apuntes convergence+

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TELEPHONY Legacy Hybrid and IP Telephony Systems Telephone systems have begun the migration from sending voice across dedicated circuit switched lines using time division multiplexing to capturing voice and using IP packets to send the voice across a shared data network to the receiver. Hybrid telephony systems integrate key-systems and wide area connections to allow voice calls to move across traditional phone systems for local calls. If the call is destined for long distance, the voice is encapsulated and sent across the WAN link in packet form. The Internet Protocol (IP) is an OSI layer three protocol. This protocol uses an address to uniquely identify every host connect- ed directly to the Internet. The speaker’s voice is captured and segmented into separate blocks called packets. There are various reasons to migrate to an IP telephone system. First, the phone devices now have universal access. Wherever there is Internet access an IP phone may make calls to anywhere in the world. IP phones contain additional features not found on traditional phones. Cost reductions are made when companies migrate to a single infrastructure running both voice and data over the same wire. The number of technical experts required to operate a dual telephone/data structure will be reduced. Key Systems

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Page 1: Apuntes Convergence+

TELEPHONY

Legacy Hybrid and IP Telephony Systems

Telephone systems have begun the migration from sending voice across  dedicated  circuit switched  lines  using  time  division multiplexing to capturing voice and using IP packets to send the voice across a shared data network to the receiver.

Hybrid telephony systems integrate key-systems and wide area connections to allow voice calls to move across traditional phone systems for  local calls. If  the call  is destined for long distance, the  voice is  encapsulated  and  sent  across  the  WAN link in packet form.

The Internet Protocol (IP) is an OSI layer three protocol. This protocol uses an address to uniquely identify every host connect- ed directly to the Internet. The speaker’s voice is captured and segmented into separate blocks called packets.

There are various reasons to migrate to an IP telephone system. First, the phone devices now have universal access. Wherever there is Internet access an IP phone may make calls to anywhere in the world.

IP phones contain additional features not found on traditional phones. Cost reductions are made when companies migrate to a single infrastructure running both voice and data over the same wire. The number of technical experts required to operate a dual telephone/data structure will be reduced.

Key Systems

The smallest businesses usually begin with the same sort of single line telephone installed in most homes. However, as a business grows and adds staff, it needs the flexibility of multiple lines. For a business too big for a single telephone, but much too small for a large-scale office switching system, key systems are the answer.

Key systems are fairly simple on-site telephone systems geared to organizations with fewer than 100 telephones. Like a PBX, they switch  calls  to and  from  the public  network  and within users’ premises.  However,  key  systems  are  simpler  than  a  PBX, reducing the administrative workload for small businesses.

Key System Components

The first multiline business telephone system was called the 1A key  telephone  system. It consisted of  a  red hold button, four telephone line buttons, and an office intercom button.

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This system became  the workhorse of  small businesses, and many of these systems are still installed today.

A key system provides multiple telephone extensions access to a group of single telephone lines. For example, if a small office has six single lines, it can use a key system to access any of those lines from each of its telephones. Each telephone extension would have six buttons (one for each line); this is  known as a squared  line configuration. To connect a telephone extension to a line, a caller simply presses one of the unlit line buttons and if a line is free the caller will hear dial tone. The concept of key systems is illustrated on the Key Telephone System Diagram

Key Telephone System

All telephone sets in a key system were connected to a central device called a Key Service Unit (KSU), which connected each telephone set to a group of outside business lines. Today, new KSU-less systems offer all the functionality of KSU within each telephone set.

The main point to remember about a key system is that it can sup- port only as many incoming or outgoing telephone calls, or “call paths,” as there are lines installed. In other words, if a customer has  100  telephone  extensions  in  an  office, but  only  40  lines installed, the maximum number of simultaneous calls, coming in or going out, is limited to 40. If the 40 lines are all in use, outgoing callers must wait for a free line, while incoming callers receive a busy signal.

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In many business settings, such as large retail centers or factory floors, cordless key systems provide employees telephone service while allowing them freedom of movement. Wireless transmission is used to connect these mobile extensions to the main business lines, and to each other by means of intercom features. This type of technology is presented on the Cordless Key System Diagram

Cordless Key System

The use of wireless telephones inside buildings requires special base stations with antennae located on every floor. There are gen- erally also special outside base stations with antennae for nearby outdoor areas between buildings on a campus. The base stations must be wired with twisted pair to specialized circuit packs within the  telephone  system cabinet. Specialized  wireless telephones associated  with key  systems and  PBXs are  high profit margin peripherals. These telephones operate at higher frequencies than home telephones and have specialized features associated with particular key and PBX telephone systems.

On-site wireless telephone systems use a cellular digital switching technology  similar to Personal Communications Service (PCS). Calls are transferred between base stations when a user walks out of the range of a particular antenna. Some mobile telephone units can function both inside and outside of the business campus. They sense  when  they are out  of the range of  the base system, and automatically switch calls to a cellular telephone network.

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Limitations of Key Systems

A key system provides a cost-effective way for a small business to share a moderate number of telephone lines. However, key systems offer fairly unsophisticated functionality and features. In addition, their main advantage, simplicity, becomes a liability as a business adds telephone lines past a certain point.

As we have seen, each telephone in a key system has a button to access each telephone line, plus a hold button, CO telephone lines, and  intercom  lines. Therefore,  if  a  business needs  18 business lines, its telephone sets would have at least 20 buttons. By the time a business requires 20 or more lines, each telephone has become quite complex, hard to use, and expensive. Can you imagine an extension  telephone with 50 or more  buttons on it? Therefore, we can upgrade a true key system only so far before we need to try a different approach.

Hybrid Systems

With  the  integration  of  computer  technology  inside  telephone systems, key telephone systems became more and more advanced. Gradually, they began to include features previously found on only full  PBX  systems. Thus, the term  hybrid  describes  a telephone system that includes features of both a key system and PBX.

A characteristic of a hybrid key system is the grouping of outside trunks  into  pools,  by function  or  organization.  For  example, certain trunks are allocated to a particular department.

Electronic Key Telephone Systems (EKTS) often cross the line into the PBX world, providing switching capabilities, as well as impres- sive  functionality and  feature  content. EKTS is  a  key  telephone system in which electromechanical relays and switches have been replaced  by  electronic  devices, often  in  the  telephone sets and central  cabinet. The inner  workings of  the central  cabinet of an EKTS more resemble a computer than a conventional key system.

IP Telephony Systems

“IP Telephony (Internet Protocol)  is a  means  for  handling  your phone calls and faxes over the Internet as opposed to a traditional phone  line.  It  is  becoming  the  preferred technology  for  large organizations  because it  saves  money, is  easy  to  maintain, and produces a superior ROI compared to PBX systems.

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IP Telephony is less expensive to install than a PBX phone system. Less structured cabling is required. Once a drop is wired, it can be used for data and voice.

Maintenance  costs  are  lower  compared  to  a  PBX  system. Technicians trained in converged networks are able to maintain both  data and  voice  systems. In  fact, you  can handle routine maintenance in house, such as add-ons, moves and changes.

Monthly operating costs are less because the system does not require dedicated lease lines for voice and data.

Employees favor IP Telephony as it offers better tools to archive voice mail and keep track of faxes. In addition, faxes and voice mail  can  be retrieved off  the  Internet anywhere  in the world. Features are visible on your computer making it relatively simple to operate.

The system provides call accounting, giving you documentation on employee performance, carrier billing and call trends.

Call NIC for a total solution. We are a single source for all of your technology requirements. Services include consulting, design, project   management,  system installation,  maintenance,  and support for all network systems” http://www.nicweb.com/en/ services/network_systems/ip_telephony.html

Voice Transmission Fundamentals

When a caller lifts the handset off hook the PBX signals the router to seize a trunk. The PBX then forwards the dialed digits to the router. The router’s dial plan maps these digits to an IP address and initiates a call establish request to the remote router. The end nodes  are responsible, on a VoIP call, for call  connection and signaling.

The ITU-T Q.931 recommendation manages the call setup and teardown. Steps to initialize the call include:

SETUP of a connection,

CALL PROCESSING determines the remote terminal received the call,

ALERTING informs the calling party the remote terminal is ringing,

CONNECT tells the calling party the remote terminal is now off hook,

RELEASE COMPLETE happens when either end of the conversation hangs up the call

The call control sequence may use RTP – Real-Time Transport Protocol  in  conjunction with  RTCP – Real-Time  Transport Control Protocol to manage the audio and/or video

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streams as requested. RTP running atop UDP carries the voice and/or video stream. RTCP controls the connection and runs atop TCP for tracking and control.

H.245  control  signaling  negotiates  channel  usage  and  may negotiate agreement between all the endpoints of a conference call on technologies such as codecs, speeds, and parameters.

The call may also use various protocols to set a quality of service for the voice packets. Running on the transport layer is the RSVP – Resource Reservation Protocol for QoS. The packets may have quality of  service set  on  layer three with DSCP – Differentiated Service Code Points. And, on  layer  two  the  IEEE 802.1Q for quality of service/precedence levels for the frames themselves.

Encoding, Decoding, and Compression

It takes time to convert a voice signal from analog to digital. A similar delay, or latency, occurs when the remote end converts the digital signal back to analog.

Voice compression reduces the number of bits in a transmission by removing redundant character strings in the digital character stream. However, compression algorithms increase latency by requiring large voice stream samples before they can compress and packetize voice signals

Digitizing the Voice

As  the benefits of  a digital  telecommunications infrastructure became  apparent, it  was necessary  to take the analog voice and  convert  it  to a  digital  format. The purpose  for digitizing analog  voice  signals is  so  they  are  compatible with a digital telecommunications network.

Analog to Digital

The first part of the Analog-to-Digital Conversion Diagram (A) represents the original analog waveform. Part (B) represents digital pulses that control the sampling rate of the analog waveform. The digital pulses open a “gate” for the duration of their pulse widths, reading the amplitude of the analog waveform for this period of time. The sampled analog waveform appears as pulses (C) that are each correlated to a specific number (D). This number represents one sample of the voice signal. The binary representation of this number (E) is transmitted digitally across a circuit. At the receiving end, the reverse process takes place to convert the digital signal back to the original analog waveform

A typical sampling rate (number of times a byte is generated) is 8,000 times per second, approximately twice the bandwidth required for an analog voice signal. Research has determined that a sampling rate of at least two times the highest frequency compnent of the

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original signal results in accurate representation of the original intelligence. This is called the Nyquist Theorem.

A coder/decoder (codec) is the device that takes the analog voice signal  and  converts it  to digital (binary) format  for transmission over a digital circuit. The Analog-to-Digital Conversion Diagram illustrates this concept.

Analog-to-digital conversion is also called “A-to-D conversion” or “ADC.” The most common example of this is found in a codec. This device takes the analog voice signal and converts it to digital (binary) format for transmission over a digital path, such as a T1. The output  of  a  codec is  combined  with  other  outputs  and multiplexed onto a high-speed digital network

DTMF Signaling

As you saw earlier, the first step-by-step switches were designed to  work with  rotary-dialed  telephones. Those telephones used “dial pulse” signaling, which produced short, regular interruptions of the direct current flowing between a telephone and switch. The number of interruptions, or pulses, corresponded to the value of the digit. In other words, when you dial the number 5, you hear five clicks

As  CO switching  went  digital, telephone sets  also improved the  way they  transmitted telephone  numbers. The  dual  tone multifrequency (DTMF) system, commonly called Touch Tone, uses a pad of 12 buttons. When pressed, each button sends out a combination of  two pure tones not  found in nature: one high- frequency and one low-frequency. The DTMF Touchpad and Tones Diagram illustrates this concept

DTMF Touchpad and Tones

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By assigning one tone to each row and column, only seven unique tones are needed to identify each of the 12 buttons. These tones can easily be detected by a telephone switching system.

How a Call is Made

When you pick up a telephone handset, a sequence of predefined operations occurs that provides you the ability to use the telephone network. Now that you understand the basic components of the telephone system, let us see how they work together to complete a typical telephone call.

Dial Tone

When you lift the receiver, placing the telephone in the off-hook position, the telephone’s internal switch closes the local loop circuit with the CO switch. This allows electrical direct current to flow through the circuit; the presence of this current signals the CO switch that you need a telephone connection. In telephony terms, we say the CO switch has detected the off-hook condition.

The switching module of the CO switch then tests the line and determines its suitability for call processing. If the line tests good, the switch provides dial tone to the caller’s telephone.

The off-hook signal also alerts the switch to receive incoming touch tones. If the switch does not receive these tones in a timely manner, it sends a recorded message that reminds the customer that the telephone is off the hook.

Entering a Telephone Number

As soon as the CO switch detects the tones that represent the first digit, it removes dial tone from the line. The switch continues to detect tones and record the corresponding digits, while checking that the number of digits is correct. If the caller enters too few or too  many digits, the  switch sends  the  caller  error  tones or a recorded message.

