
</dl>
</h3>
<p>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.</p>
<h3 class="mceTemp mceIEcenter">
<dl>
<dt><a href="http://www.ronek.com/files/2010/09/Fig_6_8-26-2010-3-02-58-PM.png" rel="lightbox[319]"><img class="size-large wp-image-320" title="Fig_6_8-26-2010 3-02-58 PM" src="http://www.ronek.com/files/2010/09/Fig_6_8-26-2010-3-02-58-PM-1024x522.png" alt="" width="652" height="522" /></a></dt>
<dd><strong>Figure 6. &#8220;We&#8217;re not seeing anything but go ahead and send us your graphs and we&#8217;ll forward them up the line.&#8221;  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.”</strong></dd>
</dl>
</h3>
<p>But that was when the playing field consisted of AT&amp;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.</p>
<h4><strong><span style="text-decoration: underline">Parting Shots</span></strong></h4>
<p>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 ;</p>
<p>1)      Don’t look at packets they way they are actually used</p>
<p>2)      Don’t want your help when trouble shooting</p>
<p>3)      Often don’t have the software  tools, training or even the inclination to  look beyond the single port of a router.</p>
<p>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.</p>
<p style="text-align: center">
<p style="text-align: center">Copyright Eric Knaus 2010</p>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="What is the mean time to send the 4 packets to the above web site?">What is the mean time to send the 4 packets to the above web site?</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="what are the port requirements for appneta appliances to connect back to the vox pathview cloud servers?">what are the port requirements for appneta appliances to connect back to the vox pathview cloud servers?</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="voip ready check appliance">voip ready check appliance</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="voice Graph">voice Graph</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="troubleshoot website blocking Paetec">troubleshoot website blocking Paetec</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="paetec communciations port blocking">paetec communciations port blocking</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="nec ux5000 choppy voice">nec ux5000 choppy voice</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="nec ux5000 calls drop every 15 minutes">nec ux5000 calls drop every 15 minutes</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="nec ux phone server garbled speech">nec ux phone server garbled speech</a></li><li><a href="http://www.ronek.com/blog/2010/09/remember-3-ds-dealing-isps/" title="nec phones voip ready">nec phones voip ready</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready? &#8211; Glossary</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/#comments</comments>
		<pubDate>Tue, 11 May 2010 03:22:38 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[Blog]]></category>
		<category><![CDATA[bandwidth]]></category>
		<category><![CDATA[bandwidth saturation]]></category>
		<category><![CDATA[content filtering]]></category>
		<category><![CDATA[dropped packets]]></category>
		<category><![CDATA[hop]]></category>
		<category><![CDATA[ICMP]]></category>
		<category><![CDATA[ISP]]></category>
		<category><![CDATA[jitter]]></category>
		<category><![CDATA[LAN]]></category>
		<category><![CDATA[latency]]></category>
		<category><![CDATA[NIU]]></category>
		<category><![CDATA[point-to-point]]></category>
		<category><![CDATA[PSTN]]></category>
		<category><![CDATA[QoS]]></category>
		<category><![CDATA[SIP]]></category>
		<category><![CDATA[TDM]]></category>
		<category><![CDATA[VPN]]></category>
		<category><![CDATA[WIC]]></category>

		<guid isPermaLink="false">http://cms.ronek.com/?p=265</guid>
		<description><![CDATA[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 &#8211; and the [...]]]></description>
			<content:encoded><![CDATA[<table border="0" cellspacing="0" cellpadding="0" width="500">
<tbody>
<tr>
<td>Bandwidth Saturation</td>
<td rowspan="2">The point in which all available bandwidth on an Internet</p>
<p>connection is used up.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Bandwidth</td>
<td rowspan="2">The amount of data passing through a connection over a given time.<br />
It is usually measured in bps (bits-per-second) or Mbps (Megabits<br />
per Second). As a general rule, get as much as you can afford &#8211; and<br />
the make sure you are getting it.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Content Filtering</td>
<td rowspan="2">On the Internet, content filtering (also known as information</p>
<p>filtering) is the use of a program to screen and exclude from access</p>
<p>or availability Web pages or e-mail that is deemed objectionable.</p>
<p>Content filtering is used by corporations and governments as part of</p>
<p>Internet firewall computers and also by home computer owners,</p>
<p>especially by parents to screen the content their children have access</p>
<p>to from a computer</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Dropped Packets</td>
<td rowspan="2">Packets (i.e. small data &#8220;packages&#8221;) are occasionally dropped, or</p>
<p>lost, on the network for various reasons. For instance, two nodes</p>
<p>may be communicating at widely disparate transfer rates. TCP</p>
<p>packets are resent, UDP s are not.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Hop</td>
<td rowspan="2">In a packet-switching network, a hop is the trip a data packet takes</p>
