IPTV Course

8/20/2019 IPTV Course

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Silicon-IPTV-Broadcast --11

Notes: Notes:

IPTV Broadcasting, Protocols

and Switching

IPTV Broadcasting, ProtocolsIPTV Broadcasting, Protocols

and Switchingand Switching




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© Copyright: All rights reserved. Not to be reproduced without prior written consent. Silicon-IPTV-Broadcast -2

Course ObjectivesCourse Objectives

When you have completed this course you will be able to

• Understand the equipment and software used to deliver IPTV and VoD services

•

Describe the architecture of a these modern TV services• Compare Cable, over-air terrestrial, satellite and Internet delivery systems• Appreciate the trend in the technologies

• Understand addressing schemes for IP network prefix configurations• Examine resilience for MAC/IP mappings for reliable redundancy switching

• Select the best routing and switching s trategy for server and delivery networks

• Analyze protocols used to carry multimedia and troubleshoot services problems

• Appreciate how multicast routing protocols function

• Specify requirements for firewall transit of video services• Compare how DiffServ, DSCP, RSVP, WFQ, MPLS and 802.1P/Q can provide quality

of service

• Select the most appropriate quality of service option




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Course MaterialsCourse Materials

• Course Notes — Copies of all slides and supplemental presentation material




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Course ContentsCourse Contents

• Chapter 1 Television Architecture and Evolution• Chapter 2 Broadband Distribution Systems• Chapter 3 IP Delivery of Multimedia Services• Chapter 4 Layer 2 Addressing

• Chapter 5 Layer 3 Addressing• Chapter 6 Routing

• Chapter 7 Multicasting• Chapter 8 Management of Devices With SNMP

• Chapter 9 Next Generation Network Technology

• Chapter 10 Customer Home Network• Chapter 11 Industry Trends

• Chapter 12 Summary and Evaluation




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Course ScheduleCourse Schedule

Each day, the course will follow this schedule:

Start class 9 a.m.

Morning break 10:15 a.m. (approximately)

Lunch Noon

Resume class 1 p.m.

Afternoon break(s) As needed

Adjourn 4:30 p.m.






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Course InstructorCourse Instructor

• Background and education

• Current position

• Experience




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Attendee IntroductionsAttendee Introductions

• Your name

• Organization name


• Experience in:- — Television Technology — Networking and LANs — Telecommunications Technology

• Expectations




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Television Architectures and

Evolution

Television Architectures andTelevision Architectures and

EvolutionEvolution

Chapter 1Chapter 1




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Chapter ObjectivesChapter Objectives

In this chapter we will

• Examine what the major TV systems in the world are

• Explore how the various systems have evolved

• Compare various system capabilities

• See how digital and analogue systems differ




Notes: Notes:


What is Television Today?

Analogue and Digital Compared

Delivery Systems: What are they

Chapter Summary

Television Architectures and EvolutionTelevision Architectures and Evolution




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Human VisionHuman Vision

• What we see as essentially white light is a band of energy

• Individual colours map on to particular wavelengths

• The eye can be fooled into seeing white by using 3 primary colours

• Other colours can be formed by mixing these in proportion

The light that lights up our world and allows us to see that world is solar energy inwhat is known as the visible region of the Spectrum. This visible region is a verynarrow segment of this spectrum extending from ~ 440nm in the extreme blue (nearultra violet) to ~ 690 nm in the red region--with green in the middle @ ~ 555 nm.

Human vision is such that what appears as white light is really composed ofweighted amounts of a continuum of so-called Black Body energy. Tungsten lamps

have a similar spectral distribution.Sodium, Mercury vapor, fluorescent (a variant of Mercury), etc., on the other hand,do not have this continuum of spectral energy, but are composed of several discretewavelengths in proportions that "fool" the eye.

Color cameras are designed to "see" three (overlapping) segments of this spectralcontinuum by the action of red, green and blue optical bandpass filters. Theencoded color signal from the camera does not convey any real wavelengthinformation relative to the original hue.






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Mixing ColoursMixing Colours

• Primary colours can be mixed in proportion to form white

The addition of colors in the correct proportion creates white; unlike paint whichdarkens, e.g., black is the addition of Yellow, Cyan and Magenta pigments. Yellowabsorbs all but yellow light so it in fact absorbs blue removing it from what we see.

In order to produce "White" light to the human observer there needs to be 11 %blue, 30 % red and 59% green (=100%). However, if you shifted, say the red lightsource to a longer wavelength, the white light would appear more toward cyan.

White balance could be restored by changing the three color's weights, i.e. otherthan the original 11, 30, 59 percent ratios.

Each phosphor is formulated as a compromise between its quantum efficiency anddesired hue or color. An example of this is the fact that red phosphor requires moreenergy to cause it to "appear" equally bright to the human observer. Evidence ofthis can be seen when a CRT is over driven, the first color to bloom, is red.

One point should be made: the human observer is very discriminating when itcomes to flesh or skin tones.




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The Colour PalletThe Colour Pallet

The luminance of the image seen will affect the perceived colour as well. Byadjusting the luminance, effectively the black to white level, at the same time aschanging the proportion of different proportions of red, green and blue light the fullrange of colours needed to produce a television picture can be formed.




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Forming Television Picture Colour Test PatternForming Television Picture Colour Test Pattern

In a test pattern different combinations of luminance level and colour mixes areused to provide the range of signals needed in a full picture. This allows flaws inthe systems caused by malfunctions or incorrect adjustment of signal levels to bedetected.




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PAL D1 test PatternPAL D1 test Pattern

On CRT displays it is difficult to maintain straight lines and focused colourmapping. Modern flat panel display systems are able to maintain this with lessdifficulty.




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WidescreenWidescreen

Early TV systems had square or near square aspect ratios because this made bestuse of broadly circular CRT display efficiency. Human vision is more letter-boxshape and 16x9 aspect rations.




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Digital Image Standards ComparedDigital Image Standards Compared

Improving the resolution and interlacing, displaying alternate lines in consecutiveframes, provide better picture quality. Interlacing delivers better movement qualitywith limited increase in transmission bandwidth and complexity.




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ResolutionsResolutions

Horizontal Vertical Pexils RGB Color Detail %Television:

NTSC 427 525 224,175 100/100/100

HDTV 1050 600 630,000 100/100/100

Computer:

VGA 640 480 307,200 100/100/100

SVGA 800 600 480,000 100/100/100

Camera:

One Mega 1280 960 1,228,800 25/50/25

Two Mega 1600 1200 1.920,000 25/50/25

Three Mega 2048 1536 3,145,728 25/50/

Resolution means picture sharpness and is measured in lines of horizontal resolution.If you looked through a window with a giant Venetian blind and could observe a distant ladder andcount 625 rungs on that ladder, then you could say you had a vertical resolution of 625 lines. If youcouldn't count the rungs, because they were fuzzy or blocked by the slats of the Venetian blind, youwould have less than 625 lines of vertical resolution. You could have someone bring the laddercloser and eventually you could count all the rungs. In reality we have 575 not 625 visible lines.It would seem that 575 scan lines would give you a verticalresolution of 575 discernable lines on our ladder. This is not really the case. If one scan linedisplayed one rung, the next scan line would need to show the space between the rungs, and thefollowing line would show the next rung in order for therungs on the ladder not to merge together. Put another way, if each scan line saw a rung, then theladder would look like it was made of solid rungs with no spaces. Thus, an image that goes "rung-space-rung-space" is defined as 4 lines of vertical resolution and it took four scan lines to do it.Thus, 575 scan lines can show only 288 actual rungs on the ladder, but still the TV industry still callsthe vertical resolution 625 lines!I have oversimplified. The vertical resolution available from 575 scan lines calculates to .7 x 575 =403 lines of resolution. Why the .7? Imagine for a moment that you lookedthrough your Venetian blind at the ladder and could see all the rungs inbetween the slats. Now if youmoved your head up just a little bit, all of the rungs would be hidden behind the slats and you wouldsee only the spaces between the rungs, erroneouslycoming to the conclusion that the ladder had no rungs.






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Comparative ResolutionsComparative Resolutions

Name Prog. Total Active Vert. Horz. Opt. Asp. freq.or lines lines res. res. view ratio MHzinter. dist.

HDTV p 1050 960 675 600 2.5 16/9 8USA,analogHDTV p 1250 1000 700 700 2.4 16/9 9Europe,analogHDTV NHK i 1125 1080 540 600 3.3 16/9 20

NTSC i 525 484 242 330 7 4/3 4.2conv.

NTSC prog. p 525 484 340 330 5 4/3 4.2

PAL i 625 575 290 425 6 4/3 5.5conv.PAL prog p 625 575 400 425 4.3 4/3 5.5

SECAM i 625 575 290 465 6 4/3 6conv.

SECAM p 625 575 400 465 4.3 4/3 6prog

The basic concept behind high-definition television is actually not to increase thedefinition per unit area ... but rather to increase the percentage of the visual fieldcontained by the image.

The majority of proposed analog and digital HDTV systems are working towardapproximately a 100% increase in the number of horizontal and vertical pixels.(Proposals are roughly 1 MB per frame with roughly 1000 lines by 1000 horizontal

points). This typically results in a factor of 2-3 improvement in the angle of thevertical and horizontal fields. The majority of HDTV proposals also change theaspect ratio to 16/9 from 4/3 -- making the image more "movie-like".

The following table summarizes a few of the more conventional analogue HDTVproposals in comparison with existing TV system.

The aspect ratio of the picture is defined to be the ratio of the picture width W to itsheight H. The optimal viewing distance (expressed in picture heights, H) is thedistance at which the eye can just perceive the detail elements in the picture.




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Chapter Summary





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Why Digital?Why Digital?

• Human eyes are analogue sensors and our ears hear analogue sounds

• Both eyes and ears have a wide dynamic range

— We can see in almost total darkness yet also in bright sunshine• But

• To produce TV that matches this quality takes very high frequencies

• We are limited by noise — Analogue signals can take any value so signal and noise look similar — Digital signals take discrete values (0 or 1) small variations can be removed — Similar quality in less bits with digital signals — Computers can compress more cheaply

Analogue Digital

To transform a signal from analogue to digital, the analogue signal must go throughthe processes of sampling and quantization. The better the sampling andquantization, the better the digital image will represent the analogue image.

Sampling is how often a device (like an analogue-to-digital converter) samples asignal. This is usually given in a figure like 48 kHz for audio and 13.5 MHz forvideo. It is usually at least twice the highest analogue signal frequency (known as

the Nyquist criteria). The official sampling standard for standard definitiontelevision is ITU-R 601 (short for ITU-R BT.601-2, also known as "601"). Fortelevision pictures, eight or 10-bits are normally used; for sound, 16 or 20-bits arecommon, and 24-bits are being introduced. The ITU-R 601 standard defines thesampling of video components based on 13.5 MHz, and AES/EBU definessampling of 44.1 and 48 kHz for audio. Quantization can occur either before orafter the signal has been sampled, but usually after. It is how many levels (bits persample) the analogue signal will have to force itself into. As noted earlier, a 10-bitsignal has more levels (resolution) than an 8-bit signal.




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Digital SamplingDigital Sampling

• For picture quality to be maintained we must sample often enough

• Nyquist proved (in 1929) that we must sample at least twice the highest

frequency — To obtain audio with 20 kHz signal we sample at 44,100 samples persecond

— We may sample the video at 14 MHz• A full bandwidth digitally sampled PAL signal takes about 160 Mbit/s

— This is impractical for transmission but contains lots of redundancy

Ratios such as 4:2:2 and 4:1:1 are an accepted part of the jargon of digital video, a shorthand takenfor granted and sometimes not adequately explained. With single-channel, composite signals such asNTSC and PAL, digital sampling rates are synchronized at either two, three, or four times thesubcarrier frequency. The shorthand for these rates is 2fsc, 3fsc, and 4fsc, respectively. With three-channel, component signals, the sampling shorthand becomes a ratio. The first number usually refersto the sampling rate used for the luminance signal, while the second and third numbers refer to therates for the red and blue color-difference signals. A 14:7:7 system would be one in which awideband luminance signal is sampled at 14 MHz and the narrower bandwidth color-differencesignals are each sampled at 7 MHz. As work on component digital systems evolved, the shorthand

changed. At first, 4:2:2 referred to sampling luminance at 4fsc (about 14.3 MHz for NTSC) andcolor-difference at half that rate, or 2fsc. Sampling schemes based on multiples of NTSC or PALsubcarrier frequency were soon abandoned in favor of a single sampling standard for both 525- and625-line component systems. Nevertheless, the 4:2:2 shorthand remained. In current usage, "4"usually represents the internationally agreed upon sampling frequency of 13.5 MHz. Other numbersrepresent corresponding fractions of that frequency. A 4:1:1 ratio describes a system with luminancesampled at 13.5 MHz and color-difference signals sampled at 3.375 MHz. A 4:4:4:4 ratio describesequal sampling rates for luminance and color difference channels as well as a fourth, alpha keysignal channel. A 2:1:1 ratio describes a narrowband system that might be suitable for consumer useand so on.Unlike 4:1:1, however, the samples in 525 line systems don't come from the same line as luminance,but are averaged from two adjacent lines in the field. The idea was to provide a more even andaveraged distribution of the reduced color information over the picture.




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CompressionCompression

• Compression is possible once we are in the digital domain

• Video pictures are inherently full of redundancy if we have storage

— In the majority of cases the next frame is largely the same as the last — By sending just the differences we can reduce bandwidth• Methods used today are dominated by Motion Picture Experts Group

Some people say that compressing video is a little like making orange juice concentrate or freeze-dried back-packing food. You throw something away (like water) that you think you can replacelater. In doing so, you gain significant advantages in storage and transportation and you accept thefood-like result because it's priced right and good enough for the application. Unfortunately, whileorange juice molecules are all the same, the pixels used in digital video might all be different. Videocompression is more like an ad that used to appear in the subway which said something like: "If u cnrd ths, u cn get a gd pying jb" or the kind of language used in SMS text messages.