Call Routing

The  switching  module  of  the  CO switch  then  checks with  its administrative module to determine the physical transmission path, or routing, the call must take to reach its destination.

If the called party is connected to the same CO switch, the call is connected by that switch. However, the administrative module is advised that those connections are in use, and notes are made for billing purposes.

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If the called party is connected to a different CO, the call goes through the caller’s CO, through a tandem switch, into the called party’s CO, then to the called party.

What if a particular telephone call is not originated and terminated within the same geographic region? How do we call another city, state, or country? The answer, of course, is to connect the caller’s CO to a higher-echelon CO.

Therefore, if a call is not local, it goes through the caller’s CO, up to  a Class 4 CO (or “toll switch”), into  the receiver’s local CO, then to the called subscriber.

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If the path is blocked at the Class 4 CO, the call is rerouted to another toll switch, into the receiver’s local CO, then to the called party (if necessary, the call may be routed up to a Class 1 office).

Ringing

After a call  has been routed  to the  destination CO switch, the switch tests the line to the called party, to determine whether the line  is  capable  of  processing  a call. If  the line tests good, the switch sends a ringing signal to the destination telephone.

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After  the  destination  telephone  answers,  by  going  off-hook, the CO switch removes the  ringing signal  from the destination telephone. At the same time, the destination CO switch signals the calling CO switch to remove the ringing signal the caller hears. Each CO switch also records that the call was completed, so that the proper party may be billed for the call.

Ending the Call

When either caller hangs up, putting one telephone in the on-hook condition  that  breaks the  local  loop  circuit,  the  absence  of electrical current sends a signal to the nearest CO switch that the call is finished. Both CO switches then perform a series of tests, advise their administrative modules that the call is ended, and label the  communications connections as  idle  and  ready for another connection.

Long Distance Voice Routing

This combination of digital switching and touch tone signaling made it simple to introduce direct dialing of long distance and international calls. And, telephone competition is made practical by powerful computers that can track and record the changing relationships between telephone companies and their customers. The typical sequence of steps to switch a long distance call is as follows:

1.- The caller lifts the handset and receives dial tone from the lo- cal CO switch.

2- The caller enters the called party’s telephone number.

3- The local CO switch identifies the call as long distance, based on the number and pattern of the digits.

4.- The local CO switch looks up the customer’s record to deter- mine which long distance company the customer uses, then routes the call to that company’s long distance switch (prob- ably in a Class 4 office).

5.- The long distance switch looks up the called party’s number to locate the CO switch nearest the called party, and connects to that switch.

6.- The destination CO switch tests the line and rings the destina- tion telephone.

7.- The called party picks up the handset and begins the conversa- tion.

8.- The source and destination CO switches record the comple- tion of the call, and begin to track the duration of the call.

9.- Either telephone goes on-hook and the circuit is disconnect- ed.

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10.- The source and destination CO switches record the final dura- tion of the call for billing purposes, perform testing, and label their connections as idle and ready for another call

Signaling

As you can see, an important part of the call routing process is the private communications, or signaling, that CO switches use to coordinate the work of setting up and tearing down telephone connections. In general, signaling is the exchange of information between  call components required  to  provide  and  maintain service.

Signaling means that service-related information is sent between a telephone company and its customers, between components of the same telephone company, and between one telephone compa- ny and another. For example, your local CO sends ringing or busy signals to your telephone. When you dial a number, you send an addressing signal to that CO, which then passes the number on to  other  COs across  the  country. When you  end  a call to a distant state, the CO switches that participated in the connection exchange duration information for billing.

In-Band vs. Out-of-Band Signaling

it is important to distinguish between “in-band” and “out-of- band” signaling. In-band signaling shares a single transmission channel with the voice conversation; voice and signaling must take  turns using  the same  transmission path. Analog (POTS) lines use in-band signaling. Therefore, on a POTS line, you can either talk or signal, but you cannot do both simultaneously.

Out-of-band  signaling is carried  over a separate channel  from voice. In other words, it does not take place over the same trans- mission path as a conversation. For example, Integrated Services Digital Network (ISDN) - Basic Rate Interface (BRI) a popular digital service, uses one 16 kbps digital channel for signaling and two 64 kbps digital channel, for voice or data, all carried over the same pair of copper wires at the same time. Sophisticated electronic  hardware  treats the  three channels as  if they were carried over different wires

Prefixes

The first couple of numbers dialed is the telephone number prefix. Within  the  North American Numbering Plan (NANP)  the  first three digits after the country code is the prefix and is mapped to geographic calling area. These three numbers are often called the area code

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Telephone Numbering System

As we have seen thus far, today’s heavily used telephone system depends on  the ability of callers  to place calls without the help of a human operator. That ability, in turn, relies on a system that assigns a unique numbered address to each telephone customer.

However,  there  is  no global  standard  for  telephone numbers. There is a North American standard, called the North American Numbering Plan, which is used to assign telephone numbers in the United States, Canada, Puerto Rico, and U.S. Virgin Islands, as shown in the North American Numbering Plan Table

In  this  table,  the  symbols  “N” and “X”  indicate  the  type of number that may appear in a particular position:

N: any digit from 2 to 9

X: any digit from 0 to 9

Therefore, as you can see in the table above, an NPA (area code) or NXX (local exchange prefix) may begin with any number from 2 to 9.

However, area codes and local exchange prefixes may not begin with  0 or 1, because  these numbers  have special meaning to the telephone  switch. If  the  first  number  the switch receives is a 0, regardless of what number follows, the switch immediately connects the call to  the operator. A number 1 in the first  position identifies the call  as  long distance. If a switch receives  a  leading number 1, it immediately transfers the call to the customer’s preferred IXC.

Each Class 5 CO is assigned blocks of NPA-NXXs to distribute to all customers who want telephone service within that wire center. Therefore, all subscribers of the same CO share the same three- digit area code and three-digit local exchange prefix.

The Dwindling Supply of Numbers

Each  NPA-NXX block can  identify  10,000 unique telephone subscribers, because the four digits of the subscriber line identifier include unique numbers from 0000 to 9999. Following a similar principle, each area code can support approximately 8,000,000 numbers. (Some area codes are set aside for special uses, as we will see later.)

Although  this may sound  like  a  lot  of  numbers, the supply of unique telephone numbers is being exhausted. Urban areas continue  to add  population, and  each  individual subscriber now wants multiple  lines  for  fax  machines, additional  voice services, and Internet access.

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Telephone number management in the United States is presently conducted by NeuStar, a company selected by the FCC to serve a  five-year  term, beginning  July, 2003, as  the North American Numbering Plan Administration (NANPA). As the world consumes more and more telephone numbers, this task becomes increasingly more complicated and difficult.

One solution, now being used in large metropolitan areas, is to use multiple area codes, called overlay codes, in the same geographic area. For example, a city’s dense downtown district may have a different area code than its outlying suburbs. Or, in heavily populat- ed cities, several area codes may serve the same area. This means residents of those cities must routinely enter 10-digit numbers to make local calls. In addition, the person next door, or even in the apartment below, may have a different area code.

This situation is further complicated by the presence of different telephone companies, such as CLECs and wireless carriers, who have their own NXXs that differ from the LECs. It is even possible that a business subscriber could use one LEC for inbound service, another for outbound service, and still another for Internet access or other services.

Currently, most telephone companies assign telephone numbers in three ways:

Random assignment, or “what you get is what you get,” is the most common method of assigning numbers. The telephone company  simply  assigns  the  next  number  from a pool of available numbers in the customer’s local exchange.

Special-request numbers are also available for an extra fee, providing  easy-to-remember business  numbers  such  as 444-9000 or 444-TAXI.

Numbers  may also  be reserved  for  future use. However, no LEC guarantees a number assignment until it is actually installed

Special Area Codes

Digital switching made it possible to use the telephone numbering system to access special services instead of geographical areas. A brief overview of some of the most popular services available through special-purpose NPA (area) codes is presented below.

Special NXX Codes

NXX codes usually identify CO; however, some NXXs, with or without a special NPA, are set side to access special services as described below.

Special Information: 555

The 555 numbers access special information services, such as long distance directory assistance. The line number (XXXX) identifies each individual service.

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Hearing-Impaired Services: 800-855

The 800-855 numbers, in the format 800-855-XXXX, provide free access to statewide relay services, such as Telecommunications Relay Service (TRS) and Message Relay Service, which provide trained assistants that translate calls between the voice telephones of  hearing customers  and  the  teletypewriters used  by hearing- impaired customers.

Service Codes, or N11 Numbers

Like speed-dial numbers, service codes are three-digit numbers that directly connect customers to local exchange special services. They are commonly called “N11” numbers because of their num- bering format, which follows the same rules as NPAs and NXXs. In the United States, the Federal Communications Commission (FCC) administers N11 numbers, which include:

211—Community Information and Referral Services (United States)

311—Nonemergency Police and other Governmental Services (United States)

411—Local Directory Assistance (sometimes 1411)

511—Traffic and Transportation Information (United States), Reserved (Canada)

611—Repair Service

711—TRS

811—Impending U.S. nationwide One-Call service for advanced  notice of  excavation activities.  This  allows contractors and others to dial a single number and notify utilities of their intent to excavate.

911—Emergency

Toll-Free: 800, 888, 877, etc

Toll-free calls (the called company, not the caller, pays for the call) have been available for more than 20 years, and consumers have come to expect companies to provide them as a customer service feature. With such a great demand for these calls, the 800 NPA code has run out of available numbers. Therefore, additional NPA codes, such as 888, 877, and 866, are now used for toll-free calls.

Some companies use toll-free numbers to dial into their PBX systems, which then gives them access to special outgoing long distance lines. This PBX system feature is called Direct Inward System Access (DISA). Although DISA can often be a cost- effective system for providing long distance service for traveling employees, hackers have attacked some companies and used these lines to steal long distance service.

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Premium: 900

Unlike 800 numbers, which companies provide free as marketing and  customer  service tools,  a  900  number  is  often used as a revenue-generating product. When calling a 900 number, the customer not only pays for the long distance charge, but is also charged a premium by the called company. The extra fee can be as high as $50 per minute.

Companies are using 900 numbers for customer support, fund raising, and pay-as-you-go services. By using a 900 number, a company can simplify its billing procedure. Because the customer’s telephone number is billed, the company does not need to create credit card transactions or issue a bill.

The  Federal  Trade  Commission (FTC)  has  established  rules that  dictate how vendors advertise  their  900 number services. For example, they must include in their print, radio, and television advertisements  the total  call  cost, if it  is a flat fee call, the per- minute rate if so billed, the fee ranges if they provide different call options, the costs of any other 900 number to which you may be transferred, and any other applicable fees. Other rules apply, and there are exceptions. See http://www.ftc.gov for more information

700 Services

IXCs  can use  the 700 area code  to  implement  new  services. These  numbers’ destinations  are  carrier  dependent, and some premium services use 700 area codes.

Follow-Me: 500

The  500 area code  was  originally reserved  so  that  carriers  to provide personal “follow-me” services. Some “telesleaze” vendors use 500 numbers to redirect callers to international numbers, and then bill the caller for the call at rates exceeding $2.00 a minute.

Number Formats

Currently, telephone  numbers are  decimal  digits  dialed  via the pressing of buttons. Each button press sends two tones down the phone line to the central office. These two tones are called DTMF – dual tone multifrequency.

The phone numbers dialed are used to dial local, long distance, or international  numbers. The format of  the numbers  is determined by  ITU-T  standards. These standards  are listed  in  the  E.164 recommendation. According to the standard, the telephone number must be 15 or fewer digits and begins with an international country code. Each country is responsible for defining the numbering plan within its own telephone network.

Telephone numbers adhere to the following structure:

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1.- If necessary, an access code may be dialed. This access code is required for international and non-domestic calls.

2.- The country code is dialed just after the access code. The coun- try code for the United States and Canada is 0. International numbers often are dialed with a + sign preceding the country code.

3.- The  area  code  points  to  the  called  geographical  area. Many locations currently require 10-digit dialing. When the geographical area “ran out of numbers” an overlay plan was implemented and the area code is dialed even within the local calling area.

4.- The local number is seven digits long. The first three select a local exchange or central office. The last four digits dialed select a port within that central office and represent the local loop for the subscriber called.

Number Blocking

Use Call Blocking to prevent the called party from viewing your direct phone number when you make a phone call. This can be done on a per-call basis by dialing “195” before placing your call. The person you are calling will see a “blocked call” or “private” message  on caller  ID. The actual  message  that  they  see will depend on the telephone company that they are using.

Anonymous Call Blocking

The anonymous call blocking feature of caller ID prevents a caller’s name and number from being sent with a telephone call. Telemar- keters, of course, may take advantage of this concept; however, a person who receives an anonymous call may simply choose to not answer it.

