QoS

Posts Tagged ‘QoS’

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

milliseconds.

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

(WAN).

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

conferences.

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? – 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!