The real difference is perhaps the scale of the compression in that we can now deliver a viablepicture in about 2% of the bandwidth of the original. A 2 Mbit/s video stream replacing a 166 Mbit/soriginal. The price we pay is quality. The notion of quality in any medium is inherently a movingtarget. We've added color and stereo sound to television. Just as we start to get a handle oncompressing standard definition signals, high definition and widescreen loom on the horizon. Therewill never be enough bandwidth. There is even a Super High Definition format that is 2048x2048pixels--14 times as large as NTSC.

Perhaps former Tektronix design engineer Bruce Penny countered the quip best when he said,"Compression does improve picture quality. It improves the picture you can achieve in thebandwidth you have.”




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Chapter Summary





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Television Broadcasting IndustryTelevision Broadcasting Industry

ProgrammeProduction

FilmNews

TV Production

Content

Channels

EntertainmentGovernment and Politics

ReligionEducation

Community

Distribution

Marketing and Delivery

Over-the-airCable

SatelliteInternet and IP

Delivery

Community antenna television (now called cable television) was started by JohnWalson and Margaret Walson in the spring of 1948. The Service Electric Companywas formed by the Walsons in the mid 1940s to sell, install, and repair GeneralElectric appliances in the Mahanoy City, Pennsylvania area. In 1947, the Walsonalso began selling television sets. However, Mahanoy City residents had problemsreceiving the three nearby Philadelphia network stations with local antennasbecause of the region's surrounding mountains. John Walson erected an antenna ona utility pole on a local mountain top that enabled him to demonstrate thetelevisions with good broadcasts coming from the three Philadelphia stations.Walson connected the mountain antennae to his appliance store via a cable andmodified signal boosters. In June of 1948, John Walson connected the mountainantennae to both his store and several of his customers' homes that were locatedalong the cable path, starting the nation’s first CATV system.John Walson has been recognized by the U.S. Congress and the National CableTelevision Association as the founder of the cable television industry. John Walsonwas also the first cable operator to use microwave to import distant televisionstations, the first to use coaxial cable for improved picture quality, and the first todistribute pay television programming (HBO)




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Architecture of Cable TV DistributionArchitecture of Cable TV Distribution

ProgrammeProduction

FilmNews

TV Production

Content

Channels

EntertainmentGovernment and Politics

ReligionEducation

Community

Distribution

The Head End: The control center of a cable television system. The headend receives incomingsignals from satellites, television antennas and locally produced programs and amplifies, converts,processes, combines and transmits the signals through a cable network to subscribers. The headendincludes antennas, preamplifiers, frequency converters, demodulators, modulators, processors,scrambling and descrambling equipment.

The uplink sends programming signals to satellites to be relayed back to earth. Cable programmershave large uplinks, which are more powerful than, but similar to earth stations.

Earth Stations receive satellite signals. This parabolic antenna is also known as a TVRO (TelevisionReceive Only) antenna. A number of earth stations are located at the cable system to receiveprogramming from dozens of services like MTV, ESPN and HBO. Also called "dishes" because ofits shape, earth stations can be 15 meters or more in diameter, or as small as 18 inches. Millions ofindividuals and businesses also own dishes to receive programming directly from satellites.

A network of coaxial cable and fiber optic cable used by cable providers to deliver programming tocustomers. A broadband cable system is capable of delivering analog and digital communicationsignals. The first segment, the trunk line system, connects the headend to the first bridgingamplifiers or fiber optic nodes. Trunk lines can also include power supplies and other electroniccomponents. The next segment, the feeder system, carries signals to individual neighborhoods. Thelast segment, the drop line part of the network, is coaxial cable which connects individual subscriberlocations to the feeder trunk.




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Cable Distribution SystemCable Distribution System

In a modern cable network other non-TV services might be added. In particularInternet access via cable modems within the set-top box or directly connected to it.By adding two way data access services independently of telephone networks thecable operator can both add new data services and uses the internetworkingcapability for telemetry control of programme access.

The industry trend is towards greater and greater use of IP transport of both

programmes and control services. Throughout the TV industry there is a transitiontowards IP taking place. This is moving at such a pace that many industry expertsexpect the majority of YV channels to be distributed over IP transports as theirprimary method by 2007.




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Expanded Television ServicesExpanded Television Services

• Expanded services are those that go beyond the distribution of TVprograms

•

Provision of Telephony services• Information services

• Internet access

• Interactive Gaming

In the end users do not make use of raw communications capacity but use services.The diversity of services now available have increased well beyond just TV.




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Telephony ServicesTelephony Services

• Telephony is - or was - a high value service — Since 2001 there has been

a reduction in voice prices — In 2004 UK fixed line voice

revenues fell more than 25%• Cable operators can add this service

• Easy additional revenue generation

• Regulation is the biggest hurdle

• Competition now with other Internet access

• TV over phone lines is the next technology

At the time of relatively high telephony charges during the 1980s and 1990s theopportunity to add telephony to cable TV networks provided and opportunity foradditional revenues for cable TV providers. Analogue cable networks were almostentirely unidirectional because the line amplifiers worked in one direction only.Building digital networks that have bidirectional capability, even if at differentspeed deliver greater flexibility.




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Cable ModemsCable Modems

• Internet access can be provided via cable modems

• Early broadband access via cable offered 500 kbit/s services

• Lower initial price than ADSL broadband

• Extended ADSL services at 1 Mbit/s, 2 Mbit/s and up to 4 Mbit/s — These are likely to be difficult for cable to match

• VDSL at 10 Mbit/s and eventually up to 50 Mbit/s may replace cable — TV over IP is feasible along with all services in the long term

Once networks were bidirectional it became feasible to carry data. Normally this isused for access to the Internet. By using more bandwidth from network to user thanin the reverse direction paterns of operation better match normal service use.




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Information ServicesInformation Services

• All TV distribution systems must provide information on programmes

• The same technology can provide information on other things

• May be possible to bill for some information — Sports results — Ticket bookings — Travel — Advertising

In the end all services can be viewed in one way or another as information.




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Interactive GamingInteractive Gaming

• Interactive gaming takes 3 major forms

• Gambling

— Event betting — Interactive poker and other games of chance — Lottery

• Games played via dedicated head-end servers — Trivia quizzes played for entertainment — Arcade games using set-top box processing — Games uploaded into special gaming consoles

• Peer-to-peer group gaming — Interconnected networked games from PC or gaming consoles

– e. g. Network quake

On the early commercial Internet only services were found to be quickly profitable– sex and gambling. While these continue to be in demand interactive gaming hasprogressed beyond just gambling into areas of network entertainment. Somesectors of the market believe that this area will become the most important oncetelevisions evolve into Internet attached media centres with lots of processingpower.




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Chapter Summary





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Chapter SummaryChapter Summary

In this chapter, we have

• Examined what the major TV systems in the world are

• Explored how the various systems have evolved

• Compared Various system capabilities

• Seen how digital and analogue systems differ




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Broadcast Distribution SystemsBroadcast Distribution SystemsBroadcast Distribution Systems

Chapter 2Chapter 2




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When you have completed this chapter you have learned how to

• Examine component parts of a TV distribution networks

• Explore how the various systems options

• Identify the key interfaces

• Predict how the technology will evolve in the near future

• Examine the encoding and compression standards






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Components of a Cable TV SystemComponents of a Cable TV System

Interf ace to programme,channel and contentsuppliers

Headend: Control,switching, encoding andmanagement Fiber and coax

cable distribution

Set-top box forconditional access,interfacingand decoding

Around the globe, cable TV operators are investing to upgrade their networks in order to offeradditional TV channels and two-way interactive services such as high-speed Internet access andtelephony. The main issues are:- How can these upgrades be designed to maximize bandwidth,reliability, quality and flexibility while remaining cost-effective? - How can the resulting platformremain as open to future expansion as possible?- What needs to be done in order to support further expansion into promising new markets, such asbusiness voice and data services?

Up to now, the large majority of subscribers are offered two basic types of services from their localcable TV company. For a fixed monthly fee, the cable TV company provided a few dozen TVchannels that could be viewed "in the clear", which means directly on any standard TV set. This iscalled the "basic tier". Subscribers can also elect to pay additional fees to get access to "premium"channels. The premium channels require the use of a set-top decoder in order to be descrambled.

From a network infrastructure standpoint, cable TV is delivered via an analog broadband distributionplant based on coaxial cable for end delivery to the subscriber and optical fiber for distribution. Thetransmission capacity of the network ranges between 330 and 860 MHz, with most modern plantsoperating at 550 MHz. This type of network architecture is by far the most widely used by cable TVoperators and is called the Hybrid Fiber Coax (HFC) network.




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Traditional Cable TV Head End ComponentsTraditional Cable TV Head End Components

The "headend" is the primary facility of any cable network. The headend's function is to collect allthe basic and premium TV channels and combine them for delivery to subscribers over a single coaxcable. TV channels are collected in three ways: using standard TV antennas to pick up signals "off-air" the same way any TV set can pick them up, via satellite dish, or via direct fiber feed from localTV affiliates to maximize reception quality. Premium channels are also scrambled to preventunauthorized viewing. The combined broadband signal is then sent to subscribers via the HFCnetwork. Most HFC networks are designed so that optical fiber is deployed to pockets of around 500homes, then converted to coax cable for delivery to the home. Along the way, the signal will be splitand re-amplified several times using a "tree-and-branch" topology. Premium channel subscribers are

provided with a special unit called a TV set-top converter to descramble the premium channels towhich they have subscribed. Some premium channels are also controlled on a "pay-per-view" basis,where each particular broadcast on the channel is charged to the subscriber.Each individual TV channel is received using specific equipment. For satellite-fed channels, an"Integrated Receiver Decoder" (IRD) is used to convert the signal to its baseband NTSC or PALform. At this point, the signal could be viewed on a TV set as NTSC/PAL is the standard signal thatyour TV set receives. Similarly, TV channels that are received "off-air" via an antenna aredemodulated from their original carrier frequency and converted to NTSC/PAL by a RadioFrequency (RF) demodulator. All signals belonging to "premium" service tiers (mostly satellite-fed)are fed to a "scrambler" unit which encodes the signal to prevent its unauthorized viewing.Finally,each signal is fed into a bank of RF modulators where they are assigned a specific channel slot. Theresulting modulated signals are fed into a passive RF combiner, which multiplexes all modulatedsignals together into a single broadband 330-to-860 MHz signal. This signal is then converted tooptical and fed to subscribers via the HFC network.




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Enhanced Cable TV Network ServicesEnhanced Cable TV Network Services

The HFC TV plant described above poses two limitations to the modern-day cable TV operator.First, it can carry only up to 80 TV signals. The ability to carry more channels can providesubstantial additional revenues by enabling the offering of additional premium TV channelpackages. Second, bandwidth constraints limit the capability to serve the seemingly insatiabledemand for high-speed Internet access, which promises even greater revenue growth. Cable's veryhigh bandwidth can offer access speeds measured in megabits per second, or about 1000 times thespeed of ordinary telephone modems. Once upgraded for high-speed Internet access, the cable TVnetwork will also be able to carry telephone conversations, providing yet-another very significantrevenue increase potential.

In order to support more TV channels as well as high-speed Internet access and telephone services,the cable TV headend needs to be upgraded. At the headend, links to the mainstream telco networkare required in order to support two-way Internet and voice services. These are provisioned usingstandard 34 Mpbs or 140 Mbit/s feeds. At the home, a new unit called the "cable modem" will bedeployed to those subscribers that have ordered the provider's voice and/or Internet services. Thisunit will make the link between the coax cable plant and the subscriber's PC and/or telephone set.The technique used to provide for more TV channels is digital compression, which typically yields afivefold increase in capacity. To that end, compressed TV signals are received via satellite receiversor local cable feeds and are converted to analog using advanced Quadrature-Amplitude Modulation(QAM) techniques and then fed into the cable plant via standard RF modulators. At the subscriber'shome, a special digital TV set-top decompresses the signal, converts it to analog baseband and feedsit to the TV set.




Notes: Notes:


Enhanced Head EndEnhanced Head End

ExtraChann els

NetworkAccess

Internet

The process of sending and receiving Internet data via the cable plant uses QAMdigital modulation techniques as well. In the headend, Internet data received fromthe backbone via the telco network is fed to a standard TCP/IP router. This data isthen converted to analog using 'cable modems', which use QAM modulation toconvert the Internet data into an analog signal. This signal is then fed to the cableplant. At the home, the signal is received by a special 'cable modem', which ishooked to the coax cable on one side and to the subscriber's computer via Etherneton the other side. Speeds can reach around 30 Mbps 'downstream', that is from thebackbone towards the subscriber, and anywhere from 128 Kbps to 2 Mbps'upstream', or from the subscriber towards the backbone. Such modems are called'asymmetrical', since unlike standard telephone modems, their upstream anddownstream bandwidths are different. Most cable modems on the market are fullyinteroperable between various manufacturers and comply to the MCNS-DOCSISstandard published by CableLabs, the cable industry's standardization body.

The same technique can support the deployment of telephony via cable, and in factmost cable modem units also sport a standard telephone jack to be connected to thesubscriber's phone set.




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Multiple Cable OperatorsMultiple Cable Operators

Most companies are referred to as Multiple Systems Operators (MSOs), since they operate dozens,sometimes hundreds of cable systems.