Digit Translation

Digit translation rules may add or remove digits to/from the dialed number, before a call is routed, so the call will reach its destination. Digit translation changes the dialed number to a different number. The dialed number may not be the number used by the system. The number may not be known on the PSTN – public switched telephone network. Area codes for local calls may be removed.

A digit  translation may  be set  in a  transformation mask value in a  Called  Party Transform  Mask  Field. Valid  (Cisco example) entries include the numbers 0 through 9 a wild card character and a blank. If no configuration is set the number will be sent without a transformation, i.e. the dialed digits are sent exactly as dialed.

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Digit  translation may be used  manipulate the caller’s automatic number identification (ANI), who is calling you, or to manipulate the dialed number identification service (DNIS) digits, the number the  caller  actually  dialed, for  a  voice call. DNIS  captures the dialed DTMF - dual tone multifrequency digits.

Toll Fraud

Hackers try to use enterprise telecommunications systems to gain “free” access to outside facilities. Though free to them, toll fraud annually costs companies millions of dollars in unauthorized toll charges. Potentially more severe is the operational impact; once in, hackers can block outbound and inbound line access

Once they have located a weakness, hackers attack relentlessly, often sharing this information with others. They attack in a number of  ways:  through  maintenance  ports, voice  mail,  automated attendants, and remote access services. Hackers may gain system access by shoulder surfing (observing and intercepting dialed digits), dumpster diving (acquiring telephone records from a company’s trash), diverting calls, and placing agents inside the company.

We can protect our voice systems against toll fraud in a number of ways:

Remote access/Direct Inward System Access (DISA)— To  enter  customer  premise-based  PBX  systems,  avoiding inbound  call  charges,  hackers  frequently  first  call  toll-free numbers. Once connected, these hackers use random number generators  and password-cracking  programs  to  locate  a number combination that provides them outside facility access.

We can protect our voice systems from this type of toll fraud by first evaluating our remote access needs. If unnecessary, we can turn off this feature and close a system backdoor. If we need this service, we can begin our protection efforts by using unpublished inbound numbers. We can configure the system with barrier and authorization codes of the maximum length, and administer the system so that it does not provide dial tone while waiting for barrier or authorization code entry.

We can set the system to disconnect after a preset number of invalid remote access attempts. We may restrict remote access outbound area codes and access hours. We may set authoriza- tion codes to raise Class of Restrictions (COR) for specific users or user groups.

We  can  protect  our remote maintenance  ports  by  changing these ports’ login IDs and passwords, as toll defrauders are well aware of  the vendor-supplied  default  IDs and passwords. If using PC-based emulation programs for administrative access, we can ensure that we do not store dial-up numbers, logins, or passwords as part of an automatically executed script.

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We can install port security devices on each end of the main- tenance link. Avaya markets a Remote Port Security Device (RPSD) that consists of a modem-sized lock and key. We place the lock on the PBX-end serial port, while we place the key on the maintenance terminal port. The lock and key must match before the devices will open a communications link

Automated  attendants—Automated attendants are vulnerable as well. Hackers try to find a menu choice, even one unannounced, which would lead to an outside facility.

The following auto attendant security tips apply:

1.- Never allow a menu choice to transfer to an outgoing trunk without defining a specific destination.

2.- When any of  the digits zero  through nine are not  menu options, program  them  to transfer  to an  attendant, an announcement, a disconnect, or other intercept treatment. If the numbers eight  and  nine are  PBX feature access codes, translate these menus to an extension.

3.- To  prevent specific  call  types or calls  to  other  CORs, assign specific CORs and Facility Restriction Levels (FRLs) to each port. Since the PBX treats the auto attendant ports as stations, we can configure these ports as we do stations. FRLs work  with call permissions and  route  patterns  to determine where calls may be placed. The higher the FRL, the  greater  the calling privileges. We would  want  to set auto attendant ports to the lowest possible FRL.

4.- Assign the auto attendant ports a class of service (CoS) restricting outward call access.

5.- Restrict our auto attendant menu options to transfer only to internal extensions or announcements.

Voice messaging—Voice messaging systems are also toll fraud targets. Criminals attempt to transfer to automatic route selection (ARS) dial access codes, trunk access codes, trunk verification  codes,  facility  test  call  access  codes, or data origination  codes  through  the voice messaging  system. A criminal needs to have only a touch-tone telephone to break into an inadequately secured voice messaging system.

The following security tips apply:

1.- Protect voice messaging systems by restricting transfers back to the host PBX. Instead, disallow transfers, use enhanced  call  transfer  features, or  allow  transfer to subscribers only.

2.- Use maximum length passwords where feasible

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3.- Deactivate unassigned voice mailboxes, creating voice mailboxes  only  when  needed. Upon  new  extension activation, require  users  to  immediately  change their voice mail password.

4.- Never announce  that  the called  individual will accept third-party   billed   calls.   This allows   unauthorized individuals to charge calls to the company.

5.- Never use obvious and trivial passwords, such as the telephone  extension,  room number,  employee  ID, social security number, or easily guessed number/letter combinations.

6.- Change adjunct default passwords immediately.

7.- Lock out consecutive unsuccessful voice mailbox entry attempts.

8.- Discourage users from writing down passwords, storing them, or  sharing  them with others. Secure passwords that must be documented.

9.- Never program passwords into auto dial buttons.

10.- Ensure that the carrier provides reliable disconnect to the PBX. Reliable disconnect prevents the CO from returning dial tone after the called party disconnects.

IP  PBXs—Since  VoIP  systems  are  fundamentally  data networks  carrying  voice traffic, not  only are they vulnerable to toll fraud, but also to the many security threats typical data networks confront. VoIP systems commonly tie into the PSTN. This  means that  a hacker  who  gains  access  to  the VoIP network can potentially call anywhere in the world by means of the network’s PSTN gateway.

These  security  measures  are  only  as  effective  as  those  who exercise  them. We must educate users on  their  implementation, and establish a written toll fraud security policy. Additionally, we must  exercise effective  physical  security, so that only authorized personnel may access administrative and attendant consoles, PBX and messaging systems, and wiring closets

ENUM

ENUM stands for Electronic Number. ENUM is described in RFC 2916. ENUM is based on Domain Name Services (DNS). ENUM maps telephone numbers to IP addresses and domain names. The telephone numbers are defined by ITU-T E.164 recommendation.

1.- ITU-T E.164 defines the structure, format, and hierarchy of telephone numbers

2.- A fully qualified E.164 number has a country code, an area code or city code and a subscriber or station number.

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3.- The IP addresses are defined by the RFCs of the Internet by the IETF

Naming Authority Pointer (NAPTR) is defined in RFC 2915 and is a DNS Resource Record (RR). When given a telephone number the  DNS server, using  NAPTR will  return a populated Uniform Resource Identifier (URI).

Examples of URIs include:

1.- http - hypertext transfer protocol

2.- https - hypertext transfer protocol secure

3.- ftp - file transfer protocol

4.- mailto - email address

5.- tel - telephone number

6.- sip - session initiation protocol

7.- ldap - lightweight directory access protocol

URI’s  are  pointers  to  land  line  phones, mobile  phones, faxes, SIP  Servers, voicemail, email  addresses  including VPIM (Voice Protocol  for  Internet  Mail), instant messenger and  IRC aliases, web pages, global call forwarding, and others.

The Réseaux IP Européens (RIPE) is responsible for administration and coordination of WAN IP networks in their area of authority. The  RIPE  operational  instructions  for  an E.164  domain  are provided here: www.ripe.net/enum/instructions.html

To build an ENUM to domain translation,

1.- Take the initial phone number: +1-800-555-1212

2.- Remove all characters but keep the numbers: 18005551212

3.- Separate the numbers with dots/periods: 1.8.0.0.5.5.5.1.2.1.2

4.- Reverse  the  numbers, this is  to map  the  number  to  DNS reverse number lookup: 2.1.2.1.5.5.5.0.0.8.1

5.- Append  the  Tier-0  DNS  zone  of  e164.arpa, this domain has not  been agreed  upon by all  nation states in the world: 2.1.2.1.5.5.5.0.0.8.1.e164.arpa

Thus, when the telephone number of +1-800-555-1212 enters the network; a resolver on the client will query a DNS server. The DNS server will find the phone number 1-800-555-1212 in the domain e164.arpa and return a URL.

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An ITU-T E.164 phone number of (07) 3224 8444 (Queensland, Australia,  Disability Information)  becomes  in  ENUM  format: 4.4.4.8.4.2.2.3.7.0.e164.arpa (Advanced Research Projects Agency).

Starting from the right:

1.- e164.arpa is the top level (Tier-0) DNS domain name for ENUM

2.- the 7.0 is the reversed country code zone number

3.- the first digits 4.4.4.8.4.2.2.3 are the local, within the nation, phone number

(For  a  list  of  country  code  telephone  zone  numbers  go to http://www.wtng.info/wtng-cod.html)

1.- United Kingdom +44

2.- France +33

3.- Australia +61

4.- United States +1

5.- Canada +1

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Each member state is responsible for the final decisions concerning usage and adoption of the ENUM within their sovereign nation's zones.

It is planned that all public phone numbers will enter the domain e164.arpa. The domain structure from e164.arpa on down will be the authoritative 'root' for E.164 telephone numbers. Thus a search from a phone number to a URL will only traverse one branch of the DNS tree. In the global dialing plan the domain is: e164.arpa. Private phone numbers may be attached to any needed domain. In a private dialing plan the domain could be your company's domain: e164.example.com

Electronic numbers supports all IP-based communications. This in- cludes voice, video, fax, voice mail, instant messaging, SMS, MMS, paging, etc. After passing a telephone number to DNS, the user could be contacted at their phone, email address, web site, IRC identity, SIP gateway, etc. The phone companies are considering using the ENUM number as each individual's one and only phone number since it may be converted via DNS to nearly any URL

When  an endpoint  starts  a VoIP  call  using an  IP address,  the ENUM system will, if necessary, translate that phone number into an IP address. The ENUM system will first determine if there is a registered IP address for the called party. If that IP address exists the call  is made  IP  to  IP. If that IP  address is not registered, the request  is sent  to  the PSTN - public switched telephone network to complete the call. SIP connections will map the PSTN telephone number to sip:[email protected] via a DNS request.

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Electronic Number – ENUM Using DNS for Translation

Each telephony product should ultimately include a DNS resolver to initiate the requests from that handset, or endstation.

Countries may decide for themselves whether to join the ENUM system.  The  control   of assigning   E.164  phone  numbers  is the  responsibility  of   each  sovereign   state.  Each DNS  zone administrator is responsible for adding their zone's phone numbers into their e164.arpa domain.

ENUM is managed by the ITU - International Telecommunications Union but is operated by IAB - Internet Architecture Board

Local Number Portability

As we saw earlier, the Federal Communications Commission (FCC) has mandated that a long-term solution  to local  number portability (LNP)  be implemented  by  the telecommunications  industry. In other words, the FCC wants customers to be able to switch service from a LEC to a CLEC (or back again) without losing their existing telephone numbers.

As of  November, 2003, both  wireline  and  wireless  telephone service providers must support LNP in the same local geographic area. In other words, if you choose to change carriers, whether from  wireline  to  wireline, wireline  to  wireless, or  wireless  to wireline, in most cases you can keep the same telephone number. The  FCC has granted  certain smaller carriers temporary stays from this requirement, and litigation is still ongoing in many regions. Additionally, in some areas you can now port an existing number to a Voice over Internet Protocol (VoIP) carrier, such as Vonage or AT&T’s Callvantage service.

The Advanced Intelligent Network (AIN) architecture supports LNP. The AIN uses the SS7 network to enable carrier switches to share information such as Location Routing Number (LRN). An LRN identifies a switching port (the carrier) for a particular telephone number.