<p>from one router or intermediate point to another in the network. On</p>
<p>the Internet (or a network that uses TCP/IP), the number of hops a</p>
<p>packet has taken toward its destination (called the &#8220;hop count&#8221;) is</p>
<p>kept in the packet header.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>ICMP</td>
<td rowspan="2">Internet Control Message Protocol is a message control and</p>
<p>error-reporting protocol between a host server and a gateway to the</p>
<p>Internet. ICMP uses Internet Protocol (IP) datagrams, but the</p>
<p>messages are processed by the IP software and are not directly</p>
<p>apparent to the application user.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>ISP</td>
<td rowspan="2">An ISP (Internet Service Provider) aka Carrier, aka Provider, is a</p>
<p>company that collects a monthly or yearly fee in exchange for</p>
<p>providing the subscriber with Internet access.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Jitter</td>
<td rowspan="2">The difference in latency from one packet to the next measured in</p>
<p>milliseconds.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>LAN</td>
<td rowspan="2">A Local Area Network is a computer network that spans a</p>
<p>relatively small area. Most LANs are confined to a single building or</p>
<p>group of buildings. However, one LAN can be connected to other</p>
<p>LANs over any distance via telephone lines and radio waves. A</p>
<p>system of LANs connected in this way is called a wide-area network</p>
<p>(WAN).</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Latency</td>
<td rowspan="2">In a network, latency, a synonym for delay, is an expression of how</p>
<p>much time it takes for a packet of data to get from one designated</p>
<p>point to another. Typically, latency is measured by sending a packet</p>
<p>that is returned to the sender. The round-trip time &#8211; measured in</p>
<p>milliseconds &#8211; is considered the latency.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>NIU</td>
<td rowspan="2">A Network Interface Unit (sometimes called a network interface</p>
<p>device) is a device that serves as a common interface for various</p>
<p>other devices within a LAN , or as an interface to allow networked</p>
<p>computers to connect to an outside network.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Ping</td>
<td rowspan="2">Loosely, ping means &#8220;to get the attention of&#8221; or &#8220;to check for the</p>
<p>presence of&#8221; another party online. Ping operates by sending a packet<br />
to a designated address and waiting for a response. The computer<br />
acronym (for Packet Internet or Inter-Network Groper) was<br />
contrived to match the submariners&#8217; term for the sound of a returned<br />
sonar pulse.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>Point to Point</td>
<td rowspan="2">Point-to-point telecommunications generally refers to a connection</p>
<p>restricted to two endpoints, usually host computers.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>PSTN</td>
<td rowspan="2">Public Switched Telephone Network is the world&#8217;s collection of</p>
<p>interconnected voice-oriented public telephone networks, both</p>
<p>commercial and government-owned</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>QoS</td>
<td rowspan="2">Quality of Service. Describes the ability of a e.g. router to prioritize</p>
<p>certain packets</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>SIP</td>
<td rowspan="2">Session Initiation Protocol is an application-layer control</p>
<p>(signaling) protocol for creating, modifying, and terminating</p>
<p>sessions with one or more participants. It can be used to create</p>
<p>two-party, multiparty, or multicast sessions that include Internet</p>
<p>telephone calls, multimedia distribution, and multimedia</p>
<p>conferences.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>TDM</td>
<td rowspan="2">Short for Time Division Multiplexing, a type of multiplexing that</p>
<p>combines data streams by assigning each stream a different time slot</p>
<p>in a set. TDM telephone sets (often referred to as digital  sets)</p>
<p>differ from IP sets in that they do not go on a LAN s infrastucture</p>
<p>are compatible with analogue wiring schemes and can work on cable</p>
<p>runs oup to 1,600 feet.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>VPN</td>
<td rowspan="2">(pronounced as separate letters) Short for Virtual Private Network,</p>
<p>is a private network that uses a public network (usually the Internet)</p>
<p>to connect remote sites or users together. VPNs use &#8220;virtual&#8221;</p>
<p>connections routed through the Internet from a company&#8217;s private</p>
<p>network , a remote site or employee.</td>
</tr>
<tr>
<td></td>
</tr>
<tr>
<td>WIC</td>
<td rowspan="2">WAN Interface Card is installed in a router and is the component</p>
<p>that a Internet T-1 will physically plug in to.</td>
</tr>
<tr>
<td></td>
</tr>
</tbody>