Each individual cable TV distribution system is equipped with its own headend. Typically, a cableTV headend can serve between 20,000 and 60,000 subscribers. This means that a large metropolitanarea would normally count between 5 to 15 independent cable TV systems. As each one of thesesystems is upgraded for digitally-delivered video, voice and data, it needs to upgrade the distributionplant to two-way, install the related equipment, and establish a local connection to the Internet viafacilities leased from the telco network.

This deployment approach, while simple to implement, presents several issues to the MSO. First,each individual headend needs its own set of Internet connection equipment, as well as its ownconnection to the Internet. The same is true for all equipment and connections required for thedeployment of additional channels via compressed digital video feeds. There is no way to shareInternet access bandwidth between the various headends, as none of the connections are shared.There is no mechanism in place to provide centralized management, which implies that eachindividual headend needs to be managed independently. Finally, no mechanism exists to provide forredundancy within a given headend.

In short, the MSO operates its cable TV network as a collection of isolated islands, with no realvalue-added derived from any kind of interconnection and complete duplication for capital,operating and maintenance costs.




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Interconnected Head EndsInterconnected Head Ends

Standards-based optical fiber networks offer a much more compelling strategy for upgrading cableTV systems. The basic idea behind Regional Cable TV Headend Interconnection (RCHI) networksis that instead of developing independent islands, one headend in the network will serve as a primaryhub to feed all the others.

One headend is designated as the 'main' hub, and the others serve as 'remote' headends. In theexample above, EastBurg serves as main, while CenterVille and NorthTown serve as remotes. Allheadends in the RCHI network are linked together using a 2.4 Gbps SONET/SDH OC48/STM16digital fiber ring.

SONET/SDH is the worldwide standard used by all telecommunications carriers in order to buildinteroperable fiber networks between central offices. In fact, SONET and SDH are similar standardsused in different parts of the world, where SONET is used in North America, SDH is used in Europeand Latin America, and both being used in Asia. It is fair to say that all the fiber used by today'stelco carriers carries video, voice and data according to the SONET/SDH standard, which issupported by dozens of equipment manufacturers on a completely interoperable basis. The ringarchitecture used by SONET/SDH provides complete protection against fiber cuts, which cause over85% of network failures according to a recent NPRS study. SONET/SDH dictates that any fiber cuton the network will be result in traffic being rerouted in less than 50 milliseconds.






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Head End Signal ReceptionHead End Signal Reception

• Digital Satellite Receiver

• Encrypted and Direct Video Broadcast (DVB) modes of operation

• 3 to 40 Mbit/s operation

• Advanced Serial Interface (ASI) input and output — Most advanced units now support Gigabit Ethernet instead

Inputs to the Headend will come from satellite feeds from programme makers andchannel feeds. Modern satellite receiver series employ the latest in MPEG-2/DVBdigital technology.

Exceptional end-user reception and signal quality is achieved by using robustQPSK satellite demodulation, forward error correction, and MPEG-2decompression circuitry, all housed within a professional rack-mountable chassis.

They process Standard Definition transport streams, including, encrypted signalsand unencrypted DVB signals. The latest include the ability to process HighDefinition (HD) transport streams, via an ASI output, for external HD decoding.With additional key features such as Video Broadcast Interface (VBI) reinsertion.




Notes: Notes:


Encoding and Trans-codingEncoding and Trans-coding

• An important part of the Headend function is encoding TV signals

• Feeds may arrive in one satellite modulation format and be re-coded to

another for more efficient onward transmission• NTSC feeds may be converted to PAL

• Encoding of analogue to MPEG-2 or even MPEG-4 may be required

• The selection of the vendor for headend equipment is often based uponthe quality of such codecs and trans-coding

The Integrated Receiver Transcoder (IRT) receives a modulated QPSK carrier andtranscodes it into a more bandwidth efficient 64 QAM format. The unit accepts LBand input from a satellite downconverter and produces a signal appropriate totransmission in a 6 MHz television RF channel.The IRT decrypts and performs Forward Error Correction (FEC) on the incomingsatellite data stream. It then clears information streams not intended for local cabletransmission and inserts new information streams for this purpose. It prepares the

signal for transmission across the terrestrial cable system by re-encryptingprograms under local headend control. IRTs are linked via an Ethernet connectionin a local headend LAN.The IRT provides local generation and insertion of program specific data, includingtier level, purchaseability, price and rating codes. The unit can also be controlledvia a management system. IRTs may be optionally daisy chainedtogether via themultidrop port and controlled remotely over the satellite link where no Ethernetconnectivity exists. The IRT also provides an expansion interface port so thatexternal devices can be cascaded to allow for processing beyond the capacity of asingle IRT.




Notes: Notes:


Video RouterVideo Router

• Acts as a switch between video feeds and output to cable

• Requires enough inputs for all channels and backups

• Sizes up to 1024 x 1024 possible

• May support redundant operation

• ASI interfaces are common — Latest systems may use Gigabit Ethernet

• Conforms to SMPTE 291M or 292

Channels and feeds must be switched from input of the head end via transcoding tothe correct format for distributions and then on to the distribution network.

In the real world equipment fails from time to time and so redundancy provision isalso required. This all demands a switch at the core of the headend capable ofinterconnecting, and switching all the feeds. This unit is called a video router.

With the migration from ASI digital feeds to IP this component will become a

gigabit switch carrying video feeds over IP. While technically possible, only thelatest state-of-the-art systems are yet all IP. However during 2005 it is expectedthat several large systems will migrate in this direction. The whole cable TVindustry is moving in the IP direction and so too will the routers.

In the terminology of the Internet a “switch” has special hardware assistance toundertake high speed switching, while a router works at layer 3 of the OSI modeland may have slower software store and forward control. These units in reality willbe switches no routers, but often are formed from multi-layer switches. These notonly have hardware to speed up the switching but also extensive software controlfor the flexibility of Internet protocols for streaming and security.




Notes: Notes:


Control SystemsControl Systems

• Headend equipment must be controlled

• Older systems use illuminated buttons

• Latest units based on Windows PCs — Easy-to-use graphical user interfaces to configure equipment — Control conditional access and MPEG encoding rates — Broadcast equipment and receivers — Easy ‘drag and drop’ grouping feature for your receiver base — Graphical user interface to schedule receiver control and conditional access

parameters on an — Immediate, one time, daily or weekly basis — On-line help — Password protection on user interfaces — Full redundancy and back-up options — Remote access of head-end control station

European companies currently lead the world in TV control systems. TANDBERGhas a complete range of management system for both small and large MPEG-2broadcast head-ends for configuration, system monitoring and redundancy. Ideallysuited to controlling and monitoring satellite, cable and terrestrial super head-ends,especially where n+m multiplexing is required to save costs. Powerful re-multiplexing capabilities make it perfect for digital turnaround applications. Costeffective device only control is available for the simpler regional head-end.These have recently been installed in the largest cable systems in the world andcontinue to dominate the control of state of the art headend control.

The latest generation systems introduced in 2005 have the capability to work usingall IP services. While the channel and studio side has been IP enabled on manysystems for a year or so, now even distribution can be based on IP. The first All IPsystem deploying MP4 encoding for HDTV was installed in Europe during 2004.This is likely to spread throughout the whole industry over the next few years.




Notes: Notes:


Traditional Coaxial Cable Distribution ComponentsTraditional Coaxial Cable Distribution Components

Line AmplifiersAttenuator

Splitter/Combiner Tap

DistributionCable

RF cables are designed to carry RF signals from one point to another, not from one point to many. Inother words, you can't run RF signals to multiple locations by wiring all the destinations in parallel.The reason is that the residential RF distribution scheme is based on 75 ohm terminatedtransmissions. Meaning that the transmitting side expects to see one, and only one, 75 ohm load onthe other end of the cable.A splitter is a small device that has one input (the 75 ohm load) and 2 or more outputs, each drivinga separate 75 ohm load. Essentially they are transformers that split the power in the input signal tomultiple outputs, while maintaining the 75 ohm impedance. However, there is no free lunch! Everytime you split an RF signal with a splitter, you drastically decrease the signal's strength. An RFsignal only has so much power. Logic dictates that splitting this signal in two with a "passive" devicewill result in two signals that each have--at most--half of the original signal's strength.A combiner is simply a splitter hooked up backwards. It combines the channels on two or moreseparate cables onto one cable. The only drawback to this piece of magic, is that the cables beingcombined cannot have any channels in common with each other. The resulting signal on that channelwould be trashed.Taps are similar to splitters, but are "wound crooked" so that the outputs are not equal in signalstrength. The "through" output of a tap may only reduce the signal level by a very small amount,while the "tap" output is a small fraction of the signal level. Taps are primarily used in complexcommercial distribution installations.Attenuators are simple "one in, one out" devices that reduce the signal strength. Attenuators come invarious sizes and are useful when tuning up the video distribution system.




Notes: Notes:


Designing a Distribution SystemDesigning a Distribution System

• The goal of design is to deliver good signal levels to each consumer

• Cables, taps, splitters and combiners cause loss

• Amplifiers increase signals but also add noise

• Signal to noise ratio limits demodulation and thus the size of system

Device Loss (-dBmV)

2-Way Splitter/Combiner 4.0




100 ft RG6 4.0

The RF signal looses strength as it passes down the cable and through combiners and splitters. Tocounter this loss (or "attenuation") we use RF amplifiers. In the ideal RF distribution system, thesignal level at each wall-plate should be about the same as the signal level coming in from the cableTV system or antenna. This ideal is called "unity gain." By applying a little math, and the tablebelow, you can calculate the approximate losses and gains in your system to approach this goal.RF signal levels are measured in dBmV which is a logarithmic scale of signal relative to onemillivolt. Since decibel values represent power levels, and are logarithmic, they can be calculatedwith simple addition and subtraction. The main thing to remember about dB (for short) values is thatif the level drops below 0 dB (into the negative dB range), you are loosing actual signal informationand no amount of amplification will be able to recover this lost information (picture quality.) In fact,amplifying a signal that is below 0 dB will usually make the picture worse since the noise is nowbeing amplified and picked up. So you must insure that your signal levels never drop dangerouslynear 0 dB anywhere in your distribution system. This is why the main RF amplifier us usuallyconnected near the input side of the distribution system; so the signal is boosted early, and neverdrops precariously low. The only way to actually measure the signal level is with an RF signal levelmeter specifically designed for this task. We ended up buying one (they go for $1000 up) that werent out to our local customers that are having trouble tuning up their very complex systems. Butmost folks get by just fine by just doing the calculations up front. Cable TV companies are supposedto deliver around 15 dB of signal strength at the side of the house, but I've seen this range frombelow 0 to well over 25 dB. An antenna can deliver a wide range of signal strengths depending onthe strength and distance of the stations.The optimum level at the wall-plate is between 8 and 15 dB




Notes: Notes:


Signal TransmissionSignal Transmission

• Higher frequencies suffer more loss over coaxial calbes

• This leads us to shift distribution from UHF down to VHF

• The set top box can reverse the shift and deliver channels on their originalfrequency

• Better performance can be obtained from digital coax and fiber

The signals provided in the cable cover a range of frequencies from 54-88 MHz (VHF/low channels2 to 6), 88-108 MHz (FM radio), 174-216 MHz (VHF/high channels 7 to 13), to 470-806 MHz(UHF channels 14 to 69). Because cable doesn't carry actual UHF frequencies very efficiently (100feet of RG-59 loses 80-90 percent of UHF), the UHF channels are converted by your cablevisioncompany to a set of lower frequencies. This is why you need a converter box, or a "cable-ready" TVset.

Whenever the signal is split, it becomes half as strong. It isn't like the three-way outlet of anextension cord where all the appliances receive the same voltage, as they would if connected directlyto the wall. It's more like a farmer irrigating a crop by dividing a stream of water, every time it issplit in two there is only half as much water.

Connections from the splitter to wall outlets in your home are made with RG-59 coaxial cable.Putting the F-fittings on the ends of the cable is not difficult, but if you don't want to do this, just buylengths of cable with the fittings already attached, and coil up any excess cable or stuff it into a wallcavity. The excess length may have a slight loss, but since it has been amplified anyway it won'tmake any noticeable difference.

Unused outlets (outlets which are not connected to TV sets) used to require terminating resistors toprevent reflection of signals. This is something you might try if you find poor reception on only oneor two channels using an older amplifier. The resistors are designed to plug directly into the unusedoutlets. Today you can find amplifiers that don't require terminators.




Notes: Notes:


Migration to Digital Fiber SystemsMigration to Digital Fiber Systems

• Optical systems also depend upon loss levels

• Digital regeneration removes noise

• Digital services can be delivered over larger area

• More consistent quality is possible Digital Fiber OpticTransmitter

In the USA the The FCC has set the year 2006 as the deadline for broadcasters to switch fromstandard definition television (SDTV) to digital television (DTV) and high definition television(HDTV). Among the many advantages of this transition, transmission distance and repeaters (signalregenerators) do not affect the quality of digitized video. A visit to any major broadcast industrytrade show, such as those sponsored by the National Association of Broadcasters (NAB) or Societyof Motion Pictures and Television Engineers (SMPTE), reveals that cameras, tape decks, mixingboards, matrix switches, effects boxes, etc. operate the digital format.

Fiber optics plays a big part in the move to the new television standards, providing the only viablemeans of signal transport by offering the bandwidth required for these television standards.Currently, analogue video signals can be carried over relatively long lengths of coax cable. With abandwidth of only 4.5 MHz, analogue signals do not tax the limited bandwidth of coax cable, buteven so, coax cable introduces a great deal of frequency dependent distortion requiring anequalization network. A digitized video signal's increased bandwidth usurps coax's ability to carrythe new signal.