When a caller dials a number, the local CO switch uses SS7 mes- sages to query a number database. This database provides the LRN assigned to the dialed number. Using the LRN, the CO switch can route the call to the called number’s current LEC. If the subscriber changes carriers, only the LRN changes, not the telephone number. AIN also allows cellular telephone users to roam between networks. The AIN Support for LNP Diagram shows how the AIN supports LNP

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Network  Equipment  Building  Standards  (NEBS)— Originally developed by Bell Labs, the NEBS requirements enable telcos to build networks that can survive and continue operating  under   the  most   severe  conditions.  The  two documents describing NEBS requirements are:

GR-63—Physical equipment protection

GR-1089—Electromagnetic  compatibility  and electrical safety

Local Number Portability (LNP)—The  two  Telcordia GRs  addressing  local  number portability  services  and operations are:

GR-2936—Switching  and  signaling  requirements  to support  portability between service providers  located on the same or different rate centers

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GR-2982—Specific provisioning, call processing, network management, signaling, and other requirements involved in supporting portability across rate center boundaries

Voice  over  Packet  Technologies—These  GRs  outline performance  and  operational requirements  for  interfacing voice over packet  networks to  legacy  telecommunications technologies. Some important requirements include:

GR-3051—Voice over packet call connection agents (CCAs) (gatekeeper functions) including call flows (call setup and release), CCA functional architecture, network access, interfaces to other network elements, call connec- tion and processing procedures, gateway operations, call routing, and network management and operation

GR-3053—Signaling interfaces between voice over packet and the SS7 networks (signaling gateway)

GR-3054—Interfacing voice over packet networks to the PSTN (trunk gateway)

GR-3055—Interfacing  PSTN  line-side  devices  (analog telephones, ISDN equipment, PBXs, and other equipment) to a voice over packet core network (access gateway)

GR-3060—Transporting   legacy   telecommunications services across voice over packet networks

Emergency Service

E911 is Enhanced 911 and based on FCC guidelines that are designed for public safety and emergency preparedness. It is required that “interconnected” VoIP providers supply E911/911 service. Both the origination and termination call locations must supply  E911 access  via  wireline, wireless,  and  broadband network connections. If the “interconnected” provider supplies phone service, then 911 must also be supplied.

Interconnected VoIP service is defined as a broadband Internet connection  over  which packet-based  telephone  calls  and/or standard PSTN telephone calls are made. (DSL, cable networks or broadband wireless networks)

The telephone networks of most countries support an emergency services telephone number. This phone number allows a caller to contact emergency services like police, fire department, medical, or emergency rescue services. Although services and service phone numbers differ by country, most emergency numbers are short, typically three digits, to help make them memorable.

Emergency numbers like 911 in the United States are intended to be used in emergencies only. For routine inquiries or non-emer- gency services, traditional 7 or 10 digit numbers should be used. Routine calls, prank calls, and other non-emergency calls should not be

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made to the emergency services numbers. Persons making inappropriate calls to emergency services are often prosecuted in a court of law. Traditional phone numbers are assigned to a local loop, a specific house or business - a specific geographic location. This makes it easy for emergency response teams to pinpoint exactly where their services  are  needed. The location of  the  number remains fairly static. Thus, emergency service operators can, when an emergency call is disconnected, call back to the location of the emergency, With packet-based telephony, mobility is a strong benefit, except in cases of emergency. The call made in an emergency on a cable access network could have been made from anywhere. When using VoIP E911  services. The consumer  must  register  their physical with their VoIP service provider and keep that information up to date

The 9-1-1 system was initiated by AT&T in 1965. “The National Emergency Number Association (NENA) estimates that as of February 2005, some form of 911 service was available to nearly 99 percent of the population in 96 percent of the counties in the United States” As referenced in See National Emergency Number Association, 911 Fast Facts (visited Apr. 25, 2005) http://www. nena.org/911_facts/911fastfacts.htm (NENA 911 Fast Facts).

NENA also states approximately 200 million calls are made within the United States to the 911 emergency services each year. The cost of building and maintaining the 911 infrastructure is borne by the state and local governments across the nation. In August of

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1999 the United States adopted 911 services for wired as well as wireless calls, thus there is a required end-to-end emergency system for wireline calls, wireless calls and VoIP calls

Voice Terminals

Today’s telephone technologies offer a variety of ways to com- municate using voice, video or both. Customers have the choice of communicating via a computer, a standard analog handset, a USB to computer connection, and more. These virtual or physical phones support multiple protocols and codecs, including H.323, SIP, G.711, G.729 and many others

IP Phones

A computing device with embedded telephone VoIP software. No computer is required only Internet access. Instead of a standard RJ11 interface to connect to an analog POTS line, the phones con- tain an RJ45 to Ethernet connector. These IP phones are available for a price of $70 for a basic model up to around $700 for multi- featured models that run additional applications and have a color LCD interface. These IP phones may be powered by standard AC transformers or can receive their power via the new IEEE 802.3af Power over Ethernet standard. Cisco PoE phones currently use a proprietary Cisco standard for transmitting power. To enable Power over Ethernet on a non-PoE phone an additional converter will be required.

TDM Phones

Time division multiplexing (TDM) phones sample the speaker’s voice in blocks, but these blocks of voice payload are sent across the TDM bus (a separate Ethernet segment) with a small header prepended to the payload. These blocks are then sent, in order, across  the  bus based on  time – thus you have a virtual circuit running Ethernet packets.

Standard phones operate using TDM. When MACs (moves, adds, and changes) are required, a phone technician must be called and a change must  be  scheduled. When  the phone  is IP-based, the phone merely is unplugged from the Ethernet jack and moved to a new Ethernet jack. The phone will work with no changes required in the closet, punch down block, or PBX

Analog Phones

The most  common method  used  to connect  an analog phone to VoIP is through an ATA an Analog Telephone Adapter. This device communicates directly with  the VoIP server using  the  configured protocol. Possible protocols include: H.323, SIP, MGCP, IAX, etc. The  ATA and not  the analog phone  will  process the coding and encoding of the voice. ATA boxes do not require a PC

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Analog Telephone Adapter Connecting Analog Phones to IP networks

Dialup Hard Phone

A dialup hard phone is an IP phone with a built-in modem instead of a built-in Ethernet NIC. The device dials a remote VoIP server and a call is made. In areas without broadband access, a dialup hard phone, a phone line, and an Internet Service Provider (ISP) are the only requirements to make a VoIP call.

PC-Based Soft Phones

A soft phone is a virtual phone created in software. Soft phones are installed on computers to emulate a hard phone. The functionality of  hard  phones and  more  may  be emulated  in the soft phone. Many soft  phones will support  both voice and video streams. A soft  phone requires an  audio  card, microphone, and  headset or speakers. Or, the  phone may be a USB connected handset. Soft phones are cheaper than their hardware counterparts and may often be downloaded for free. Soft phones may support H.323, SIP or both. Versions  are available  for  Apple Macintosh OS X, Linux, Microsoft Windows operating systems, mobile PCs and PDAs

Microphones and Headsets or Speakers

The local computer’s audio card captures the voice stream from the local microphone and delivers the audio from the remote end via the local headset or speakers. VoIP speakers accept a digitized audio stream from IP phones, whether hard or soft phones. This audio stream is the audio from the far end playing locally through a USB phone, a pair of speakers, or an overhead paging system

USB Hand Sets

USB (Universal Serial Bus) phones are remote control devices for the soft phone running on the computer. The USB device provides microphone, speakers, and a dial  interface. The soft  phone itself does  the  voice  encoding/decoding and connection management. Be aware  before  buying  a  USB  hand  set, the USB software/ hardware device drivers are operating system specific

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SIP Phones

Phones designed to communicate using Session Initiation Protocol (SIP) may  be a  hard phone, soft  phone or an adapter allowing the caller  to use  their ‘legacy’ analog  phone. The  hardware SIP phone has an RJ45 connector to directly connect the phone to an Ethernet network. Two or more parties may converse when using SIP phones. To make a call from SIP phone to SIP phone, the following pieces are required: a soft  phone or a hardware  phone adapter  on  both end stations, for quality digital voice call, a broadband connection to the Internet is required, to dial from a SIP phone to a phone on the PSTN – Public Switched Telephone Network, a virtual number is assigned to your SIP phone.

PDA Soft Phones

PDA soft phones have the same functionality as PC-based soft phones but instead run on a personal digital assistant on a mobile computing software platform like Windows Pocket PC, Windows Mobile, or Palm OS.

WiFi Phones

Wireless  Fidelity phones are hard  phones  with an  integrated WiFi transceiver. There must be a wireless base station to allow the WiFi phone to contact the VoIP server. WiFi phones often include support for  the cellular network  as well. Thus if  wireless is not available the phone will  switch over  to a GSM network. GSM stands for Global System for Mobile communications

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NETWORK ENGINEERING

Network Requirements for Convergence

If a network is to carry a wide variety of traffic, it must have the following capabilities:

• Identification—The  network  must  be  able  to  tell  the difference  between  frames that contain  video, audio, file transfers, or interactive commands.

• Prioritization—The network must detect whether some frames are more urgent than others.

• QoS or class of service (CoS)—The network must be able to provide varying types of service, according to the type and urgency of each frame. For example, the network must know that a high-capacity, long-delay link is better for file transfers, while a low-delay link (even with less capacity) is better for streaming multimedia

Converged Network Architecture

There  are  many  ways  to  implement  a  converged  network. We might  assume  that convergence  requires  a  homogeneous infrastructure, so that a network is either completely packet based and connectionless (such as shared and switched LANs, or packet- service WANs), or completely connection-oriented (such as ATM to the desktop and long-distance ATM clouds).

A Hybrid Approach to Convergence

In  practice, neither  type of  homogeneous  architecture  is viable, due  to  the different economic and  performance  requirements for LANs and WANs. A converged network that spans large distances, shown in the Wide Area Converged Network Diagram, has a WAN core network surrounded by LAN edge networks.

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In general, the edge networks will use different technologies than the core. Furthermore, for many reasons, each edge network may be  based  on  a different  technology  than  the others. One edge network may be based on a switched Ethernet fabric (one without Layer 3 routing),  another  on routed  Ethernet  segments, and  a third on ATM LAN technology.

The WAN core may consist of a single technology network, such as frame relay, ATM, or the Internet. Alternatively, it may consist of multiple parallel networks, some connection-oriented and some packet-switched, as shown in the WAN Core Technologies Diagram.

The WAN core may consist of a single technology network, such as frame relay, ATM, or the Internet. Alternatively, it may consist of multiple parallel networks, some connection-oriented and some packet-switched, as shown in the WAN Core Technologies Diagram.

Parallel WAN Paths for QoS

An architecture that uses multiple parallel paths offers significant advantages to a converged network. It is possible to solve many QoS  problems  in  a  LAN  simply  by radically  oversupplying network capacity (in other words, by throwing bandwidth at the problem). However, this  is not  economically  feasible in a wide area network (WAN) because WANs are engineered to optimize their resource use for a particular class of traffic. This means that a single WAN technology cannot provide good service for all the types of traffic that cross a converged network.

Connectionless packet-based networks, which make up a large portion of the Internet, provide good service to bursty, non-time- critical traffic. They do not deliver good service to traffic with tight bandwidth,  delay,  and  jitter  requirements.  On  the  other hand, connection-oriented   networks,  such  as  ATM,  provide  good service to traffic with tight bandwidth, delay, and jitter requirements. However, it is costly to use ATM networks for bursty traffic.

Consequently, a converged network is likely to have a core that consists of multiple WAN networks. The edge LANs carry voice, data, and  video  traffic  over  a  common  physical infrastructure. However, at the LAN/WAN boundary, traffic is classified by QoS and routed over the WAN network that provides the most appro- priate QoS. For example, bursty, non-time-critical traffic will be routed over a packet-switched WAN. Multimedia data, however, will probably be routed over a connection-oriented network that provides QoS guarantees

Application-Level QoS

Application-Level QoS Converged networks may be able to use the application-level QoS principle to optimize application performance by customiz- ing network devices on an application-by-application basis. For example, a network could filter traffic according to the

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appli- cation, then handle the traffic according to each application’s unique processing requirements. Active networking, in which applications download small programs or configuration data into network devices, is another example of this principle.

Before networks can guarantee QoS on a per-application basis, some  important  issues must  be  addressed,  such  as  security, resource management, and inter-device coordination. However, an organization could enjoy a significant competitive advantage from a network that could optimize services for important applications.

Voice Quality

In a converged network, changing network conditions have dif- ferent effects on network performance and voice quality. “Voice quality” is a  subjective  term that  describes  how accurately a digitized and transmitted signal represents the original sound of a human voice. The perception of voice quality is generally a combination of several different factors:

• Clarity describes how much of an original sound is recorded by a digital  signal, and how much outside  noise is included. Clarity  can  be  affected  by  many  factors, such  as silence suppression, voice coding and decoding, jitter, loss, and noise.

• Echo occurs when some of a transmitted signal returns to the sender. As network delay increases, so does echo.

• Delay occurs when packets are not received when they are expected. If packets do not arrive promptly and at a constant rate,  the  illusion  of  a  continuous  voice  signal  cannot be maintained. Delays can cause annoying gaps in a conversation. The effect of delay is not usually noticeable until the sum of all delays exceeds 500 ms for a single end-to-end transmission (one  way).  Thus,  network  engineers  work  to  limit  these combined voice network component delays to no more than 200 ms. We will discuss delay in more detail shortly

The QoS Parameters

All discussions concerning QoS in data networks focus on some or all of the following parameters:

• Bandwidth—The raw measure of the network’s physical or logical data handling capacity. This is measured as an absolute value (10 Mbps Ethernet), or a minimum value (a frame relay circuit’s Committed Information Rate [CIR] of 56 kbps).