</table>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="China Internet network Latency">China Internet network Latency</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="china internet latency">china internet latency</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="73ms jitter">73ms jitter</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="lan saturation">lan saturation</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="latency china internet">latency china internet</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="nec networking glossary">nec networking glossary</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="network saturation available bandwidth">network saturation available bandwidth</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="Router interface saturation">Router interface saturation</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="saturasi bandwidth">saturasi bandwidth</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-glossary/" title="site available bandwidth saturation">site available bandwidth saturation</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready? &#8211; The Road to China: Content Filtering to the Max</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:37:21 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://cms.ronek.com/?p=260</guid>
		<description><![CDATA[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, [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">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 &#8211; because it does not &#8211; 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</div>
<div id="_mcePaste">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 &#8211; 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 &#8211; you guessed it &#8211; 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 &#8211; the network just could not handle it!</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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,</div>
<div id="_mcePaste">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.</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/" title="internet latency china">internet latency china</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/" title="DATA COMMUNICATINS BUREAU">DATA COMMUNICATINS BUREAU</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/" title="max latency for voip to china">max latency for voip to china</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-road-to-china/" title="the target site is in a town just south of shanghai called hangzhou">the target site is in a town just south of shanghai called hangzhou</a></li></ul>]]></content:encoded>
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		<item>
		<title>Are You VoIP Ready? &#8211; What Stable Connections Look Like</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:28:28 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://cms.ronek.com/?p=258</guid>
		<description><![CDATA[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 [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">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.</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="how stable is VoIP">how stable is VoIP</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="get over ex">get over ex</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="how much bandwidth do you need for nec aspire">how much bandwidth do you need for nec aspire</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="is voip ready?">is voip ready?</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="is voip stable?">is voip stable?</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="voip ready cable connection">voip ready cable connection</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="what a stable looked like">what a stable looked like</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-what-stable-connections-look-like/" title="what stable look like">what stable look like</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready? &#8211; The &#8220;X&#8221; Factor</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:23:25 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://cms.ronek.com/?p=255</guid>
		<description><![CDATA[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 [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">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</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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.</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="how to connect chinaroy ip pbx">how to connect chinaroy ip pbx</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="nec ux basic tdm">nec ux basic tdm</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="nitsuko paging is choppy">nitsuko paging is choppy</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="TDM sets">TDM sets</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="voip and LAN wiring">voip and LAN wiring</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="will nec ux5000 and cisco IP phones talk with each other">will nec ux5000 and cisco IP phones talk with each other</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-the-x-factor/" title="xfactor volp communications">xfactor volp communications</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready? &#8211; QoS (Quality of Service)</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:16:50 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://cms.ronek.com/?p=252</guid>
		<description><![CDATA[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 [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">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</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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.</div>
<div></div>