A standard NTSC video signal typically requires a serial bit rate of 143.2 Mb/s. By contrast, high-end HDTV standards require serial bit rates of 1,485 Mb/s. Coax cable can carry such high-speeddigital data streams short distances, typically 300-600 meters for NTSC and 30-60 meters forHDTV. Fiber optics, on the other hand, can easily carry the full range of digital signals up to tens ofthousands of meters. Figure 1 shows a typical digital fiber optic video transmitter.




Notes: Notes:


Fiber Optic TransmissionFiber Optic Transmission

Some 10 billion digital bits can be transmitted per second along an optical fiber link in a commercialnetwork, enough to carry tens of thousands of telephone calls. Hair-thin fibers consist of twoconcentric layers of high-purity silica glass the core and the cladding, which are enclosed by aprotective sheath. Light rays modulated into digital pulses with a laser or a light-emitting diodemove along the core without penetrating the cladding.The light stays confined to the core because the cladding has a lower refractive index—a measure ofits ability to bend light. Refinements in optical fibers, along with the development of new lasers anddiodes, may one day allow commercial fiber-optic networks to carry trillions of bits of data persecond.Total internal refection confines light within optical fibers (similar to looking down a mirror made inthe shape of a long paper towel tube). Because the cladding has a lower refractive index, light raysreflect back into the core if they encounter the cladding at a shallow angle (red lines). A ray thatexceeds a certain "critical" angle escapes from the fiber (yellow line).STEP-INDEX MULTIMODE FIBER has a large core, up to 100 microns in diameter. As a result,some of the light rays that make up the digital pulse may travel a direct route, whereas others zigzagas they bounce off the cladding. These alternative pathways cause the different groupings of lightrays, referred to as modes, to arrive separately at a receiving point. The pulse, an aggregate ofdifferent modes, begins to spread out, losing its well-defined shape. The need to leave spacingbetween pulses to prevent overlapping limits bandwidth that is, the amount of information that canbe sent. Consequently, this type of fiber is best suited for transmission over short distances, in anendoscope, for instance.




Notes: Notes:


Fiber TypesFiber Types

Graded Index Multimode Fiber Single Mode Fiber

Step Index Multimode Fiber

GRADED-INDEX MULTIMODE FIBER contains a core in which the refractiveindex diminishes gradually from the center axis out toward the cladding. The higherrefractive index at the center makes the light rays moving down the axis advancemore slowly than those near the cladding. Also, rather than zigzagging off thecladding, light in the core curves helically because of the graded index, reducing itstravel distance. The shortened path and the higher speed allow light at the peripheryto arrive at a receiver at about the same time as the slow but straight rays in the coreaxis. The result: a digital pulse suffers less dispersion.SINGLE-MODE FIBER has a narrow core (eight microns or less), and the index ofrefraction between the core and the cladding changes less than it does formultimode fibers. Light thus travels parallel to the axis, creating little pulsedispersion. Telephone and cable television networks install millions of kilometersof this fiber every year.




Notes: Notes:


Fiber ConnectorsFiber Connectors

There is now a wide rance of connectors supported in the industry for fiber cables.ST and SC connectors are among the most well established within the industry.




Notes: Notes:


Fiber CablesFiber Cables

Indoor/Outdoor Tight Buffer

Indoor/Outdoor Breakout Cable

Aerial Cable/Self-Supporting

Armored Cable

Indoor/Outdoor Tight Buffer

FIS now offers indoor/outdoor rated tight buffer cables in Riser and Plenum rated versions. Thesecables are flexible, easy to handle and simple to install. Since they do not use gel, the connectors canbe terminated directly onto the fiber without difficult to use breakout kits. This provides an easy andoverall less expensive installation. (Temperature rating -40ºC to +85ºC).

Indoor/Outdoor Breakout Cable

FIS indoor/outdoor rated breakout style cables are easy to install and simple to terminate without the

need for fanout kits. These rugged and durable cables are OFNR rated so they can be used indoors,while also having a -40c to +85c operating temperature range and the benefits of fungus, water andUV protection making them perfect for outdoor applications. They come standard with 2.5mm subunits and they are available in plenum rated versions.

Aerial Cable/Self-Supporting

Aerial cable provides ease of installation and reduces time and cost. Figure 8 cable can easily beseparated between the fiber and the messenger. Temperature range ( -55ºC to +85ºC)

Armored Cable

Armored cable can be used for rodent protection in direct burial if required. This cable is non-gelfilled and can also be used in aerial applications. The armor can be removed leaving the inner cablesuitable for any indoor/outdoor use. (Temperature rating -40ºC to +85ºC)




Notes: Notes:


Cable ConstructionCable Construction

Distribution CablesLightweight, flexible, small diameter cable design.Lower total installed costs.Color-coded 900 µm buffered fibers.2 to 156 fiber counts

Breakout CablesMost rugged cable design.2.5 mm subcable jacket for each fiber.Designed for direct lashing and "J" hook applications.2 to 108 fiber counts

Individual Fibers

Grouped Fibers

What's the best way to terminate fiber optic cable? That depends on the application, costconsiderations and your own personal preferences. The following connector comparisons can makethe decision easier.Epoxy & PolishEpoxy & polish style connectors were the original fiber optic connectors. They still represent thelargest segment of connectors, in both quantity used and variety available. Practically every style ofconnector is available including ST, SC, FC, LC, D4, SMA, MU, and MTRJ. Advantages include:• Very robust. This connector style is based on tried and true technology, and can withstand the

greatest environmental and mechanical stress when compared to the other connector technologies.• This style of connector accepts the widest assortment of cable jacket diameters. Most connectors ofthis group have versions to fit onto 900um buffered fiber, and up to 3.0mm jacketed fiber.• Versions are. available that hold from 1 to 24 fibers in a single connector.Installation Time: There is an initial setup time for the field technician who must prepare aworkstation with polishing equipment and an epoxy-curing oven. The termination time for oneconnector is about 25 minutes due to the time needed to heat cure the epoxy. Average time perconnector in a large batch can be as low as 5 or 6 minutes. Faster curing epoxies such as anaerobicepoxy can reduce the installation time, but fast cure epoxies are not suitable for all connectors.Costs: Least expensive connectors to purchase, in many cases being 30 to 50 percent cheaper thanother termination style connectors. However, factor in the cost of epoxy curing and ferrule polishingequipment, and their associated consumables.




Notes: Notes:


Standard Single Mode Fiber ProfileStandard Single Mode Fiber Profile

0.2

0.3

0.4

0.5

0.6

A t t e n u a

t i o m

( d B / k m )

-20

-10

0

+10

+20

D i s p e r s i o n

( p s

/ n m x

k m )

1300 1400 1500 1600

Single Channel Transmission at 1330 nm

Attenuation Standard Single-mode fiberDispersion

Wavelength (nm)

• Historically transmission at 1310 nm dominated

• Characteristics of dispersion at 1500 nm needed addressing

DWDM

The three principal windows of operation, propagation through a cable, areindicated. These correspond to wavelength regions where attenuation is low andmatched to the ability of a Transmitter to generate light efficiently and a Receiverto carry out detection. The 'OH' symbols indicate that at these particularwavelengths the presence of Hydroxyl radicals in the cable material cause a bumpup in attenuation. These radicals result from the presence of water. They enter thefiber optic cable material through either a chemical reaction in the manufacturingprocess or as humidity in the environment. The illustration shows the variation ofattenuation with wavelength for, standard, single-mode fiber optic cable.There are 3 major windows for fiber. At about 700nm for multimode fibers silicondiodes similar to those used in a TV channel changer can be used to deliver lowcost services over short ranges.

For ranges of 5 km and above single mose fibers using transmitters at 1330nm or1550 nm are used. In the 1550 nm band it is now possible to deploy differentwavelengths over the same fiber with potentially up to 100 wavelengths. Eventuallyit is though likely that we will be able to deliver as many as 240 wavelengths eachcarrying 2.5 Gbit/s on each fiber.




Notes: Notes:


Simple Passive Fiber NetworkSimple Passive Fiber Network

• Traditional fiber connection requires at least one fiber per subscriber — Couplers at each end attach transceiver

• Heavy on fiber and transceiver costs but resilient solution

The Transmitter was typically designed using discrete electrical and Electro-optical devices. Thisvery quickly gave way to designs based upon hybrid modules containing integrated circuits, discretecomponents (resistors and capacitors) and optical source diodes (light emitting diodes-LED's or laserdiodes). The modulation function was generally performed using separate integrated circuits andeverything was placed on the same printed circuit board.

By the 1980's higher and higher data transmission speeds were becoming of interest to the data linkarchitect. The design of the Transmitter while still generally customized became more complex to

accommodate these higher speeds. A greater part of the Transmitter was implemented using VLSIcircuits and attention was given to minimizing the number of board interconnects. Intense researchefforts were undertaken to integrate the optical source diode and the transistor level circuits neededfor modulation on a common integrated circuit substrate, without compromising performance. Atpresent, the Transmitter continues to be primarily designed as a hybrid unit, containing both discretecomponents and integrated circuits in a single package.

By the late 1980's commercially available Transmitter's became available. As a result, the linkdesign could be kept separate from the Transmitter design. The link architect was relieved from theneed to do high-speed circuit design or to design proper bias circuits for optical diodes. TheTransmitter could generally be looked at as a black box selected to satisfy certain requirementsrelative to power, wavelength, data rate, bandwidth, etc. This is where the situation remains today.




Notes: Notes:


Active Fiber DistributionActive Fiber Distribution

• Active distribution can significantly reduce fiber costs — Less fiber and fewer transceivers

• Active plant outside local exchange reduces resilience

Last half Kilometre could be CopperOr fiber

To do a proper selection of a commercially available Transmitter you have to beable to know what you need in order to match your other link requirements. Youhave to be able to understand the differences between Transmitter candidates.There are many. We can not begin to approach this in total.

However, we can look at this in a limited way. Transmitter candidates can becompared on the basis of two characteristics. Transmitter candidates can becompared on the basis of the optical source component employed and the methodof modulation.

By delivering multiple channels on a single distribution fiber we can reduce therange of the final fiber section and reduce the total number of fibers over most ofthe distribution. Near to the user, perhaps a few hundred meters away, a powereddevice will be deployed to deliver the final service. The last few hundred metersmay be over fiber or over copper. Indeed by using UTP for the last few hundredmeters it is possible to deploy xDSL technology.




Notes: Notes:


Passive Network With Advanced SplittersPassive Network With Advanced Splitters

• Advanced splitters divide on wavelength

• Only passive components as outside plant

All Fiber with different wavelengths for each subscriber

Eventually it should be possible to deliver a fully passive optical solution. Eachdistribution would be over a different wavelength controlled optically at a passivesplitter using a control wavelength. This would deliver two way channels to eachuser if required enabling not just TV but interactive information services at multi-megabit speeds.




Notes: Notes:


Technology: Active Ethernet and PONTechnology: Active Ethernet and PON

CAPEX/OPEX, Reliability, Standardization, Scalability, Futureproofing

Considerations:Considerations:




Notes: Notes:


Standard Single Mode Fiber ProfileStandard Single Mode Fiber Profile

0.2

0.3

0.4

0.5

0.6

A t t e n u a

t i o m

( d B / k m )

-20

-10

0

+10

+20

D i s p e r s i o n

( p s

/ n m x

k m )

1300 1400 1500 1600

Single Channel Transmission at 1330 nm

Attenuation Standard Single-mode fiberDispersion

Wavelength (nm)

• Historically transmission at 1310 nm dominated

• Characteristics of dispersion at 1500 nm needed addressing

DWDM

Standard single mode fiber will carry signals at many different wavelengths, butthere are particular peaks in the loss curve caused by water and other moleculespenetrating the glass. The attenuation in the fiber will be minimised at particularwavelengths. These are called “windows”.

1330 and 1550 nano-meters are particularly important values.




Notes: Notes:


Actual Fiber PerformanceActual Fiber Performance

Actual Single Mode FiberPerformance

Bit Rate < 10 Gbit/sUnamplified

Optimal OperatingRegion

10 100 500 2000 10000

Transmission Distance in km

2.5

100

200

400

800

1600

G b i t / s p e r

f i b e r

In reality fibres can now be constructed to carry data at very high speeds and oververy very long distances. However as the data rate and distance between poweredrepeaters increases so does the cost.

The economical operating range is typically measured in 10s or hundreds of Gbit/sand up to about 80 km in length.




Notes: Notes:


Fiber for Course Wavelength Division Multiplexing (CWDM)Fiber for Course Wavelength Division Multiplexing (CWDM)

0.2

0.3

0.4

0.5

0.6

A t t e n u a

t i o m

( d B / k m )

-20

-10

0

+10

+20

D i s p e r s i o n

( p s

/ n m x

k m )

1300 1400 1500 1600

Attenuation Dispersion

Wavelength (nm)

Low Water Peak Fiber allows CWDM over full available spectrum

SSMF

LWPF

By using different wavelengths of light down the same fiber it is possible toincrease the data carried.

Course Wavelength Division Multiplexing can be undertaken on most fibers, andby improving the water peak up to about 8 wavelengths can be carried.




Notes: Notes:


Long Haul Fiber With Dense WDMLong Haul Fiber With Dense WDM

0.2

0.3

0.4

0.5

0.6

A t t e n u a

t i o m

( d B / k m )

-20

-10

0

+10

+20

D i s p e r s i o n

( p s

/ n m x

k m )

1300 1400 1500 1600

Attenuation

Standard Single-mode fiberDispersion

Wavelength (nm)

Long HaulFiber

Using much more precision and deploying very narrow bands of light it is possibleto pack many frequencies into the 1550 nm band.

This is known as dense wavelength division multiplexing.