• Throughput—Throughput is the actual data carrying capacity of  a  network. Throughput is  calculated  by  subtracting delay, overhead, and other components of normal network operation from the raw bandwidth. On a 10 Mbps Ethernet segment, once overhead,

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collisions, device latency, and other factors are considered, actual throughput may only approach 2 Mbps.

• Delay, or Latency—Delay is the time it takes packets to cross a network. Any device that manipulates a packet can introduce delay. The measure is a maximum value, such as a maximum of 200 ms round trip delay, or 110 ms end-to-end delay.

• Jitter—Jitter is variable delay. Jitter is measured in a number of ways, such as a percentage of the maximum network delay (+/- 10 percent), or as a variation above or below the average network delay (+/- 10 ms of the average network delay over a period of time).

• Wander—Wander is the variation of a signal from its timing reference. Synchronous circuits use a clock reference to ensure that  each device  sending or  receiving a  digital data stream knows where the information begins and ends. If a clock ref- erence degrades or becomes unavailable, the digital signal will eventually wander from the reference. Other network devices will lose track of where data is located in the data stream.

• Packet Loss—Also known as information loss or just loss, packet loss measures the number of packets lost across the network as a ratio of the total sent. For example, packet loss between network nodes could measure 1 packet in 10,000 undelivered (lost).

• Availability—Availability measures the amount of time the network is available for use, as a percentage of the total time in a day, week, month, or year. Much is mentioned about achieving five 9s (99.999 percent) availability on a network.

• Security—Security is not a measured value, but nonetheless is an important consideration when determining the QoS on a network. We would want only authorized calls on the packet voice network, and we may want to encrypt voice triffic traversing the public Internet.

Bandwidth

Bandwidth  refers  to  the  raw  data  carrying  capacity  of  a particular carrier technology. For example, a T1 circuit carrying several frame relay permanent virtual circuits (PVCs) provides a bandwidth of 1.544 Mbps. The individual PVCs provide some bandwidth up to the T1 port’s maximum physical bandwidth.

In any network, whether voice or data, bandwidth is a primary QoS concern. As network traffic grows, so does bandwidth utiliza- tion. As bandwidth utilization increases, throughput decreases. At some point, even the common practice of “throwing bandwidth at the problem” reaches its practical limits, and so we are forced to address bandwidth limitations in our network designs.

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In converged networks, voice traffic must often contend with bursty data for scarce network bandwidth. Bursty traffic, if uncontrolled, can monopolize the network’s bandwidth. Hence, we must use QoS techniques such as admission control and traffic shaping to control and buffer bursty traffic entering and traversing our networks.

Voice codecs require a minimum amount of bandwidth to operate. The G.711 codec defines 64 kbps PCM voice encoding and requires at least 64 kbps of bandwidth. If we try to push PCM encoded voice calls across a link with insufficient bandwidth, the call will fail.

Data applications will generally perform better if given more bandwidth;  they  are  said  to be  bound  by  the  network’s bandwidth, or bandwidth bound. Voice applications, on the other hand, do not perform any better if given more bandwidth than the application requires. Voice application performance is bound to the network’s delay; that is, once given enough band- width to enter the network, the network delay decides how well they perform. Delay is not determined by network bandwidth alone, but also by network component and link performance.

Throughput

Throughput measures the actual network performance, that is, the network’s actual ability to transfer data. Bandwidth describes the  ideal  condition,  while  throughput  is  a  more practical measurement  of  the network’s capacity. Many  factors can effect network throughput:

• Collision rates on contention-based networks

• Traffic volumes on network segments • Network device performance (switches, routers, firewalls, gateways)

• Maximum transmission unit (MTU)

• Fragmentation rates

• Delay

As you might imagine, calculating actual throughput can be quite daunting, as  the  list of variables  indicates. However, we  can calculate the instantaneous throughput packets experience on the network if we know the packet size and how much time elapses as they travel from the sender to the receiver.

For example, assume you send a 500 byte packet across a network segment. Using a packet sniffer, you capture the time the packet leaves the sending node. You are also able to capture the time the packet arrives at the recipient node. Since you already know the packet size, you can calculate the throughput the packet experi- enced by dividing the packet size

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by the end-to-end transmission time. If the packet took 90 ms to reach the recipient, then the net- work throughput provided the packet equals (500 bytes * 8 bits/ byte) / 90 ms = 44,440 bits / second.

Delay

Delay, also known as latency, is the time a circuit requires to carry bits from one point in the network to another. Bandwidth-bound applications  realize a  lower  delay when provided  with more bandwidth; the greater the bandwidth, the lower the delay.

Delay-bound (voice) applications, also known as latency-bound applications, specify the maximum allowable delay voice packets may experience as they traverse the network. For example, many VoIP applications dictate that voice packets must not experience more than 200 ms delay end-to-end.

We can calculate a packet’s delay at any point on the network by comparing the time the packet arrived at a point to the time it leaves that point. For example, if an IP packet arrives at a router’s ingress port at time 14:25:00.000 and it leaves the egress port at time 14:25:00.090, we can subtract the departure time from the arrival time. The difference is 14:25:00.090 - 14:25:00.000 = 90 ms, or 90 ms delay.

Delay can be measured at many points in the network. The TCP/ IP traceroute utility allows us to measure delay at each hop in a packet’s path. We can use packet sniffers and protocol analyzers to evaluate end-to-end and round trip delay by monitoring out- bound and inbound TCP and RTP messages, and calculating the time it takes for the sender to receive TCP acknowledgments, or by comparing the RTP send time to the received time. Important to remember is that delay will vary at each hop depending on net- work bandwidth, congestion, and the QoS mechanisms in place

In a real-time, delay-sensitive application, we won’t necessarily decrease delay if we only throw more bandwidth at the problem. This is because bandwidth alone does not determine the delay packets will experience across a network. Additional processing delay, also called “serial delay,” is added by devices that encode and decode signals, compress data, or assemble data into packets.

Encoding, Decoding, and Compression

It takes time to convert a voice signal from analog to digital. A similar delay, or latency, occurs when the remote end converts the digital signal back to analog.

Voice compression reduces the number of bits in a transmission by removing redundant character strings in the digital character stream. However, compression algorithms increase

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latency by requiring large voice stream samples before they can compress and packetize voice signals.

Packetization

It also takes time for a device to accumulate enough voice samples to fill a packet. For example, codecs sample an analog voice signal once every 1/8000th of a second, or every .125 ms, and represent each sample  with  eight  binary  bits. To  fill  the  data portion  of a  default  IP packet  (576 bytes), we  would  need  556 bytes  x .125 ms/byte = 69.5 ms.

Delay also occurs when a large packet is fragmented to cross a link that has a smaller MTU. To reduce or eliminate this delay, we can adjust network devices and endpoints to use the smallest MTU on the path.

Thus, by reducing packet size, we can eliminate fragmentation, and reduce the time needed to build packets. However, too short a packet  results in  high  packet  overhead. Hence,  we need to weigh packetization delay against packet overhead to come to an agreeable compromise.

Achieving this balance can be a challenge on high-speed LANs that are optimized for data transfers. For example, some Gigabit Ethernet networks use a proprietary approach, known as “jumbo frames,”  to  provide  a  better ratio  of  data payload  to packet overhead. The MTU of a jumbo frame is larger than the Ethernet standard MTU of 1,518 bytes; some jumbo frames can be as large as 9 Kb. Although  jumbo  frames can  improve  overall network performance, they can increase the problem of packetization delay by requiring more time to fill each frame.

Jitter

Delay creates problems even when it is consistent and predictable. But  there  are  also many  sources  of  variable  delay. Varying packet  sizes,  varying  levels  of  network segment  congestion, network device performance, QoS techniques that queue packets based on network conditions (DiffServ), etc., all cause packets to experience varying delay conditions.

For example, a sending node may transmit some packets immedi- ately, but queue other packets while waiting for free bandwidth. This is common on shared-media networks, when collisions cause the sending device to back off and attempt retransmission.

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Intermediate devices can also create queuing delays. For example, if a switch or router treats all packets with the same priority, a voice packet can get stuck in the queue behind other, less time- critical packets; however, the voice packet right behind it might not incur the same delay.

When a signal is subjected to variable delays, several packets may arrive over a large time interval, or conversely, a clump may arrive all at once. We call this effect “jitter.”

Jitter Buffers

A jitter buffer, in the receiving endpoint’s audio codec, can reduce the perceived delay by accumulating packets and releasing them at a constant rate. However, jitter buffers introduce additional delay because of the time needed to fill them. The length of this delay depends on the network’s expected maximum jitter.

For example, assume that voice packets will experience a maximum of  100 ms  delay between  the sending  and  receiving endpoints (actual  packet-by-packet  delay  could  vary from  some  value less than 100 ms to the maximum, and possibly beyond). In that case, we can configure the receiving codec’s jitter buffer to hold 100 ms of audio samples before releasing them for playback.

If the sending codec creates voice samples every 10 ms, those samples  should (ideally) reach  the  receiver at  10 ms intervals. However, we know that the end-to-end network delay is variable, so the second sample might arrive 20 ms after the first, the third 15ms after the second, and so on.

The receiving buffer queues up 100 ms of samples before playing them out to the listener. The receiving codec plays out a sample every 10 ms. If the packets come every 10 ms, the buffer will remain full. If they come slower, the buffer queue will shorten, but will still play back a steady audio stream to the listener, as long as packets continue to arrive in time to keep the buffer full. If delayed packets arrive in a clump, they refill the buffer queue

Playback problems develop, however, if network conditions vary too much. If packets arrive too slow, the buffer can underrun (empty too fast), causing gaps, or “clipping,” in the playback audio. If the packets come too quickly, the buffer may overrun and drop packets.

A static jitter buffer, one set to a specific value, is vulnerable to overruns and underruns. Additionally, the static buffer always introduces a fixed delay, regardless of network performance.

Dynamic jitter buffers adjust to the varying rates at which packets are  received, so  they can  reduce or  increase their latency as network performance changes. The result is a reduced likelihood of buffer overruns and underruns.

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Measuring Jitter

Jitter can be measured in a number of ways. If a VoIP application specifies acceptable jitter as a relative value, jitter can be measured as a percentage deviation relative to a maximum allowable delay value. If the application states an absolute jitter value, this means that the stated jitter value is the absolute allowable range. As a deviation from the average jitter, the application could specify an acceptable jitter range that references an average value over time. No matter how jitter is specified, jitter outside the stated range will cause poor voice quality.

As you may have surmised, non-real time traffic is less sensitive to jitter than is real-time traffic. Hence, we need not spend a great deal of effort chasing down jitter sources for non-real-time traffic.

However, when we combine real-time and non-real-time traffic, we must consider jitter if we wish to play back voice messages, send faxes, and push other jitter-sensitive traffic across the same network segments over which our bandwidth bound traffic travels

Wander

Wander occurs when synchronous data deviates from its timing source. Since IP networks are asynchronous, wander is not an issue. However, we need to consider wander whenever voice traffic requires an associated clock to ensure accurate timing, such as on a synchronous WAN link.

WAN data circuits use buffers to minimize wander. These buffers store a certain amount of the received signal, and may retime the signal for transmission to upstream devices.

Packet Loss

Packet loss, to a certain extent, is a normal occurrence on best- effort networks. TCP provides loss recovery mechanisms, such as message acknowledgments and sequence numbers. Thus, non- real-time traffic is generally more forgiving of lost packets than is real-time traffic.

However, the human ear is not so tolerant. Unless the network employs specific countermeasures, such as forward error correc- tion or error concealment, users will hear annoying gaps and clicks. Whether a user can perceive this effect depends on the packet loss rate and compression algorithm used. The higher compression schemes are less forgiving.

There are many reasons why networks lose packets. Some QoS architectures and techniques deliberately discard packets as part of their traffic control protocols. Routers, switches, and other network devices may drop packets due to network congestion or device

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malfunctions. Even redundant network links can drop in-transit packets as the network recovers from an outage.

VoIP applications will commonly specify the amount of packet loss for which they can compensate. The application can also specify whether it can utilize packet retransmission as a recovery method. Note that if a real-time application allows for lost packet retransmission, this will probably be implemented through the use of acknowledgments and buffers in the sending node, which will add latency to the network

Availability

If a network is not available for users, it is just an expensive conglomeration of wires and devices. To some businesses, occasional network downtime is only a minor nuisance. To others, such situations can cause serious damage, including:

• Lost productivity because of idle workers

• Lost revenue when customers go elsewhere

• The direct support expense of returning the local area network (LAN) to service

Some network downtime cannot be avoided; equipment failures and link outages do occur from time to time. A key to maintaining high network availability is preventive maintenance. However, pre- ventive maintenance at times requires network downtime, as well.