<div>* More about SIP servers in coming articles.</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="voip ready check">voip ready check</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="voip ready router">voip ready router</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="nec voip qos">nec voip qos</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="nec qos">nec qos</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="UX5000 qos">UX5000 qos</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="ux5000 voip qos">ux5000 voip qos</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="QOS on Nec voip phones">QOS on Nec voip phones</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="qos nec voip">qos nec voip</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="@ronek com">@ronek com</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-qos-quality-of-service/" title="qos for NEC vOip phones">qos for NEC vOip phones</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready?  &#8211; Latency</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:10:12 +0000</pubDate>
		<dc:creator>admin</dc:creator>
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		<guid isPermaLink="false">http://cms.ronek.com/?p=249</guid>
		<description><![CDATA[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 [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">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 &#8211; or lack thereof &#8211; has to be extrapolated from that.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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 &#8211; if there is a problem, it will likely show up within that sample.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">(usually) but with different spacing. It is essentially the difference in latency time from one packet to the next. When jitter is high &#8211; anything over 15% variance between samples &#8211; it usually points to bandwidth problem. To get an idea of the impact of high jitter, imagine the sound of an</div>
<div id="_mcePaste">audio CD that is played while alternating between pause and fast-forward. The garbled sound is characteristic of jitter.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">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 &#8211; that</div>
<div id="_mcePaste">they supplied &#8211; 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  &#8211; i.e. it was referencing itself as a clock source &#8211; instead of</div>
<div id="_mcePaste">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.</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="voip no latency">voip no latency</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="latency to houston">latency to houston</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="voip latency">voip latency</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="audio samples of voip latency">audio samples of voip latency</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="sample voip project plan">sample voip project plan</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="ping too high for voip">ping too high for voip</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="ping times high for voip phones">ping times high for voip phones</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="ping from a NEC ip phone">ping from a NEC ip phone</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="ping communication voip">ping communication voip</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-latency/" title="ping communication enable voip">ping communication enable voip</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready? &#8211; Bandwidth</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/#comments</comments>
		<pubDate>Tue, 11 May 2010 01:03:11 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[Blog]]></category>
		<category><![CDATA[3M]]></category>
		<category><![CDATA[bandwidth]]></category>
		<category><![CDATA[cable]]></category>
		<category><![CDATA[CCNA]]></category>
		<category><![CDATA[circuits]]></category>
		<category><![CDATA[Cisco]]></category>
		<category><![CDATA[compression]]></category>
		<category><![CDATA[connection]]></category>
		<category><![CDATA[data]]></category>
		<category><![CDATA[device]]></category>
		<category><![CDATA[DSL]]></category>
		<category><![CDATA[g.117]]></category>
		<category><![CDATA[g.723]]></category>
		<category><![CDATA[g.729]]></category>
		<category><![CDATA[internet]]></category>
		<category><![CDATA[ISP]]></category>
		<category><![CDATA[latency]]></category>
		<category><![CDATA[modems]]></category>
		<category><![CDATA[MPLS]]></category>
		<category><![CDATA[packets]]></category>
		<category><![CDATA[PBX]]></category>
		<category><![CDATA[QoS]]></category>
		<category><![CDATA[routers]]></category>
		<category><![CDATA[sampling rate]]></category>
		<category><![CDATA[speed]]></category>
		<category><![CDATA[T-1]]></category>
		<category><![CDATA[upstream]]></category>
		<category><![CDATA[voice]]></category>
		<category><![CDATA[VoIP]]></category>
		<category><![CDATA[wires]]></category>

		<guid isPermaLink="false">http://cms.ronek.com/?p=246</guid>
		<description><![CDATA[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 [...]]]></description>
			<content:encoded><![CDATA[<div id="_mcePaste">Bandwidth -</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">the device is going to send over the wires to the far end.</div>
<div id="_mcePaste">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).</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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</div>