Notes: Notes:


ITU-T Standard Spacing for DWDM ChannelsITU-T Standard Spacing for DWDM Channels

Standard ITU Wavelengths for DWDM 50 GHz, and 100 GHz Spacing

Lα Lβ Cα Cβ Sα Sβ

THz nm THz nm THz nm THz nm THz nm THz nm186.00 1611.79 186.05 1611.35 191.00 1569.59 191.05 1569.18 196.00 1529.55 196.05 1529.16186.10 1610.92 186.15 1610.49 191.10 1568.77 191.15 1568.36 196.10 1528.77 196.15 1528.38186.20 1610.06 186.25 1609.62 191.20 1567.95 191.25 1567.54 196.20 1527.99 196.25 1527.60186.30 1609.19 186.35 1608.76 191.30 1567.13 191.35 1566.70 196.30 1527.22 196.35 1526.83186.40 1608.33 186.45 1607.90 191.40 1566.31 191.45 1565.90 196.40 1526.44 196.45 1526.05186.50 1607.47 186.55 1607.04 191.50 1565.50 191.55 1565.09 196.50 1525.66 196.55 1525.27186.60 1606.60 186.65 1606.17 191.60 1564.68 191.65 1564.27 196.60 1524.89 196.65 1524.50186.70 1605.74 186.75 1605.31 191.70 1563.86 191.75 1563.45 196.70 1524.11 196.75 1523.72186.80 1604.88 186.85 1604.46 191.80 1563.05 191.85 1562.64 196.80 1523.34 196.85 1522.95

190.50 1573.71 190.55 1573.30 195.50 1533.47 195.55 1533.07 200.50 1495.22 200.55 1494.85190.60 1572.89 190.65 1572.48 195.30 1532.68 195.65 1532.29 200.60 1494.48 200.65 1494.11190.70 1572.06 190.75 1571.65 195.70 1531.90 195.75 1531.51 200.70 1493.73 200.75 1493.36190.80 1571.24 190.85 1570.83 195.80 1531.12 195.85 1530.72 200.80 1492.99 200.85 1492.62190.90 1570.42 190.95 1570.01 195.90 1530.33 195.95 1529.94 200.90 1492.25 200.95 1491.88

• There is now an ITU-T standard for DWDM with 240 different wavelengths

There is now an ITU-T standard that allows 240 different wavelengths on the samefiber. No implementations of this number yet exist but there are examples of asmany as 80 wavelengths each running 2.5 Gbit/s on a single fiber pair.




Notes: Notes:


SONET/SDHSONET/SDH

• Synchronous Optical Network (SONET) was developed in the early 1990s

• Known as SDH Internationally for rates above 150 Mbit/s

— OC = optical carrier — STM = synchronous transport module — STS = synchronous transport signal

SONET (ANSI) Mbit/s SDH (ITU)

STS-1 or OC1 51.84STS-3 or OC3 155.52 STM-1

STS-12 or OC12 622.08 STM-4STS-24 or OC24 1244.16STS-48 or OC48 2488.32 STM-16

STS-192 or OC192 9953.28 STM-64

SONET was developed first in the USA during the early 1990s. The aim was toproduce a transmission system that could run at much higher rates that PDH, carryany kind of traffic and become a world standard rather than just a North Americanone. The lowest rat of SONET,51.84 Mbit/s is arranged so that it could carry a DS3at 45 Mbit/s and have enough margin in bit rate to allow for slippage wheredifferent clocks are used.

The next rate of 155.5 Mbit/s was selected so that it could carry an E4 at 140 Mbit/swith enough margin again to allow for clock slippage. From 155.52 Mbit/sSONET and the international equivalent standard, Synchronous Digital Hierarchyare essentially the same. Higher rates are constructed by selecting multiples of fourtimes for SDH.

Notice that the multiples of bit rates are exact with no additional framing overheadused in the PDH hierarchy.

SONET can be carried over any media, so the standard name for the rate starts STS.If it runs over fiber the rate starts OC. SDH is only defined for fiber so STM-1 isidentical in rate to OC3.




Notes: Notes:


SDH NetworksSDH Networks

140155

2 3 4

1 4 0

1 5 5

234

2 3 4

3 4

34

2 3 4

1 4 0

1 5 5

6 2 2

6 2 2

6 2 2

234140155

6 2 2

6 2 2

6 2 2

Break

34

2

34

2

Originatingsubscriber

Receivingsubscriber

BBXWBX

2.5-Gbit/s synchronous FOTS622-Mbit/s synchronous FOTSNetwork terminals

155 Mbit/s622 Mbit/s2488 Mbit/s

622-Mbit/s synchronousadd/drop multiplexer (ADM)155-Mbit/s synchronousadd/drop multiplexer (ADM)

Network management center

SDH networks can be constructed from many SDH components.

Synchronous Add/Drop multiplexers can insert and remove synchronous payloadsas multiples of E1, E2, E3 or E4 as required. These would be dropped initially intoSTM-1 or STM-4 services.

Wideband multiplexers can combine STM-1s and lower rates into STM-4 servicesat 622 Mbit/s.

Broadband multiplexers can then combine STM-4s into STM-16s.






Notes: Notes:


E1E3

E1E3

E1E3

E1E3ADM

SDH RingsSDH Rings

• Ring (single or dual) — can provide fault tolerance — Becoming the most popular SDH topology

Perhaps the most important topology however is the SDH ring. This enables groupsof multiplexers to be interconnected with pairs of fibers where one of the two isused and the other is a standby. In the event of a failure of a pair the service can bereconfigured to maintain delivery.




Notes: Notes:


Automatic Protection SwitchingAutomatic Protection Switching

• SONET/SDH includes standardized mechanisms for Automatic ProtectionSwitching (APS)

•

Benefits of APS — Faster restoration of service when failure occurs (or service deteriorates) – Optical path may be severed – Electronic equipment may fail or lose power – Standardization allows APS in a multivendor environment

— Protection switching may be used during maintenance or testing• APS requires a pre-provisioned protection facility (backup route)

— Operates using section (multiplexer section) APS channels

The signaling overhead of the regenerator section and the multiplexer section allowfor alarms to be transferred between devices that identify failures and themanagement center. It is then possible with Automatic Protection Switching (APS)to instruct a device to reconfigure itself automatically in the event of failure within50 msec.




Notes: Notes:


Self Healing RingsSelf Healing Rings

Working

backup

Data

Data

Break

No data

Alarm!

APS

APS

Normal Operation Fiber Break Causes Alarm

Automatic Rerouting Re-establishes Service

With APS implemented throughout a ring it is possible to produce self healingrings. Typically a network would be constructed by interconnecting these selfhealing rings at two or more points so producing networks which have multiplepaths between sites each able to offer highly reliable services.




Notes: Notes:


Cable TV Delivery Systems

Terrestrial Delivery

IP Delivery

Encoding Methods

Chapter Summary

Broadcast Distribution SystemsBroadcast Distribution Systems




Notes: Notes:


Over-The Air BroadcastingOver-The Air Broadcasting

• Transmissions are sent over radio links — Generally dedicated licensed channels in the VHF or UHF Bands — Range is limited to line of sight

– Over flat terrain may be 50 km – In hilly or mountainous areas

range my be only a few km — Multiple frequencies used

to deliver each channel in adjacent areas

There are various bands on which televisions operate depending upon the country. The VHF and UHF signals inbands III to V are generally used. Lower frequencies do not have enough bandwidth available for television.Although the BBC initially used Band I VHF at 45 MHz, this frequency is no longer in use for this purpose.Band II is used for FM radio transmissions. Higher frequencies behave more like light and do not penetratebuildings or travel around obstructions well enough to be used in a conventional broadcast TV system, so theyare generally only used for satellite broadcasting, which uses frequencies around 10 GHz. TV systems in mostcountries relay the video as an AM (amplitude-modulation) signal and the sound as a FM (frequency-modulation) signal. An exception is France, where the sound is AM.Digital Terrestrial TV is transmitted on radio frequencies that are similar to standard analog television, with theprimary difference being the use of multiplex transmitters to allow reception of multiple channels on a singlefrequency range (such as a UHF or VHF channel).The amount of data that can be transmitted (and therefore the number of channels) is directly affected by themodulation method of the channel. The modulation method in DVB-T is COFDM with either 64 or 16 stateQuadrature Amplitude Modulation (QAM). In general a 64QAM channel is capable of transmitting a greaterbitrate, but is more susceptible to interference. 16 and 64QAM constellations can be combined in a singlemultiplex, providing a controllable degradation for more important programme streams. This is calledhierarchical modulation.The DVB-T standard is not used for terrestrial digital television in North America. Instead, the ATSC standardcalls for 8VSB modulation, which has similar characteristics to the vestigial sideband modulation used foranalogue television. This provides considerably more immunity to interference, and effectively does not providefor single-frequency network operation (which is in any case not relevant in the United States).Both systems use the MPEG-2 transport stream and video codec; they differ significantly in how relatedservices (such as multichannel audio, captions, and program guides) are encoded.




Notes: Notes:


Digital Terrestrial in UKDigital Terrestrial in UK

• Receiving digital terrestrial television in the UK needs a set-top box

• There are 6 multiplexes labelled 1, 2, A, B, C and D

• Each multiplex is an error-protected bi stream of 18 or 24 megabits persecond — BBC controls Multiplex 1 — ITV and Chnnel 4 Multiplex 2 — ITV Digital controlled other services until its collapse in May 2002 — The Freeview consortium stepped in to save Digital services — Multiplex A is now largely controlled by SC4 and what remains of ONDigital — BBC acquired control of one Multiplex (B) for its own services — Crown Castle/National Grid the other two (C & D) for commercial services

Digital terrestrial television in the United Kingdom is made up of over fifty primarily free-to-airtelevision channels (including all six non-RSL analogue stations) and over twenty radio channels -primarily from the Freeview branded and Top Up TV services. It is intended that digital terrestrialtelevision will completely replace analogue television in the UK by 2012.

Digital terrestrial television launched in the UK on 15 November 1998 (just after digital satellitetelevision on 1 October 1998). The technology required that the UK government license thebroadcast of channels in six groups, or multiplex (usually abbreviated to 'mux') labelled 1, 2, A, B,C, and D[1]. Each multiplex is an error-protected bitstream of 18 or 24 megabits per second, whichcan be used for almost any combination of digitally-represented video, audio and data. The DVB-Tstandard provides a multiplex service that can make trade-offs between the number of services andthe picture and audio subjective quality.

The Independent Television Commission (ITC) allocated each existing analogue terrestrial channelhalf the capacity a multiplex each. This meant the BBC got a multiplex to themselves (Multiplex 1),ITV and Channel 4 shared Multiplex 2 (though 10% of the capacity was given to Teletext Limited )and Five and S4C shared Multiplex A. The remaining space (Muliplexes B, C and D) was thenauctioned off. A consortium made up of Granada and Carlton (members of the ITV network, whichhave now merged to form ITV plc ) and BSkyB successfully bid for these licences, and set-up thesubscription ONdigital service, (though BSkyB left the consortium prior to launch).




Notes: Notes:


DVB-T Services – Mux 1 and 2DVB-T Services – Mux 1 and 2

• Multiplex 1 — Operated by the BBC; broadcasts nationwide in 16QAM mode at 18

megabits/second

— TV: BBC One (regional variation), BBC Two (national variation), BBC Three,CBBC Channel, BBC News 24 — Radio: BBC Radio Wales (Wales only), BBC Radio Scotland (Scotland

only), BBC Radio Ulster (Northern Ireland only), BBC Radio Cymru (Walesonly), BBC Radio nan Gaidheal (Scotland only), BBC Radio Foyle (NorthernIreland Only)

— Text/Interactive: BBCi, The Engineering Channel• Multiplex 2

— Operated by Digital 3&4 (an ITV/Channel 4 consortium); broadcastsnationwide in 64QAM mode at 24 megabits/second

— TV: ITV1 (regional service), Channel 4, ITV2, ITV3, More4, E4, ITV4,Film4+1, Setanta Sports 1*, CITV Channel

— Radio: U105 (Northern Ireland only), Heart (except Scotland), Radio MusicShop (except Scotland)

— Text/Interactive: Teletext, Teletext Holidays (Wales only), Teletext Cars,Teletext on 4, Teletext on ITV

* Pay TV Services




Notes: Notes:


DVB-T Services – Mux A and BDVB-T Services – Mux A and B

• Multiplex A — Operated by SDN (owned by ITV plc); broadcasts nationwide in 64QAM

mode at 24 megabits/second

— TV: S4C Digidol (Wales only), Five, TeleG (Scotland only), ABC1 (exceptWales), QVC, UKTV Gold*, bid tv, price-drop tv, TCM*, UKTV Style*,Discovery Channel*, British Eurosport*, Five US, Five Life, Top Up Anytime1, Top Up Anytime 2, Top Up Anytime 3, Discovery Real Time*, CartoonNetwork*, S4C2 (Wales only), Teachers' TV, Television X*

— Radio: BBC Radio 1, BBC Radio 2, BBC Radio 3, BBC Radio 4, Mojo(except Wales), Heat (except Wales)

— Text/Interactive: Teletext Holidays (except Wales), Teletext Games, Top UpTV Active

• Multiplex B — Operated by the BBC; broadcasts nationwide in 16QAM mode at 18

megabits/second — TV: BBC Four, CBeebies, BBC Parliament, Community Channel — Radio: BBC 1Xtra, BBC Radio Five Live, BBC Five Live Sports Extra, BBC 6

Music, BBC 7, BBC Asian Network — Text/Interactive: BBCi (301, 302, 303), BBC Parliament (redundant ¼

screen service), The Engineering Channel* Pay TV Services




Notes: Notes:


DVB-T Services – Mux C and DDVB-T Services – Mux C and D

• Multiplex C — Operated by National Grid Wireless; broadcasts nationwide in 16QAM mode

at 18 megabits/second — TV: Sky Three, UKTV History, E4+1, SmileTV, Sky News, Sky Sports News — Radio: talkSPORT, 3C, Premier Christian Radio, Virgin Radio — Text/Interactive: Sky Text, TVTV Digital

• Multiplex D — Operated by National Grid Wireless; broadcasts nationwide in 16QAM mode

at 18 megabits/second — TV: The Hits, UKTV Bright Ideas, Ftn, TMF, Ideal World, Film4, ITV Play — Radio: BBC World Service, The Hits Radio, Smash Hits, Kiss 100, Magic

105.4, Q, Oneword, 102.2 Smooth FM, Kerrang! — Text/Interactive: 4TVInteractive

* Pay TV Services




Notes: Notes:


Multiplexing TechnologyMultiplexing Technology

• Some multiplexes carry more services than others

• Some channels share bandwidth as channels transmit at different times

• Different channels use different bandwidths — For example BBC1 uses 4.4 Mbit/s — Sky Sports News uses only 2 Mbit/s

• There are three basic modulation schemes currently in use in the UK; — QPSK (only used for tests in the Oxford and London areas) — 16 QAM — 64 QAM

• Each with a progressively higher bitrate and thus SNR — The cost is of progressively higher likelihood of signal degradation

• Currently multiplexes 2 and A use 64 QAM and the others 16 QAM

Some of these multiplexes carry a much larger number of services than others for various reasons. Firstly, anumber of services share bandwidth — so some channels turn off when others are on (for example one willnever see CBeebies and BBC Four on air at the same time, as they use the same space in Multiplex B, withCBeebies broadcasting from 6am until 7pm and BBC Four from 7pm onwards; the situation is the same forCBBC and BBC Three). In addition, some multiplexes have fewer channels so as to allocate more data to fewerservices, thus ensuring higher quality (for example, BBC One on Multiplex 1 is carried as a 4.4 Megabit stream,while Sky Sports News typically uses 2 Megabits per second).