To recognize the effect varying availability rates can have on a network, consider the following:

There are 31,536,000 seconds in a year (excluding leap years). With a network availability rate of 99%, the network would be available for use 31,220,640 seconds per year, or 23.76 hours per day. This means that the network is available for use 361.35 days per year. This also means that it is unavailable for 3.65 days per year. If a company makes $1,000,000 per day on the Internet, this means a revenue loss of $3.65 million a year due to network downtime.

However, if the network provides a 99.9% availability rate, then the network is offline only .365 days per year. This reduces their network downtime losses to just $365,000 per year.

Network device vendors target the elusive “five 9s” (99.999%) availability rate as the ultimate level of availability. Additionally, service providers may promise a service restoration time in a number of hours from the failure time; this limits the risk that multiple failures will affect service.

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Service level agreements (SLAs) often outline provider penalties for failing to meet these availability targets. Availability often varies by link, as carriers may supply multiple links as components of an overall service contract. Since entire networks rarely fail, one WAN link could fail more frequently than others. Though this link may have a lower availability rate, the provider could still meet their contract obligations, their availability numbers bolstered by better performing links. Hence, SLAs should be written on a link by link basis, rather than as an aggregate performance specification

Security

Though previously not considered a QoS parameter, the ever- increasing incidence of network break-ins and virus attacks has brought security into light as another important network QoS factor.

Several concerns comprise network security issues. Enterprise users want to protect proprietary and business critical data while at the same time allow network users remote access to voice and data  applications. Internet call  centers  must expose corporate resources  to everyday  security threats  while at  the same time protecting these same resources from theft or destruction.

Some methods used to protect network data include encryption, either public-key or private-key. User authentication methods such  as  strong  passwords, digital  signatures, and  biometric authentication, used individually or combined, serve to ensure only authorized users access protected resources.

Various software vendors, working groups, and professional organizations have devised architectures for insuring data privacy, confidentiality, and validation. One such architecture is IPsec, a standards-based secure IP protocol used to support secure Internet transactions. IPSec can be used in VoIP applications as well. IP packets are marked to identify their security requirements, and all IPSec aware devices recognize and support this header marking.

As  a  part  of  an  overall  QoS  policy, VoIP  network  devices can specify that a traffic stream’s packets receive a certain secure treatment across the network. This security policy can carry across both private and public networks

IP Header Features

Even  though  IP is  a best-effort  protocol, its  header can carry information that other protocols can use to provide QoS features.

ToS Field

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The  IP header type  of  service (ToS) field indicates  the QoS desired by a packet’s original source. There is no guarantee that other devices on the network will honor this request.

IntServ

The Integrated Services (IntServ) architecture is defined in the informational RFC 1633. IntServ aims to provide predictable and guaranteed services to specific traffic flows, called streams.

Before a stream of traffic can flow, the sender and receiver must set up and maintain a path that can support that stream. Each router along the path must provide the requested resources to each defined stream. If a router cannot provide the resources the stream requires, it informs the reservation requester of this fact with an error message.

The IntServ network nodes include traffic control functions implemented with traffic classifier, admission control, and sched- uler  elements. The  traffic  classifier maps inbound  traffic into classes, based on IP header contents or some other classification value. The  admission control  element  decides  if a router can provide the stream its guaranteed QoS. The packet scheduler manages stream queues.

RSVP

Resource Reservation Protocol (RSVP), defined in RFC 2205, is used to set up and maintain stream reservations for IntServ archi- tecture networks. When a host needs to reserve network resources for  a  stream,  it  uses  RSVP  messages  to  request  a reservation for those resources

RSVP requests resources in one direction, working backward from the destination. When the receiving node requests resources for a stream, those RSVP messages travel from node to node along the stream’s desired path. Each router responds with either an acknowledgment or error message, then passes the resource request to the next-hop upstream router. Once the reservations are in place, each router maintains the stream’s state information for the duration of the connection. An RSVP request may be initiated as the result of an H.323 call’s setup procedure, where the endpoints negotiate the call resources needed to complete the connection.

CBQ

Class Based Queuing (CBQ) is a queuing algorithm developed by the Lawrence Berkeley National Laboratory. CBQ divides the available network bandwidth among different traffic classes. Each class is assigned a queue, and each queue is assigned a portion of the link’s bandwidth. CBQ allows us to classify traffic flows into hierarchies. Each class can be

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divided up into sub-classes, and each sub-class is given a specific treatment within its parent class.

You will recall that we previously discussed DiffServ access control and traffic shaping functions. In our DiffServ architecture’s ingress routers, we classified, metered, and marked inbound traffic flows as in- or out-of-profile according to their class, and provided differentiated services to each flow based on packet markings. CBQ enables us the ability to classify traffic by IP or port addresses, application, or priority (ToS or DSCP).

A unique aspect of CBQ is that it shares unused bandwidth between traffic classes. A flow that needs to burst above its assigned flow rate can “borrow” unused bandwidth from other classes. This “bandwidth borrowing” capability would support the traffic burst shaping function previously discussed

COPS

As we have seen, many QoS methods assign priorities to traffic flows, but don’t provide a mechanism to manage and enforce those priorities  across  the  enterprise. Common  Open Policy  Service (COPS)  provides  this  capability.  IETF  RFC  2748  describes COPS, a client/server model for supporting policy control over QoS signaling protocols, such as RSVP for IntServ and DiffServ PHBs:

• The  policy  repository  contains  a  central  database  that describes how to handle each inbound flow. This device could be an LDAP server, though this is not specifically defined.

• The COPS server is called the Policy Decision Point (PDP), a server where policy decisions are made based on information provided  by the central  database. The  PDP can be a policy management server or workstation.

• A COPS client is a device that must enforce policies. The client is called the Policy Enforcement Point (PEP) because it acts on the server’s instructions to apply and enforce policies on the network. The COPS client decides how to handle each flow by requesting information from the COPS server. More than one PDP may query the repository simultaneously.

The PEP can be a router, PSTN gateway, firewall, or some other network edge device capable of sending COPS messages to the PDP. For example, on a DiffServ network, the client is the ingress router that performs the admission control and traffic shaping functions.

The COPS Network Diagram illustrates a COPS network.

Though specified separately, the PEP, PDP, and policy repository can be located on the same device. A Local Policy Decision Point (LPDP) can also be used, but all policy decisions must reference a single PDP as the final decision authority.

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The COPS protocol also addresses fault tolerance, by requiring that the PEP and PDP exchange keep-alive messages to verify the connection. If the connection between the PDP and PEP fails, and the PEP can find no backup or alternative PDP, then the PEP can make local decisions. The PEP is responsible for notifying the PDP of  any  policy  changes made  under  local  control  while  the connection is down.

MPLS

Multiprotocol Label Switching (MPLS) aims to speed packet flows across a routed network by assigning each packet a small label tag. This tag represents the packet’s destination IP address, so that interior routers can route packets by the label tag rather than the destination IP address. When an MPLS tagged packet arrives at an MPLS-compatible router, the router reads the tag value and forwards the packet quickly to the next hop.

MPLS can interoperate with such QoS architectures as DiffServ. The label tag, attached ahead of the packet header, can represent not just the destination address, but additional packet information such as:

• Precedence

• VPN membership

• RSVP QoS information

• The packet’s route, as chosen by traffic engineering (TE)

A set of routers within an MPLS network composes an MPLS domain.  The  MPLS domain  routers  are  called  Label  Switch Routers (LSRs). On the edges of the MPLS domain are ingress and egress nodes. The ingress node is responsible for assigning a packet to a certain forwarding equivalence class (FEC). An FEC defines a group of packets to which the domain provides a speci- fied treatment. Each subsequent router acts upon the packet’s label rather than the IP header. Labels are locally significant, and change at each hop.

The downstream LSRs inform upstream LSRs of label mappings using a label distribution protocol (LDP). Each LSR maintains these mappings in a Label Forwarding Information Base (LFIB). LFIBs serve as packet label routing tables, replacing traditional IP routing tables. The MPLS Domain Diagram illustrates an MPLS network

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MPLS can be used in traditional frame-based LANs, such as Ethernet or Token Ring networks, or in cell-switching networks, such as ATM networks. When used in an ATM network, MPLS enables the edge switches to map inbound packets to MPLS labels, which are in turn mapped to ATM cell Virtual Channel Identifier (VCI) or Virtual Path Identifier (VPI) fields. Once mapped, the core ATM switches assign paths to the cells based on the cell label to VCI/VPI mappings.

Packets entering an ATM network through the same physical port could  potentially  be assigned  to  the same VCI. These packets would carry different types of data, such as routing table updates, network management traffic, e-mail, and voice traffic. The ingress switch would interleave these inbound packets with no regard to their  precedence. However,  if  the  ATM  switch  assigns  these packets MPLS labels based on their designated QoS requirements, and in turn maps these labels to different VCIs, the ATM network can  carry  the  QoS  requirements  of  these  packets  into  the cell-switched network

Policy Routing

Policy-based routing allows network administrators to specify additional information sources for the routing table and network model. These sources may include information imported from other protocols or information that network administrators statically configure. Such policies can be defined on a router- by-router basis and control routing information advertisements. They define who can talk to whom, who can listen to whom, and what types of information are transmitted and received.

Policy-based routing is often part of network security procedures. For example, a network administrator might specify that network information imported from other protocols be included in the routing table and network model, and subsequently shared with outside routers as part of update messages. However, routers with custom security settings and

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other private configuration information would share the information only with other routers that share the same security settings.

RED

In times of network congestion, Random Early Discard (RED) attempts to maintain the integrity of each traffic flow. It actively manages output queues by randomly dropping inbound packets once  the queue  reaches a  preset  level. This  forces underlying protocols (TCP, RTP) to initiate lost-packet recovery procedures, which slows the inbound data rate of each flow

WFQ

Weighted Fair Queuing (WFQ) algorithms attempt to give each inbound  traffic  flow  fair access  (fair queuing)  to  the outbound port’s available bandwidth. This access can be determined by each flow’s precedence (weighted) as set by the packet header ToS bits or RSVP reservations. Since WFQ assigns a weight to each flow, it is often used to support QoS architectures, such as IntServ and DiffServ.

We weight the queues by assigning each one a priority. A higher priority queue receives more of the available bandwidth, while a lower priority queue receives less. However, each queue receives some level of predictable service on the outbound port. This differs from the  first-in, first-out  queuing  that is  the  default  for many network devices. It also contrasts favorably with PQ, that gives high-priority packets absolute network access at the expense of lower priority flows.

WFQ does not discriminate based on packet size. In other words, queue A may have three 200-byte packets, while queue B could have  two  300-byte  packets.  If  the  device  gives 600  bytes bandwidth to each queue in turn, then queue A will empty before queue B forwards its packets.

WRED

Just as WFQ provides a queue weighting mechanism, Weighted Random Early Discard (WRED) adds weighting to the RED algorithm. WRED attempts to improve on RED by randomly dropping packets from low-priority flows first. If congestion increases, packets are then dropped from higher-priority flows

DiffServ

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Internet Engineering Task Force (IETF) Request for Comments (RFCs)  2474  and  2475 define  basic  ways  of  treating  different subsets of  a  network’s traffic  in  different  ways at  OSI  Layer 3. For  example, some packets  might  be routed  in ways to expedite delivery and minimize delay, while others are routed to minimize loss.

DSCP Code Points

The differentiation of these services is provided by a redefinition of the IPv4 Type of Service (ToS) field (called the Traffic Class field in  IPv6). Differentiated  Services (DiffServ) uses  the  ToS octet’s bits 0-5 to identify a Differentiated Services Code Point (DSCP), a procedure used to handle packets on a per-hop basis. The DiffServ Modified ToS Field Diagram illustrates the DiffServ octet.

The first three bits of the field (bits 0-2) should be standardized across networks. RFC 2474 uses these three bits to define eight class selector codepoints. These 3 bits prioritize the traffic, with larger numerical values having a higher relative order or priority.

Bits 3-5 can be used to define specific types of local services. However, these non-standard definitions may not be recognized by devices outside the local network.

Bits 6 and 7 are currently unused.

As you have learned, 802.1p also defines eight levels of priority. Thus, it is possible to map 802.1p traffic classes to DiffServ DSCPs to match Layer 3 traffic flows

PHB

PHB Key DiffServ operational features are Per-Hop Behaviors (PHB). A PHB is a forwarding behavior a node applies to a packet based on that packet’s condition. The packet’s DSCP indicates the particular PHB a packet should receive as it passes from hop to hop.

Standard PHBs

The DiffServ architecture standards define several standard PHBs to  help  assure predictable  behavior  across  DiffServ  domains (described later). These are:

• Default  PHB—Defined  in  RFC  2474,  the  default  PHB represents no special handling, or the network’s default best- effort service. All DiffServ nodes must support the default PHB, which sets all six bits to zero: 000000.