<div id="_mcePaste">inspection, their contract actually says something like &#8230; you may experience speeds UP TO 3M &#8230;  which usually translates into something like 700k down and maybe 256k up on average &#8211; if that.</div>
<div id="_mcePaste">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 &#8211; there are others (g.726,</div>
<div id="_mcePaste">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 &#8211; 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</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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 &#8211; 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</div>
<div id="_mcePaste">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.</div>
<div id="_mcePaste">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 &#8211; It does not work &#8230;  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!</div>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="3M bandwidth">3M bandwidth</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec voip codec">nec voip codec</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="low sound volume problem with g729">low sound volume problem with g729</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="is my network voip ready">is my network voip ready</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec aspire codec">nec aspire codec</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec aspire outbound quality">nec aspire outbound quality</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec aspire voip compression">nec aspire voip compression</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec voip choppy audio">nec voip choppy audio</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="nec aspire choppy audio">nec aspire choppy audio</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready-bandwidth/" title="paetec voip fax troubleshooting">paetec voip fax troubleshooting</a></li></ul>]]></content:encoded>
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		<title>Are You VoIP Ready?</title>
		<link>http://www.ronek.com/blog/2010/05/are-you-voip-ready/</link>
		<comments>http://www.ronek.com/blog/2010/05/are-you-voip-ready/#comments</comments>
		<pubDate>Tue, 11 May 2010 00:48:36 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[Blog]]></category>
		<category><![CDATA[bandwidth]]></category>
		<category><![CDATA[calls]]></category>
		<category><![CDATA[carrier]]></category>
		<category><![CDATA[data]]></category>
		<category><![CDATA[hop]]></category>
		<category><![CDATA[IP]]></category>
		<category><![CDATA[IT]]></category>
		<category><![CDATA[jitter]]></category>
		<category><![CDATA[latency]]></category>
		<category><![CDATA[packets]]></category>
		<category><![CDATA[phones]]></category>
		<category><![CDATA[real time]]></category>
		<category><![CDATA[router]]></category>
		<category><![CDATA[UDP packets]]></category>
		<category><![CDATA[voice]]></category>
		<category><![CDATA[voip ready]]></category>
		<category><![CDATA[WAN]]></category>

		<guid isPermaLink="false">http://cms.ronek.com/?p=242</guid>
		<description><![CDATA[If you are considering VoIP for your company &#8211; or more to the point, if YOU are responsible for the implementation of VoIP for your company &#8211; 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 [...]]]></description>
			<content:encoded><![CDATA[<p>If you are considering VoIP for your company &#8211; or more to the point, if YOU are responsible for the implementation of VoIP for your company &#8211; here are the basics you will need to understand.</p>
<p>Voice over IP performance is a function of:</p>
<p>a) Bandwidth</p>
<p>b) Latency</p>
<p>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 &#8211; 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 &#8211; and jitter  &#8211; that it is virtually unusable for</p>
<p>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&#8242;s were in a permanent Admin Down  mode and needed to have the WIC replaced.</p>
<p>WARNING TO THOSE JUST GETTING INTO VoIP &#8211; Voice is real time  &#8211; using primarily UDP packets &#8211; 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 &#8211; and continuously &#8211; 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&#8217;t work.</p>
<h4>Incoming search terms for the article:</h4><ul><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="voip latency tolerance">voip latency tolerance</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="are you ready for VoIP">are you ready for VoIP</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="are you voip ready">are you voip ready</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="a lack of bandwidth">a lack of bandwidth</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="voip jitter and latency tolerances">voip jitter and latency tolerances</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="tagging udp packets">tagging udp packets</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="nec ux5000 voip bandwith">nec ux5000 voip bandwith</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="nec ux5000 network slow">nec ux5000 network slow</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="nec aspire udp">nec aspire udp</a></li><li><a href="http://www.ronek.com/blog/2010/05/are-you-voip-ready/" title="nec aspire telephone voip">nec aspire telephone voip</a></li></ul>]]></content:encoded>
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