On top of this, the modulation of the multiplexes can be varied to squeeze higher digital bitrates out of the sameportion of the electromagnetic spectrum. This comes at the cost of making it harder to get a good signal. Thereare three basic modulation schemes currently in use in the UK; in order of bandwidth efficiency, they are:QPSK (only used for tests in the Oxford and London areas), 16 QAM and 64 QAM, each with a progressivelyhigher bitrate, at the cost of progressively higher likelihood of signal degradation. Currently multiplexes 2 andA use 64 QAM (and are consequently more prone to poor reception) while the other multiplexes all currentlyuse 16 QAM.

Furthermore, multiplexes can make use of statistical multiplexing at the MPEG video coder whereby the bitrateallocated to a channel within the multiplex can vary dynamically depending on how difficult it is to code thepicture content at that precise time, and how much demand there is for bandwidth from other channels. In thisway, complex pictures with lots of detail may demand a higher bitrate at one instant and this can result in thebitrate allocated to another channel in the same multiplex being reduced if the second channel is currentlytransmitting pictures which are easier to code, with less fine detail. The only channel on the DTT system not touse statistical multiplexing, i.e. has a constant bit rate, is BBC One. This is so the English Regions and Nationscan perform a simple transmultiplex, or T-Mux, operation and insert their local version of BBC One over theLondon feed straight into the existing BBC Multiplex 1 without having to re-code the entire multiplex at eachregional centre, requiring specialist (and costly) equipment at several locations.




Notes: Notes:


AReduced hours in Wales (not broadcast 0900-1700 Tuesday-Thursday)QVC16ANot available in Wales; broadcasts 0600-1800ABC115

2E414

2More413

CBroadcasts 0500-0100UKTV History12

CSky Three11

2ITV310

BBroadcasts 1900-0600BBC Four9

AScotland only; broadcasts 1800-1900TeleG

2Wales onlyChannel 48

1Broadcasts 1900-0600BBC Three7

2ITV26

A3Five5

A3Wales onlyS4C Digidol

2Except WalesChannel 4

4

In Northern Ireland 2UTV

In Central and Northern Scotland 2STV 2

In England, Wales, Southern Scotland, the Isle of Man and the Channel Islands 2ITV1

3

1Includes regional variations ; digital variations from analogue in Wales and Northern IrelandBBC Two2

1Includes regional variationsBBC One1

MultiplexNotesChannelLCN 1

Logical Channel Number

ITV1 is the brand name for 12 of the 15 regional ITV Network franchises forEngland, Wales, southern Scotland, the Isle of Man and the Channel Islands. Eachof these 12 franchises has a separate brand name used prior to local programming,see ITV1. STV is the brand name for the franchises for central and northernScotland. UTV operates the franchise for Northern Ireland. All 15 franchises

broadcast 0925-0600; GMTV operates the franchise for national breakfasttelevision and broadcasts 0600-0925.

Five, S4C and S4C2 will move to a public service multiplex at the start of digitalswitchover, using the bandwidth created by switching from 16QAM to 64QAMmode, so will be transmitted from all 1,154[7] UK transmitters. Multiplexes A, Cand D will only be transmitted from the current 80 transmitters after switchover butwith higher powered signals (and in 64QAM mode).




Notes: Notes:


CBroadcasts 0100-0500SmileTV37

ABroadcasts 0500-2300Five Life36

AFive US35

2Pay-per-view service (from Top Up TV); broadcasts dependent on SPL match timesSetanta Sports 134

ATop Up TV; broadcasts 1300-1800British Eurosport33

2Film4+132

DITV Play31

CE4+130

DFilm429

2Broadcasts 1800-0600ITV428

ATop Up TV; broadcasts 1800-2300Discovery Channel27

ATop Up TV; broadcasts 1300-1600UKTV Style26

ATop Up TV; broadcasts 1900-0055TCM25

Aprice-drop tv24

AReduced hours in Wales (only broadcasts 0600-1900)bid tv23

DIdeal World22

DTMF21

DBroadcasts 1800-0600Ftn20

DBroadcasts 0600-1800UKTV Bright Ideas19

DThe Hits18

ATop Up TV; broadcasts 1600-0100UKTV Gold17








Notes: Notes:


CH27Five High Definition Test Channel; London onlyFive HD Trial505CH27C4 High Definition Test Channel; London onlyChannel 4 HD Trial504

CH27ITV High Definition Test Channel; London onlyITV HD Trial503

CH31BBC High Definition Test Channel; London onlyBBC HD Trial501

APay-per-view service; placeholder (no longer broadcasting)Red Hot TV98

ATop Up TV (additional subscription); broadcasts 2300-0500Television X97

ABroadcasts 1100-1300Teachers' TV88

BBroadcasts 0600-0900Community Channel87

A3Wales only; broadcasts 0900-1700 Tuesday-ThursdayS4C286

CSky Sports News83

CSky News82

BBBC Parliament81

1BBC News 2480

2Broadcasts 0600-1800; not broadcast while Setanta Sports 1 is on airCITV Channel75

ATop Up TV; broadcasts 0900-1100Cartoon Network 72

BBroadcasts 0600-1900CBeebies71

1Broadcasts 0600-1900CBBC Channel70

ANot yet launchedTop Up TV Promo43

ATop Up TV; broadcasts 0600-1200Discovery Real Time42

ASubscription service; not yet launchedTop Up TV Anytime 340












Notes: Notes:


Wireless StandardsWireless Standards

There are 4 major classifications of wireless services based upon range. BroadcastTV has generally evolved from the WAN area while LANs are where data servicesstarted. Convergence of the technology now makes it feasible to deliver TV overany of these technologies.




Notes: Notes:


Multipoint Distribution ServicesMultipoint Distribution Services

Two versions of MDS have evolved:

• LMDS – Local MDS

— Single-duplex channel to a local hub — Generally uses 28, 35, 38 GHz bands — Can provide high speed data where wired infrastructure is inadequate — Typical range: 5–8 km

• MMDS – Multichannel MDS — Multiple simplex or duplex channels to a local hub — Generally employed in 2.4, 5 GHz bands — CATV distribution over MMDS — Typical range: 55 km

• Wimax: IEEE 802.16 is addressing physical/MAC layer, and frequencycoexistence standards

In the USA and Canada wireless local loop technologies have been used for sometime.




Notes: Notes:


Example MMDS SystemExample MMDS System

HPA = High Power AmplifierLNB = Low Noise Block Convector

SatelliteReceivers

Decoder

FrequencyTranslation

Modulation &Encryption

HPA

LNB

DecodedReceiver

Omni transmitantenna

Tree topHouse topReceive Antenna

(may be omni ordirectional)




Notes: Notes:


Typical Wireless Loop InstallationTypical Wireless Loop Installation

RST = radio subscriber terminal

Systemcontroller

Cellsite

Systemcontroller

Cellsite

Localexchange

Concentrator

RST

RST

RST

Fax machine

M i c r o

w a v e

Fiber




Notes: Notes:


Existing TechnologiesExisting Technologies

• Some suppliers use cellular equipment to provide wireless loops

DMS-MTX and 800Northern Telecom

WiLLMotorola

RAS 1000Ericsson

DAXnode 2000Nokia

Wireless Subscriber SystemAT&T

7390 LMDSAlcatel

ProductVendor

Most major telecom vendors have developed wireless loop technologies. Howeverthese are converging onto a common future set of standards.




Notes: Notes:


Licensed and Licensed-exempt SpectrumLicensed and Licensed-exempt Spectrum

WiMAX will bring this technology closer to a common reality across the world.The frequencies of use will differ but in Europe frequencies in the 3.5 GHz bandwill be used.




Notes: Notes:


IEEE 802 LANsIEEE 802 LANs

• IEEE 802 LANs started with the introduction of Ethernet — 10 Mbit/s Bus LAN based upon 0.4 inch yellow coaxial cable — Developed by Xerox in 1979 — Limited to about 1.5 km

• IEEE founded the 802 committee to standardize LANs in 1980

PhysicalLayer

DataLinkLayer

802.3CSMA/CD

Bus

802.4TokenBus

802.5TokenRing

802.6DQDBMAN

802.11Wireless

LAN

802.16WiMAXWirelessAccess

802.20Wireless

Broadband

802.1d Bridging

802.2 Logical Link Control

8 0 2

. 1 0 S e c u r i t y

8 0 2

. 1 M a n a g e m e n

t

IEEE 802 standards generally provide the standardization of protocols and servicesat the physical and data link layers. The physical layer defines the transmission ofbits and the hardware elements of connection. The data link layer is responsible forthe transmission of frames of data, error detection within those frames and thesharing of access to the physical transmission medium.




Notes: Notes:




IP Delivery

Encoding Methods

Chapter Summary





Notes: Notes:


Set Top BoxesSet Top Boxes

• Set top boxes are evolving to increase their functionality

• Initially they provided

Cable DecoderFree to air Digital

Combinationwith PersonalVideo Recorderwith HDD

Cable services generally terminate at the user site on a set top box. Thearchitecture of these is changing to add more processing power and faster, morestandardised protocols.








Notes: Notes:


3GPP3GPP

• The 3rd Generation Partnership Project (3GPP) is a collaborationagreement that was established in December 1998 — ETSI (Europe) — ARIB/TTC (Japan) — CCSA (China) — ATIS (North America) — TTA (South Korea)

• The scope of 3GPP is to make a globally applicable third generation (3G) — mobile phone system specification within the scope of the ITU's IMT-2000

• 3GPP specifications are based on evolved GSM specifications, nowgenerally known as the UMTS system.

Each Release incorporates hundreds of individual standards documents, each ofwhich may have been through many revisions. Current 3GPP standards incorporatethe latest revision of the GSM standards. 3GPP's plans for the future beyondRelease 7 are currently in the early stages of development under the title LongTerm Evolution ("LTE").

3GPP documents are made available freely on the organisation's web site. Whilst3GPP standards can be bewildering to the newcomer, they are a remarkablycomplete and detailed resource and provide insight into how the cellular industryworks. They cover not only the radio part ("Air Interface") and Core Network, butalso billing information and speech coding down to source code level.Cryptographic aspects (authentication, confidentiality) are also freely specified indetail. 3GPP2 offer similar information about their system.




Notes: Notes:


Telecommunications and Internet Converged Services forAdvanced Networking (TISPAN)

Telecommunications and Internet Converged Services forAdvanced Networking (TISPAN)

• TISPAN specifies a Next Generation Network which: — Offers standardised multimedia services based on xDSL — Uses the 3GPP IMS for service handling, ensuring FMC — Supports legacy POTS services (PSTN/ISDN Emulation)

– Same as the PSTN/ISDN Telephony service over an IP infrastructure – Enables use of ISDN Supplementary services and phone at home – So NGN will replace the soon becoming obsolescent PSTN

— Supports a set of fully-defined Supplementary Services (Simulation)including Voice – Similar - but not identical - to existing PSTN service – Based on IMS capabilities – TISPAN’s specific needs to accommodate Wireline access to IMS are

covered by a collaboration between TISPAN and 3GPP: TISPAN-specificIMS extensions are prepared in TISPAN and proposed for inclusion to3GPP IMS Rel-7. Joint meetings between TISPAN and 3GPP.

The Telecoms & Internet converged Services & Protocols for Advanced Networks(TISPAN) is a standardization body of ETSI, specializing in fixed networks andInternet convergence. It was formed in 2003 from the amalgamation of the ETSIbodies Telecommunications and Internet Protocol Harmonization Over Networks(TIPHON) and Services and Protocols for Advanced Networks (SPAN)TISPAN's focus is to define the European view of the Next Generation Networking(NGN) though TISPAN also includes much participation from regions outside

Europe. TISPAN Release 1 was published in December 2005. The Release 1architecture is based upon the concept of cooperating subsystems sharing commoncomponents. This subsystem-oriented architecture enables the addition of newsubsystems over the time to cover new demands and service classes. Thearchitecture ensures that the network resources, applications, and user equipmentare common to all subsystems and therefore ensure user, terminal and servicemobility to the fullest extent possible, including across administrative boundaries.One of the key subsystems is based upon the 3GPP IP Multimedia Subsystem(IMS) Release 6 and 3GPP2 Revision A architectures.TISPAN has adopted the IMS architecture given in release 6 and is adding wirelineaccess to the same.