• Class-Selector  PHB—Also  defined  in  RFC  2474,  the class-selector PHB sets the first three bits of the DSCP field to match the IP packet header ToS bits. This assures backward

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compatibility in networks where ToS bits are used to represent service classes. DiffServ nodes must support these PHBs where IP precedence bits are used.

• Assured Forwarding (AF) PHB—Defined in RFC 2597, the assured forwarding PHB aims to provide assured packet forwarding services across networks. Traffic is assigned to one of four traffic classes.

When  using  the  AF  PHBs,  packets  within  each  class are marked  to one of  three drop precedence levels. AF assigns resources to each class, and allocates those resources according to a packet’s drop precedence. AF provides for a total of 12 class-to-precedence pairings.

A DiffServ domain’s use of the AF PHB is optional. See RFC 2497 at http://www.ietf.org/rfc/rfc2497.txt for more information on the AF PHBs.

• Expedited Forwarding (EF) PHB—Defined in RFC 2498, the EF PHB sets a specific PHB for assignment to a single pre- ferred traffic flow within the DiffServ domain. This preferred flow would be one that absolutely must receive low jitter, low delay, and low loss treatment across the network. By setting the DSCP to ‘101110’, we guarantee that this one flow receives at least the EF PHB-defined bandwidth at each departure point. The EF PHB is frequently used in DiffServ domains with two traffic flow classes, such as real-time voice traffic carried over the same network as non-real-time data.

A DiffServ node does not have to support the EF PHB. See RFC 2498 at http://www.ietf.org/rfc/rfc2498.txt  for more information on the AF PHBs

Custom PHBs

A network  administrator may  use  bits  3-5 to  define  custom DSCPs. However, routers outside of the local network will not necessarily recognize  these custom DSCP endcodings that do not terminate in 000 (xxxyyy).

DiffServ Domains

In DiffServ terminology, the local network, whether one subnet, an entire Autonomous System, or a collection of Autonomous Systems, composes a DiffServ domain. In the DiffServ domain, all network devices, both on the network edges and intermediate nodes, are configured to recognize and act upon the packets’ DSCPs.

A PHB defines the services each node will provide the different traffic flows. PHBs can vary from hop-to-hop, depending on many variables. Each network segment can experience differing conges- tion levels, bandwidth, delay, and jitter conditions, and DiffServ allows us to define per-segment PHBs for each traffic class.

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IP Precedence

IP  provides  the  capability  to  assign  packets  one  of  eight precedence levels, according to the traffic’s importance. RFC 791 defines the different precedence levels, and how they apply across network boundaries. The RFC leaves it up to the network architec- ture as to how IP precendence is handled. As we have learned, we can map IP precedences to DiffServ DSCPs, and identify flows and streams by this IP header component

Traffic Prioritization

As we have discussed previously, a converged network must be able  to provide varying types  of  service, according  to the type and urgency of each frame. Thus, it must first be able to identify the type of traffic each frame contains, and determine the priority of each type of frame.

Two new standards work together to identify and prioritize traffic on an Etnernet LAN. First, the 802.1Q standard identifies each frame  according  to  its  VLAN  membership. Then, the 802.1p standard uses these VLAN identifiers to prioritize real-time traffic over other types of frames.

802.1Q VLAN Standard

The 802.1Q specification, published in December 1998, provides a standard definition of Ethernet VLANs and their use in switched networks. The 802.1Q standard strictly defines procedures and protocols for establishing and maintaining VLANs. The uniform rules of the standard simplify, and, in some instances, automate VLAN  configuration  and management,  regardless  of  which company’s switches and end-station interfaces are used.

The VLAN Tag

Each virtually defined workgroup of a VLAN may include individual members scattered across a large, extended LAN or WAN. The 802.1Q specification associates physical devices and ports to each defined  VLAN  in  the  network,  then  maps  and  shares  those associations with other LAN stations

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802.1Q does this by adding a 2-byte VLAN tag, or identifier, to each frame. The tag identifies the virtual workgroup to which the frame belongs, and includes priority and QoS information. With its 12-bit VLAN identifier, 802.1Q can support thousands of VLANs across switch trunks and 802.1Q-capable routers.

Tagging provides the necessary information for switches to com- municate and create the VLAN. Tagging also enables a more com- prehensive set of control functions than has been possible through proprietary VLAN frameworks. With an identifier in each frame, VLAN membership, priority, and QoS are highly manageable. The 802.1Q VLAN Tag Format Diagram illustrates this concept.

802.1Q defines two tagged frame types: a VLAN-tagged frame and a priority-tagged frame. A priority-tagged frame sets the first three tag bits to identify the frame’s priority. A VLAN-tagged frame sets the tag’s last 12 bits to represent VLAN membership information. The  three tag priority  bits allow any one of  eight priorities to be assigned  to a  specific packet. The  802.1p standard defines the eight priorities from highest to lowest (note that priority 0 is higher than priority 1):

• 7—Network management traffic

• 6—Voice traffic with less than 10 milliseconds (ms) latency

• 5—Voice traffic with less than 100 ms latency

• 4—Controlled load traffic (mission critical data applications)

• 3—Traffic meriting extra effort by the network for prompt delivery (such as an executive’s electronic mail [e-mail])

• 2—Reserved

• 0—Traffic meriting the network’s   best effort for prompt delivery (the default priority)

• 1—Background traffic such as bulk data transfers and backups

802.1Q as an Architecture

VLAN operation now includes workable guidelines for multivendor networks,   allowing consistency   in   membership  assignment, activity, and administration. The architecture of 802.1Q specifies three levels of operation:

• Relay, or forwarding of frames, based on the two-byte VLAN tag. This smaller 802.1Q “footprint” compares favorably to such vendor-specific VLAN tagging techniques as Cisco

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System’s (Cisco)  Inter-Switch  Link  (ISL)  protocol,  which  increases the original frame’s size by 30 bytes by encapsulating the frame with an ISL header and a 4-byte frame check sequence (FCS). The ISL protocol frame encapsulation causes non-ISL-aware devices to reject these frames as invalid.

• Explicit sharing of VLAN information and exchange of topology information.

• Uniform VLAN management and flexible VLAN configuration.

802.1Q Protocols

802.1Q  frame  tagging  is  key  to  uniform  sharing  of  VLAN information among switches in the network. The standard uses several specialized protocols to do this, enabling creation of VLAN configurations by propagating information between switches and end stations throughout an Ethernet network. Other protocols provide information about multicast traffic for traffic containment and efficiency:

• Generic Attribute Registration Protocol (GARP) uses tags to propagate topology information among switches and end stations.

• GARP VLAN Registration Protocol (GVRP) is the means by which VLANs themselves are configured, in some cases automatically. Using this protocol, stations request admission to specific VLANs. Membership in a VLAN is determined by a network management or policy system. GVRP helps simplify the administration of VLANs by handling registration of end stations  with  Ethernet  switches  and  maintaining  current information about membership

GVRP may be used in end stations and switches. When the protocol is used between end stations across a large network, some switches may not be GVRP-aware; therefore, GVRP information exchanged between end stations is transparent to the intermediary device. In other cases, the intermediary switch recognizes GVRP. In this context, GVRP embedded in VLAN-aware switches may be integrated with a network management console and policy server. This simplifies tracking VLAN additions and departures.

• GARP Multicast Registration Protocol (GMRP) is used to create and change multicast groups dynamically.

• Internet Group Management Protocol (IGMP) broadcasting  is  used  with Distance Vector Multicast Routing Protocol (DVMRP) to build routes for delivery of multicast messages.

802.1p Traffic Prioritization Standard

With more and more networks moving to the speed and simplicity of  Layer 2  switching, the  IEEE 802.1p  standard  provides an easy and effective method for prioritizing LAN

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traffic. Developed by IEEE, 802.1p supports priority transmission of time-critical information in a LAN environment.

Priority Tags and Traffic Classes

While Token Ring and FDDI frames have defined priority fields, Ethernet does not. Therefore, the 802.1p standard recommends use of 802.1Q VLAN tags to prioritize Ethernet frames.

As we just discussed, VLAN tags are two-byte headers that identify each frame’s VLAN, type of traffic, and priority. As we can see in the 802.1Q VLAN Tag Diagram, three bits of the tag identify the priority level of each frame

Because three bits can define eight values, 802.1p defines eight traffic classes. These traffic classes and their associated traffic types are listed in the 802.1p Traffic Classes Table.

Traffic Class (Priority) Tag Value (Binary) Traffic Types7 111 Network control6 110 Interactive voice5 101 Interactive multimedia4 100 Controlled load app3 011 Excellent effort2 010 Spare1 001 Backgroud0 000 Best effort

How 802.1p Prioritizes Traffic

The first step in prioritizing traffic is to assign a traffic class to each type of frame. Traffic prioritization policies are centrally configured and administered through management software. After a tagged frame is transmitted onto a network, 802.1p-compliant switches recognize the traffic class value and forward the frame according to its priority.

To do this, the switches must have multiple queues implemented in hardware. Lower-priority traffic is buffered in a lower-priority queue, while higher-priority traffic passes through the higher- priority queue. Therefore, switches with only one queue cannot effectively implement prioritization.

Given the increasing speed of networks, two queues (one for business-critical and time-sensitive traffic, and the other for best- effort and background traffic) are generally sufficient to ensure high QoS for applications that need it

Mapping 802.1p Traffic Prioritization to IP and ATM Networks

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By definition, 802.1p is intended for packet-based Layer 2 LANs. In WANs and the Internet, IP has a defined Type of Service (ToS) priority  field.  Originally  intended  for network  administration traffic, this field has been relatively unused.

ATM, with its connection-oriented technology, offers the ultimate in traffic prioritization and guaranteed QoS. In most instances, however, ATM is used as a backbone technology with Ethernet, Fast Ethernet, or Gigabit Ethernet deployed at the edge for connectivity to desktops and servers.

To prioritize mission-critical and time-sensitive traffic across these backbone networks and WANs, 802.1p traffic classes must be converted to the corresponding IP or ATM CoS. For example, the IETF Integrated Services (IntServ) working group is defining the cross-classification mapping of DiffServ codepoints to 802.1p traffic classes. Meanwhile, other technical working groups are developing techniques for mapping 802.1p to ATM connection services.

Using QoS to Optimize Voice Quality in VoIP Networks

Quality of Service (QoS) is the ability of a network to provide better service to selected network traffic using various technologies. You must plan for QoS by deploying features that apply these technologies throughout your network. One of the most important things is QoS must be configured throughout the entire network to improve your network’s performance.

Using QoS with Voice over IP (VoIP) will allow you to grant priority service to voice as well as servicing the data side with the priority that is specified. By utilizing QoS for the Voice you can implement reliability, predictability and eliminate poor quality voice transmissions.

Through the use of packet classification you can mark specific traffic which effectively partitions your network traffic into different levels. This classification is accomplished when you add a tag to the IP Precedence/ Differentiated  Services  Code  (DSCP),  source and  destination  IP addresses and you can classify packets using access control lists, policy- maps, and similar techniques

Any network carrying voice traffic also carries data which shares a path through the network. This interaction will have an affect the application performance which can lead in possible congestion and packet loss. Congestion is the direct result of a sustained overload of  traffic which  requires you  to finds  methods  that  will control congestion once it occurs, and  then allow you  employ strategies that will use some sort of queuing features such as those using the CISCO IOS as follows:

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• Weighted Fair Queuing—This applies priority to identified traffic to classify traffic into conversations and determine how much bandwidth each conversation is allowed relative to other conversations. WFQ classifies traffic into different flows based on  such characteristics  as  source and  destination  address, protocol, and port and socket of the session.

• Class-Based  WFQ  (CBWFQ)—CBWFQ  extends  the standard WFQ functionality to provide support for user-defined traffic classes. It can specify the exact amount of bandwidth to be allocated for a specific class of traffic. Taking into account available bandwidth on the interface, it can configure up to 64 classes and control distribution among them.

• Priority  Queuing  -  WFQ  (IP  RTP  Priority  Queuing) (PQWFQ)The IP RTP Priority feature provides a strict priority queuing scheme. This allows delay-sensitive data such as voice to be dequeued and sent first, before packets in other queues are dequeued. This feature can be used on serial interfaces and Frame Relay permanent virtual circuits (PVCs) in conjunction with either WFQ or CBWFQ on the same outgoing interface. In either case, traffic matching the range of UDP ports specified for  the priority queue  is guaranteed strict  priority  over other CBWFQ classes or WFQ flows; packets in the priority queue are always serviced first.

The  Frame Relay IP RTP Priority  feature provides  a strict priority  queuing scheme  on  a Frame Relay PVC for delay- sensitive traffic such as voice.