Notes: Notes:


xDSL

IMSIMS

FIXEDFIXED MOBILEMOBILE

HSSHSS

Role of ETSI-TISPANRole of ETSI-TISPAN

• TISPAN defines: — The Fixed Core Network — TISPAN addresses

Fixed-access impactson 3GPP’s IMS

To meet the rising demands relative to IP multimedia applications, the 3rdGeneration Partnership Project (3GPP) promotes the IP Multimedia Subsystem(IMS). 3GPP defines the specifications for radio access by both WCDMA andGSM. It acts a facilitator for R99 and R4, inclusive of antenna interfacespecifications, voice service specifications in circuit switched (CS) domains, andbasic data service specifications in packet switched (PS) domains. With respect toR5 and R6 research in relation to IP multimedia applications, R5 defines the corenetwork architecture, public components, and basic service flows of IMS. Based onthe extension of some R5 components, R6 defines the key service capability ofIMS, Quality of Service (QoS), network interoperability, and also IMS/CSintegration.The IMS architecture derived from 3GPP is broadly recognized as a reasonablycomprehensive solution to the IP multimedia domain. 3GPP2 and TISPAN haveadjusted their IP multimedia network architectures and service systems according tothe 3GPP IMS model. In terms of their responsibilities with regard to IMS, 3GPP2is handling access for cdma2000, and fixed networks are under the remit ofTISPAN (Telecommunications and Internet Converged Services and Protocols for

Advanced Networking).




Notes: Notes:


Drivers in Sizing TV ServicesDrivers in Sizing TV Services

• Network capacity needed for TV Services depends upon several factors

• Subscriber audience size

– Bigger audiences need more access• Take-up rate of services

— Take-up patterns vary from audience to audience• Resolution of TV programs distributed

— Standard resolution:2 Mbit/s to 4 Mbit/s of bandwidth — HDTV requires 5 Mbit/s of bandwidth with MPEG-4

• Number of channels offered

• Number of channels actually watched

• Kind of service — Unicast or multicast




Notes: Notes:


IPTV and VODIPTV and VOD

• Commitment to next generation broadband access network is criticalenabler for sufficient quality of service — NTT target of 30m FTTH customers by 2010 in Japan

• Innovation and market development being held back by uncertainregulatory environments

• Demand could be tempered by dual screen environment rather thanconvergence

• Growing market for consumer electronics able to timeshift viewing mayaffect IPTV take up — Sales of DVD HDD recorders reached 5.5m in 2005 — Sony X Video Station to launch this year. A PVR with 8 tuners and 2

terabytes of hard disk memory

VoD services already exist in Japan, Korea and parts of the USA. There are smallpockets around Europe too. Early indications are that take-up is price sensitive asone might expect. Where Time-slip TV is offered this is attractive to viewers andresults in greater usage than premium rate moves. Even within subscribers theusage rarely reached 15% of subscribers at any time. Usage is more dependent uponwhat programming on free-to-air channels was. Where this was strong mostviewers would not invest the time to decide what movie to watch!




Notes: Notes:


Efficient DistributionEfficient Distribution

• Efficient distribution may require the duplication of some services — VoD services are best located near to subscriber — Broadcast channels of recorded video and moves may work best duplicated

To avoid transferring large amounts of data from one side of the network to anotherduplication of servers will be necessary in many networks. Bulk transfer of contentis probably best achieved by man-in-a-van transfer.




Notes: Notes:


IPTV Bandwidth RequirementsIPTV Bandwidth Requirements

• Video — IPTV with MPEG2 compression

– Standard Definition 3.5 - 4.5 Mbps – High Definition 12 -19.3 Mbps

— IPTV with MPEG4 compression – Standard Definition 1.5 - 2.0Mbps – High Definition 5.0 - 8.0Mbps

• Access speeds for triple play might need to reach double the aggregate — How fast would access need to be? — Most locations cannot achieve this over long copper loops — Need to replace with fiber loops or accept lower quality

A key issue in the distribution of IPTV might be the stream bandwidth. Mosthouseholds have more than one TV so we must expect fixed access speed to growto match a profile that includes more than a single IP-TV stream. Given that HDTVbecomes a significant consumer demand, delivery of this over MPEG-2 requiresperhaps double the access speed. We might expect access speeds to need to bedouble the aggregated service so to deliver 2 HD-TV channels demands access toreach in excess of 20 Mbit/s with MPEG-4 and perhaps closer to 50 Mbit/s withMPEG-2.




Notes: Notes:


Centralized Architectures?Centralized Architectures?

With an aggregated access speed of about 20 Mbit/s per household the bradbandaccess nodes will need to be sized appropriately. Where a nominal 1000 loops aresupported on a single rack, the reach must support about 20 Gbit/s of switchingcapacity at least. In the broadband aggregation networks each rack might thereforerequire backhauls of two 10G Ethernet trunks per rack. Policy control services willdeliver QoS for different services to control quality through the access which willbe the limiting part of the network.The broadband edge devices will then convert to MPLS for delivery across thecore.

Should we place services closely coupled to the core delivering a centralizedservice? This may not be optimal for all services. Indeed very much NOT optimalfor VOD.




Notes: Notes:


Distributed Architectures?Distributed Architectures?

By delivering video content regionally or even closely coupled to the access itbecomes less necessary to carry large numbers of VOD sessions over the core. Acentralized storage of content for long-term access could still be of benefit butwould be transferred to regional or local servers for multiple local access.




Notes: Notes:


Network Dimensioning Is CriticalNetwork Dimensioning Is Critical

First Mile: In the access IPTV dominates the bandwidth use. This might be toreceive broadcast TV or VOD. For any one TV set it is likely that only one of thesewill be in use at any one time.

Second Mile: As we move to the aggregation level broadcast content mightdominate since many different channels are likely to be watched all of which mustbe carried through the aggregated access. It is also likely that only a minority will

request VOD at any one time.Third Mile: As we further aggregate services each individual VOD session becomesa new band to be individually carried so at the edge of the core it will become thedominant service. This will drive the deployment of VOD services from regional orlocal service points closer to the access.




Notes: Notes:


Switching CapacitySwitching Capacity

• MSAN Switching capacity must service switching from access to back-haul

• Throughput of switch should exceed twice sum of throughput

— This is necessary for queuing allowance — Eventually may be desirable to make switch non blocking• Example:-

— Offer each user 8 Mbit/s down and 2 Mbit/s up total = 10 Mbit/s — 1000 users lines: usage is 10 Mbit/s x 1000 =

10 Gbit/s capacity for non blocking access — At only 10% utilization total load is 10 x 0.1 = 1 Gbit/s so we need to provide

not less than twice this = 2 Gbit/s — At 40% utilization total load is 40 x 0.1 = 4 Gbit/s so we need to provide not

less than twice this = 8 Gbit/s

The final backhaul capacity provided and the switching capacity needed largelydepends upon the access speeds offered and the utilization levels used by users. Asaccess speeds increase, utilization levels generally reduce. There is after all a limitto how fast a user can read or click a mouse. Higher access speeds will notsignificantly increase reading speed!

However the provision of faster and faster services enablers new applications to be

delivered and migration from usage just based upon web surfing, with utilization ator below 10%. HDTV with utilization at or above 50% can drastically change thebackhaul capacity needed, and thus the total switching speed. We need to reducethe backhaul demand by ensuring the majority of TV is multicast.




Notes: Notes:


WinampWinamp




Notes: Notes:


Video LAN ClientVideo LAN Client






Notes: Notes:




IP Delivery

Encoding Methods

Chapter Summary







Notes: Notes:


Compression MethodsCompression Methods

• Image Compression Methods — JPEG — GIF 89a — Wavelet Compression

• Sound Compression — MPEG Audio Overview — MPEG Layer-3 (MP3) — MPEG AAC

• Video Compression Methods — H.261 — MPEG/MPEG-2 — MPEG-4 — MPEG-7

There are multiple forms of compression available. These have evolvedindependently in many cases. We will examine static images first and then MPEG2which will include many of the sound compressions elements used then we willexamine MPEG4 and H.264. Finally we will address MPEG7 for completeness.




Notes: Notes:


JPEG Compression: BasicsJPEG Compression: Basics

• Human vision is insensitive to high spatial frequencies

• JPEG Takes advantage of this by compressing high frequencies more

coarsely and storing image as frequency data• JPEG is a “lossy” compression scheme.

Losslessly compressed image, ~150KB JPEG compressed, ~23KB

On the left the image is compressed with a loss-less compressions systems – GIF.In the right the same image compresses to one sixth the size using JPEG. Whileinitially the two images look the same, close inspection will show loss of somedetail. The level of compression can be selective to match quality to application.




Notes: Notes:


GIF 89a examples vs. JPEGGIF 89a examples vs. JPEG

GIF Image, 7.5KB,

optimal encoding

JPEG, blotchy spots in single-colorareas






Notes: Notes:


Wavelet Image CompressionWavelet Image Compression

• Optimal for images containing sharpedges, or continuous curves/lines (fingerprints)

•

Compared with DCT, uses moreoptimal set of functions torepresent sharp edges than cosines.

• Wavelets are finite in extent asopposed to sinusoidal functions

Several different families of wavelets.

Wavelets are functions that satisfy certain mathematical requirements and are usedin representing data or other functions. This idea is not new. Approximation usingsuperposition of functions has existed since the early 1800's, when Joseph Fourierdiscovered that he could superpose sines and cosines to represent other functions.However, in wavelet analysis, the scale that we use to look at data plays a specialrole. Wavelet algorithms process data at different scales or resolutions. If we lookat a signal with a large "window," we would notice gross features. Similarly, if welook at a signal with a small "window," we would notice small features. The resultin wavelet analysis is to see both the forest and the trees, so to speak.This makes wavelets interesting and useful. For many decades, scientists havewanted more appropriate functions than the sines and cosines which comprise thebases of Fourier analysis, to approximate choppy signals. By their definition, thesefunctions are non-local (and stretch out to infinity). They therefore do a very poor

job in approximating sharp spikes. But with wavelet analysis, we can useapproximating functions that are contained neatly in finite domains. Wavelets arewell-suited for approximating data with sharp discontinuities.http://www.amara.com/IEEEwave/IEEEwavelet.html#contents




Notes: Notes:


Wavelet TransformsWavelet Transforms

• Wavelet transforms comprise an infinite set

• Wavelet subclasses distinguished by the number of coefficients

Wavelet transforms comprise an infinite set. The different wavelet families makedifferent trade-offs between how compactly the basis functions are localized inspace and how smooth they are.

Some of the wavelet bases have fractal structure. The Daubechies wavelet family isone example on the left.

Within each family of wavelets (such as the Daubechies family) are wavelet

subclasses distinguished by the number of coefficients and by the level of iteration.Wavelets are classified within a family most often by the number of vanishingmoments. This is an extra set of mathematical relationships for the coefficients thatmust be satisfied, and is directly related to the number of coefficients (1). Forexample, within the Coiflet wavelet family are Coiflets with two vanishingmoments, and Coiflets with three vanishing moments. Comparision is shown on theright.




Notes: Notes:


Wavelet vs. JPEG compressionWavelet vs. JPEG compression

Wavelet compressionfile size: 1861 bytescompression ratio - 105.6

JPEG compressionfile size: 1895 bytescompression ratio - 103.8

We can use wavelets to retrieve the true image from the noise produced by thecompression. The technique works in the following way. When you decompose adata set using wavelets, you use filters that act as averaging filters and others thatproduce details. Some of the resulting wavelet coefficients correspond to details inthe data set. If the details are small, they might be omitted without substantiallyaffecting the main features of the data set. The idea of thresholding, then, is to set tozero all coefficients that are less than a particular threshold. These coefficients areused in an inverse wavelet transformation to reconstruct the data set. Figure 6 is apair of "before" and "after" illustrations of a nuclear magnetic resonance (NMR)signal. The signal is transformed, thresholded and inverse-transformed. Thetechnique is a significant step forward in handling noisy data because the denoisingis carried out without smoothing out the sharp structures. The result is cleaned-upsignal that still shows important details.




Notes: Notes:


Wavelet compression advantagesWavelet compression advantages

Fourier basis functions, time-frequencytiles, and coverage of the time-frequency plane.

Daubechies wavelet basis functions, time-frequency tiles, and coverage of the time-frequency plane

The most interesting dissimilarity between these two kinds of transforms is that individual waveletfunctions are localized in space. Fourier sine and cosine functions are not. This localization feature,along with wavelets' localization of frequency, makes many functions and operators using wavelets"sparse" when transformed into the wavelet domain. This sparseness, in turn, results in a number ofuseful applications such as data compression, detecting features in images, and removing noise fromtime series. One way to see the time-frequency resolution differences between the Fourier transformand the wavelet transform is to look at the basis function coverage of the time-frequency plane. Theleft figure shows a windowed Fourier transform, where the window is simply a square wave. Thesquare wave window truncates the sine or cosine function to fit a window of a particular width.

Because a single window is used for all frequencies in the WFT, the resolution of the analysis is thesame at all locations in the time-frequency plane.An advantage of wavelet transforms is that the windows vary. In order to isolate signaldiscontinuities, one would like to have some very short basis functions. At the same time, in order toobtain detailed frequency analysis, one would like to have some very long basis functions. A way toachieve this is to have short high-frequency basis functions and long low-frequency ones. Thishappy medium is exactly what you get with wavelet transforms. The right figure shows the coveragein the time-frequency plane with one wavelet function, the Daubechies wavelet.One thing to remember is that wavelet transforms do not have a single set of basis functions like theFourier transform, which utilizes just the sine and cosine functions. Instead, wavelet transforms havean infinite set of possible basis functions. Thus wavelet analysis provides immediate access toinformation that can be obscured by other time-frequency methods such as Fourier analysis.