Voice  traffic  can be  identified  by  its  Real-Time Transport Protocol (RTP)  port numbers and  classified  into a priority queue configured by the frame-relay ip rtp priority command. The result of using this feature is that voice is serviced as strict priority in preference to other non-voice traffic

• Low  Latency  Queuing  (LLQ)—LLQ  provides  strict priority  queuing on ATM  virtual circuits (VCs) and  serial interfaces. This feature is able to configure the priority status for a class within CBWFQ, and is not  limited  to UDP port numbers as well as IP RTP Priority. LLQ and IP RTP Priority can be configured at the same time, but IP RTP Priority takes precedence

Network Performance Issues

An intimate knowledge of network design issues can increase the quality of voice calls over a packet-based network. There is a multitude of issues that can affect throughput on a network.

When packets are delayed by congestion or slow routing protocols, delay or latency will reduce the quality of the voice call. Propaga- tion delay describes the time taken by a signal

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to pass through the media whether that is copper, fiber, or wireless. Each device along the path from speaker to listener introduces handling delay. Han- dling delay may be caused by the encoding/decoding of analog speech to a digital stream, or by the size of each created packet. Queuing delay is based on how long it takes for a packet to leave the device that created that packet. Packets may sit in the queue due to network congestion.

Jitter is a variation in the delay of packets. Say one packet is sent every 20 milliseconds; these packets which were sent together arrive at different times. The delta time between when a packet is expected to arrive and when the packet actually arrives is called jitter.

Pulse Code Modulation (PCM) converts analog sounds, like your voice, into a digital stream. This stream is created by sampling your voice 8,000 times each second. By applying the Nyquist theorem, and sampling at twice the rate of the highest frequency a voice quality sample will be created. Since the human voice range is 300 to 3,400 – about twice 3,400 is 8,000 samples per second.

Voice compression is applied to a voice stream in North America by applying the u-law logarithmic compression algorithm. Other countries  use  the  a-law algorithm. The u-law algorithm works slightly better for low-level signal-to-noise reduction. If making an international call from u-law territory to a-law territory, it is the job of the u-law country to make the conversion.

Echo is when the speaker hears their own voice delayed on the phone line. Delays of more than 25 milliseconds are intrusive and should be avoided by setting echo cancellation. Echo is removed by the local router which stores an inverse image of the speaker’s speech pattern to lay over and cancel the sounds coming back from the remote speaker’s phone.

Packet loss is expected and part of the original design of Ethernet networks. Packet loss can be mitigated by giving precedence to time sensitive traffic over standard data packets

Voice activity detection

Digital-to-analog conversion

Tandem encoding

Transport protocols

Dial-plan design

Voice Quality

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Packet telephony is a lot like a motion picture: the illusion of a continuous voice signal can be maintained only so long as voice packets arrive promptly and at a constant rate, just as a movie’s individual frames roll off a projector reel.

Although cheap Internet telephones have unfairly given packet telephony a bad name, it is true that transmission delays across the packet-switched network can hurt voice quality. These delays cause annoying gaps in a conversation, and may be introduced at a number of points.

It takes time to convert a voice signal from analog to digital. Voice compression algorithms increase this delay, requiring large voice stream samples before they can compress and packetize voice signals. A similar delay occurs when the remote end converts the digital signal back to analog.

A packet needs time to move through the network, especially if it is relayed through multiple router hops. This propagation delay remains minimal as long as the packet remains on a LAN equipped with wire-speed switches. However, speed disparities at the LAN/ WAN edge interfaces can create considerable queuing delays.

It also takes time for a device to accumulate enough voice samples to efficiently fill a packet. RFC 879 defines a default IP packet size of 576 bytes. Codecs sample voice intelligence once every 1/8000th of  a  second, or  every  .125 ms, and represent  each sample with 8 binary bits. To fill the data portion of this default IP packet, we would need 556 bytes x .125 ms/byte = 69.5 ms.The shorter the packet, the shorter the packetization delay. However, too short a packet results in high packet overhead. Hence, we need to weigh packetization  delay  against  packet  overhead  to  come  to  an agreeable compromise.

Jitter

The variability of delay, also called jitter, creates a more trouble- some  problem  than packetization  delay. Traffic conditions on the backbone network, such as head-of-line blocking or serializa- tion delay, primarily cause jitter delays. For example, if we treat all packets arriving in a switch (or router) with the same priority, a  voice  packet  can get stuck  in  the queue behind other, less timecritical packets; the next voice packet traversing the network might not incur the same delay. The result is an accordion effect; several packets may arrive over a large time interval, or conversely, a clump may arrive all at once.

Without appropriate design, jitter can wreak havoc on a telephone conversation. Jitter buffers, which store a string of packets, can smooth  out  the  packets’  arrival  rate; however, jitter  buffers introduce additional delay.

Voice Compression

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The default voice digitization rate is the Pulse Code Modulation (PCM) voice rate of 64 Kbps. The simplest way to reduce the bandwidth  required  to transmit  digitized voice is to use voice compression. Voice compression does not actually work on the voice signal but compresses the digitized voice signal by removing redundant character strings in the digital character stream.

Latency

Network latency becomes obvious to the callers on a VoIP con- nection when the round trip delay is more than 250 milliseconds. The ITU-T recommends unidirectional, end-to-end latency never exceed 150 milliseconds, from caller to listener. If latency is too excessive, one caller will sense the pause created by the network as a pause created by the other caller and both callers will begin to speak.

Latency is a delay in packet delivery. Latency may be caused by  many  factors:  length  of the  media,  number  of  routers between callers, encryption delays, voice encoding/decoding, or any other delays on the transmission network.

To reduce latency, each endpoint and all intermediate network devices should be optimized for throughput in the jitter buffers, packet sizing, and configured with appropriate QoS levels.

Packet Loss

Most network end stations can accept or recover from an occa- sional lost packet. The human ear is not so tolerant, and unless the network employs specific countermeasures, such as forward error correction or error concealment, users will hear annoying gaps and clicks. Whether a user can perceive this effect depends on  the  packet  loss rate and compression  algorithm used. The higher compression schemes are less forgiving.

The effect of delays on a voice conversation is not noticeable until it exceeds 500 ms. At that point, the user will perceive that there is a delay occurring. In IP telephony, the use of jitter buffers as a part of the communication link can store and forward the message one

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delivery packet at a time, and reduce the perceived delay. Packet delays cause gaps in the conversation, and may require the listener to request that the sender repeat information previously sent.

Because voice traffic is real-time traffic, network engineers must work to limit these combined voice network component delays to the maximum acceptable end-to-end (one-way) delay of under 200 ms

Port Settings

Firewalls may block specific ports or a range of ports to disallow traffic in or out of the network. By default network administrators want  to  block  as many ports as possible  to reduce the risk of intrusion. For VoIP to operate, a pair of UDP or TCP ports must be opened. The first  even number is opened for the voice traffic; the very next odd numbered port is opened for control of that voice traffic. Enabling  ports  to allow VoIP to pass through the firewall opens a huge range of port numbers starting with 1024 and moving, if given the chance, two at a time all the way to 65,535. Each call opens another set of ports. Several firewall vendors offer a dynamic firewall  solution. This  solution only opens  ports on demand and closes the ports back down when the voice call is terminated.

Example ports that must be opened on the firewall for some VoIP applications include:

For Microsoft Netmeeting and others using H.323

TCP port 1720 (H.225 call signaling for hosts)

TCP ports 11000 to 65535 (H.245 capability exchange)

UDP ports 16384 to 32767 RTP audio stream)

For  http://www.callserve.com  over H.323

UDP port 1719 for signaling

TCP port 1720 for signaling

UDP ports 5000 to 65535 for the RTP audio stream For Session Initiation Protocol (SIP) UDP port 5060 for SIP signaling TCP port 5060 for SIP signaling

UDP ports 16384 to 32767 for the RTP audio stream

For  http://www.skype.com

“Ideally, outgoing TCP connections to all ports (1 to 65535) should be opened.

” Or, “open up outgoing TCP connections to port 443.

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” Or, “open up outgoing TCP connections to port 80.”

Bandwidth

Bandwidth refers to the raw data carrying capacity of a particular carrier technology. For example, a T1 circuit carrying several frame relay permanent virtual circuits (PVCs) provides a bandwidth of 1.544 Mbps. The individual PVCs provide some bandwidth up to the T1 port’s maximum physical bandwidth.

In any network, whether voice or data, bandwidth is a primary QoS concern. As network traffic grows, so does bandwidth utiliza- tion. As bandwidth utilization increases, throughput decreases. At some  point, even  the  common  practice of “throwing bandwidth at the problem” reaches its practical limits, and so we are forced to address bandwidth limitations in our network designs.

In  converged  networks, voice traffic  must often contend with bursty  data  for  scarce network  bandwidth.  Bursty  traffic, if uncontrolled, can monopolize the network’s bandwidth. Hence, we must use QoS techniques such as admission control and traffic shaping to control and buffer bursty traffic entering and traversing our networks.

Voice codecs require a minimum amount of bandwidth to operate. The G.711 Codec defines 64 Kbps PCM voice encoding and requires at least 64 Kbps of bandwidth. If we try to push PCM encoded voice calls across a link with insufficient bandwidth, the call will fail.

Data applications will generally perform better if given more band- width; they are said to be bound by the network’s bandwidth, or bandwidth bound. Voice applications, on the other hand, do not perform any better if given more bandwidth than the application requires. Voice application performance is bound to the network’s delay; that is, once given enough bandwidth to enter the network, the network delay decides how well they perform. Delay is not determined by network bandwidth alone, but also by network component and link performance.

Network Capacity Baselining

Traffic baseline measurements serve as a rough guide for the more exact process called baselining.

Baselining (also called benchmarking) documents the performance of a network by measuring its capacity and standard operating efficiency. These measurements can identify long-term trends in network operations and their impact on network performance.

Baselining can be used with traffic estimation numbers or as an alternative  to  estimates since  you  are capturing  what  is really happening  on   the  network.  Taking   baseline

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measurements requires special monitoring equipment and applications. Because both of these are expensive, many small companies skip this step and  rely on estimates  alone. However, whenever possible, it is best to use both estimating and baselining

Tools for Testing Activity

If you have an existing LAN, you can probably get detailed reports from  the  network operating  system  (NOS)  vendor  as  to  the theoretical capacity of the NOS. Many NOSs run these reports as Value Added Processes (VAPs).

Another traffic measurement tool is a protocol analyzer or packet capture program such as Network General Sniffer®, Hewlett-Pack- ard’s LAN Advisor, Novell’s LANalyzer or the free WireShark/ Ethereal  package. Each  software  package  records  traffic  over a  given period  of  time. You can  also  purchase  LAN software emulation  packages  that  monitor networks  from a  PC.  These packages provide tools for:

•    Network mapping

•    Physical network management

•    Network design

•    Network planning and simulation

Design and Modeling Tools

Design tools model the behavior of a LAN under a given load. They provide an accurate picture of a LAN’s performance, given a certain number of users, applications, and telecommunications links. Some tools include application profiles that estimate traffic generated by specific applications. They may also have user librar- ies that contain performance profiles for various pieces of equip- ment, such as bridges and routers. These profiles can be plugged into the model without doing a lot of research, and can provide a reasonable estimate of the device’s throughput and latency. Many networking products also have built-in capabilities to determine CPU utilization against network traffic. Purchasing or renting sepa- rate design tools is expensive. However, if you need an engineered network with high reliability, the cost of failure far outweighs that of the tool.

Simulation and Testing Tools

LAN traffic simulation packages (as well as Network General’s Sniffer®  and  Hewlett-Packard’s  LAN  Advisor)  can  generate actual  LAN test traffic. By varying  the size and frequency of the traffic, the effect on the LAN is measured. Progressive degradation of LAN performance can be gauged as a function of client activity using just a few PCs. Activity of each LAN device (server, bridges, routers, etc.) can be monitored to determine the delay within each component. One client can simulate many workstations.

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Network Capacity Baselining

A network baseline is a snapshot of activity and performance that can provide proactive insight about the performance of a network. It  is  a measurement  that  can be  taken periodically  over time, or  while  interesting network  activity  is  observed, such as high bandwidth utilization. A separate baseline should be run on each individual subnet, WAN link, and the network backbone, forming a collection of baselines for the entire network.

A baseline should not be taken at any specified regular interval. A baseline taken at the same specified interval or time of day has the potential of lending the same results over time. It is better to baseline each subnet of a network at random times throughout the normal business day.

The  following  pages describe steps  necessary  to  baseline a network. The  examples use Network  General’s Sniffer®, an anlaysis  tool  used  in the management of  local  and wide area networks. Examples include baseline screen shots for reference.

Packet Capture Functions

NG Sniffer® is a popular tool for measuring LAN activity. It is available in various forms, from a freestanding hardware unit to a software only tool that can be installed on a PC (laptop, other portable, or client/server platform). A device that runs Sniffer® software (dedicated hardware or PC) must be purchased with a proprietary NIC that is compatible with the network topology being analyzed.