Notes: Notes:


MPEG Compression ProtocolsMPEG Compression Protocols

• MPEG-1 ISO/IEC JTC1/SC29/WG11 ISO 11172 parts 1 to 4


• MPEG-3 abandoned but audio encoding

• MPEG-4 ISO/IEC JTC1/SC29/WG11 N4668

• MPEG-7 ISO/IEC JTC1/SC29/WG11N6828 — Adds descriptions language for multimedia

• MPEG-21 ISO/IEC JTC1/SC29/WG11/N5231 — Adds digital rights management

Moving Picture Experts Group (MPEG) a working group of ISO/IEC in charge ofthe development of standards for coded representation of digital audio and video.Established in 1988, the group has produced MPEG-1 , the standard on which suchproducts as Video CD and MP3 are based, MPEG-2 , the standard on which suchproducts as Digital Television set top boxes and DVD are based, MPEG-4 , thestandard for multimedia for the fixed and mobile web and MPEG-7 , the standardfor description and search of audio and visual content. Work on the new standardMPEG-21 "Multimedia Framework" has started in June 2000. So far a TechnicalReport and two standards have been produced and three more parts of the standardare at different stages of development. Several Calls for Proposals have alreadybeen issued






Notes: Notes:


MPEG2 Standard PartsMPEG2 Standard Parts

• ISO/IEC 13818-1:2000 Information technology -- Generic coding of movingpictures and associated audio information: Systems (available in Englishonly)

• ISO/IEC 13818-2:2000 Information technology -- Generic coding of movingpictures and associated audio information: Video (available in Englishonly)

• ISO/IEC 13818-3:1998 Information technology -- Generic coding of movingpictures and associated audio information -- Part 3: Audio (available inEnglish only)

• ISO/IEC 13818-4:1998 Information technology -- Generic coding of movingpictures and associated audio information -- Part 4: Conformance testing(available in English only)

• ISO/IEC 13818-4:1998/Cor 2:1998 (available in English only)




Notes: Notes:



• ISO/IEC 13818-4:1998/Amd 1:1999 Advanced Audio Coding (AAC)conformance testing (available in English only)

•

ISO/IEC 13818-4:1998/Amd 2:2000 System target decoder model (availablein English only)

• ISO/IEC 13818-4:1998/Amd 3:2000 Additional audio conformancebitstreams (available in English only)

• ISO/IEC TR 13818-5:1997 Information technology -- Generic coding ofmoving pictures and associated audio information -- Part 5: Softwaresimulation (available in English only)

• ISO/IEC TR 13818-5:1997/Amd 1:1999 Advanced Audio Coding (AAC)(available in English only)




Notes: Notes:



• ISO/IEC 13818-6:1998 Information technology -- Generic coding of movingpictures and associated audio information -- Part 6: Extensions for DSM-CC (available in English only)


• ISO/IEC 13818-6:1998/Amd 1:2000 Additions to support data broadcasting(available in English only)

• ISO/IEC 13818-6:1998/Amd 2:2000 Additions to support synchronizeddownload services, opportunistic data services and resourceannouncement in broadcast and interactive services (available in Englishonly)

• ISO/IEC 13818-7:1997 Information technology -- Generic coding of movingpictures and associated audio information -- Part 7: Advanced AudioCoding (AAC) (available in English only)




Notes: Notes:




• ISO/IEC 13818-9:1996 Information technology -- Generic coding of moving

pictures and associated audio information -- Part 9: Extension for real timeinterface for systems decoders (available in English only)

• ISO/IEC 13818-10:1999 Information technology -- Generic coding of movingpictures and associated audio information -- Part 10: Conformanceextensions for Digital Storage Media Command and Control (DSM-CC)(available in English only)




Notes: Notes:


Part 1 of MPEG2Part 1 of MPEG2

Part 1 of MPEG-2 addresses the combining of one or more elementary streams ofvideo and audio, as well as, other data into single or multiple streams which aresuitable for storage or transmission. This is specified in two forms: the ProgramStream and the Transport Stream. Each is optimised for a different set ofapplications

The Program Stream is similar to MPEG-1 Systems Multiplex. It results from

combining one or more Packetised Elementary Streams (PES), which have acommon time base, into a single stream. The Program Stream is designed for use inrelatively error-free environments and is suitable for applications which mayinvolve software processing. Program stream packets may be of variable andrelatively great length.

The Transport Stream combines one or more Packetized Elementary Streams (PES)with one or more independent time bases into a single stream. Elementary streamssharing a common timebase form a program. The Transport Stream is designed foruse in environments where errors are likely, such as storage or transmission in lossyor noisy media. Transport stream packets are 188 bytes long.




Notes: Notes:



XXLow level

XXXXXXMain level

XXXHigh-1440level

XXHigh level

4:2:2Multiview

HighSpatial

scalableSNR

scalableMainSimple

Part 2 of MPEG-2 builds on the powerful video compression capabilities of theMPEG-1 standard to offer a wide range of coding tools. These have been groupedin profiles to offer different functionalities. Only the combinations marked with an"X" are recognised by the standard. Since the final approval of MPEG-2 Video inNovember 1994, one additional profile has been developed. This uses existingcoding tools of MPEG-2 Video but is capable to deal with pictures having a colourresolution of 4:2:2 and a higher bitrate. Even though MPEG-2 Video was notdeveloped having in mind studio applications, a set of comparison tests carried outby MPEG confirmed that MPEG-2 Video was at least good, and in many caseseven better than standards or specifications developed for high bitrate or studioapplications.The 4:2:2 profile has been finally approved in January 1996 and is now an integralpart of MPEG-2 Video.The Multiview Profile (MVP) is an additional profile currently being developed. Byusing existing MPEG-2 Video coding tools it is possible to encode in an efficientway tow video sequences issued from two cameras shooting the same scene with asmall angle between them. This profile will be finally approved in July 1996.




Notes: Notes:


Part 3 of MPEG2 : AudioPart 3 of MPEG2 : Audio

Part 3 of MPEG-2 is a backwards-compatible multichannel extension of the MPEG-1 Audio standard.

Part 6 of MPEG-2 - Digital Storage Media Command and Control (DSM-CC) is thespecification of a set of protocols which provides the control functions andoperations specific to managing MPEG-1 and MPEG-2 bitstreams. These protocolsmay be used to support applications in both stand-alone and heterogeneous network

environments. In the DSM-CC model, a stream is sourced by a Server and deliveredto a Client. Both the Server and the Client are considered to be Users of the DSM-CC network. DSM-CC defines a logical entity called the Session and ResourceManager (SRM) which provides a (logically) centralized management of the DSM-CC Sessions and Resources




Notes: Notes:


MPEG Audio basics & Psychoacoustic ModelMPEG Audio basics & Psychoacoustic Model

• Human hearing limited to values lower than ~20kHz in most cases

• Human hearing is insensitive to quiet frequency components to sound

accompanying other stronger frequency components• Stereo audio streams contain largely redundant information

• MPEG audio compression takes advantage of these facts to reduce extentand detail of mostly inaudible frequency ranges




Notes: Notes:


MPEG-Layer3 OverviewMPEG-Layer3 Overview

MP3 Compression Flow Chart




Notes: Notes:


MPEG Layer-3 performanceMPEG Layer-3 performance

14..12:1112..128kbpsstereo>15 kHzCD

16:196 kbpsstereo15 kHznear-CD

26...24:156...64 kbpsstereo11 kHzsimilar to FM radio

24:132 kbpsmono7.5 kHzbetter than AM radio

48:116 kbpsmono4.5 kHzbetter than shortwave

96:18 kbps *mono2.5 kHztelephone sound

reduction ratiobitratemodebandwidthsound quality




Notes: Notes:


MPEG-2 Advanced Audio Coding (AAC)MPEG-2 Advanced Audio Coding (AAC)

• Sampling frequencies from 8kHz to 96kHz

• 1 to 48 channels per stream

• Temporal Noise Shaping (TNS) smooths quantization noise by makingfrequency domain predictions

• Prediction: Allows predictable sound patterns such as speech to bepredicted and compressed with better quality




Notes: Notes:


MPEG-2 AAC FlowchartMPEG-2 AAC Flowchart

Input Time Signal




Notes: Notes:



• Digital Storage Media Command and Control (DSM-CC) — is the specification of a set of protocols which provides the control functions

Part 4 and 5 of MPEG-2 correspond to part 4 and 5 of MPEG-1. They have beenfinally approved in March 1996.

Part 6 of MPEG-2 - Digital Storage Media Command and Control (DSM-CC) is thespecification of a set of protocols which provides the control functions andoperations specific to managing MPEG-1 and MPEG-2 bitstreams. These protocolsmay be used to support applications in both stand-alone and heterogeneous network

environments. In the DSM-CC model, a stream is sourced by a Server and deliveredto a Client. Both the Server and the Client are considered to be Users of the DSM-CC network. DSM-CC defines a logical entity called the Session and ResourceManager (SRM) which provides a (logically) centralized management of the DSM-CC Sessions and Resources.




Notes: Notes:



• Part 9 defines the Transport Stream for MPEG2

Part 6 of MPEG-2 - Digital Storage Media Command and Control (DSM-CC) is thespecification of a set of protocols which provides the control functions andoperations specific to managing MPEG-1 and MPEG-2 bitstreams. These protocolsmay be used to support applications in both stand-alone and heterogeneous networkenvironments. In the DSM-CC model, a stream is sourced by a Server and deliveredto a Client. Both the Server and the Client are considered to be Users of the DSM-CC network. DSM-CC defines a logical entity called the Session and ResourceManager (SRM) which provides a (logically) centralized management of the DSM-CC Sessions and Resources

Part 9 of MPEG-2 is the specification of the Real-time Interface (RTI) to TransportStream decoders which may be utilised for adaptation to all appropriate networkscarrying Transport Streams




Notes: Notes:


MPEG Transport over IPMPEG Transport over IP

Digital video and audio signals are compresses into elementary streams and thenpacketised. The Packetised Elementary Streams (PES) contain both payload andheader information. Each payload contains a single frame of audio or video andbecomes part of the MPREG-2 Transport Stream. It is further sub-divided into 188-bytes packets. A packet Identifier (PID) in the header of each transport packetassociates the packet with the program channel to which it belongs usinginformation signaled in MPEG-2 PSI.

Also placed in the packets periodically are program clock reference (PCR) values toclosely synchronize the encoder and decoder clocks. Both PID and PCR areimportant measurement parameters within the transport stream. The PID identifiesthe program to which each packet belongs and also enables determination of thebandwidth allocation between programs.




Notes: Notes:


MPEG Video CompressionMPEG Video Compression

• Supports JPEG and H.261 through downward compatibility

• Supports higher Chrominance resolution and pixel resolution (720x480 is

standard used for TV signals)• Supports interlaced and noninterlaced modes

• Uses Bidirectional prediction in “Group Of Pictures” to encode differenceframes.

“Group Of Pictures ” inter-frame dependencies in a s tream

Source: “Parallelization of Software Mpeg Compression”http://www.evl.uic.edu/fwang/mpeg.html




Notes: Notes:


Picture TypesPicture Types

• MPEG-2 uses 3 different picture types

• I-Pictures - Intra pictures - these are encoded without reference to others

— They can take advantage of spatial redundancy• P-Pictures - Predictive pictures - these use a previous I-picture or P-

picture plus motion compensation

• B-Pictures - Bidirectional pictures - these can use a previous or future I-picture or P-picture for motion compensation — When a future picture is used the frames are reordered — The receiver stores the frame, uses it for constructing the new frame and

then later plays it in the correct play sequence• B-Pictures are the most complex to construct but yield the greatest

compression

• The greater use that is made of them the greater will be the receivermemory requirements

'Intra' pictures (I-pictures) are coded without reference to other pictures. Moderate compression isachieved by reducing spatial redundancy, but not temporal redundancy. They can be usedperiodically to provide access points in the bitstream where decoding can begin.'Predictive' pictures (P-pictures) can use the previous I- or P-picture for motion compensation andmay be used as a reference for further prediction. Each block in a P-picture can either be predictedor intra-coded. By reducing spatial and temporal redundancy, P-pictures offer increasedcompression compared to I-pictures.'Bidirectionally-predictive' pictures (B-pictures) can use the previous and next I- or P-pictures formotion-compensation, and offer the highest degree of compression. Each block in a B-picture can beforward, backward or bidirectionally predicted or intra-coded. To enable backward prediction froma future frame, the coder reorders the pictures from natural 'display' order to 'bitstream' order so thatthe B-picture is transmitted after the previous and next pictures it references. This introduces areordering delay dependent on the number of consecutive B-pictures.The different picture types typically occur in a repeating sequence, termed a 'Group of Pictures' orGOP. A typical GOP in display order is:

B1 B2 I3 B4 B5 P6 B7 B8 P9 B10 B11 P12The corresponding bitstream order is:

I3 B1 B2 P6 B4 B5 P9 B7 B8 P12 B10 B11For a given decoded picture quality, coding using each picture type produces a different number ofbits. In a typical example sequence, a coded I-picture was three times larger than a coded P-picture,which was itself 50% larger than a coded B-picture.




Notes: Notes:


MPEG 1 & 2 BitstreamMPEG 1 & 2 Bitstream

Source: http://www.doc.ic.ac.uk/~nd/surprise_96/journal/vol4/sab/report.html

The MPEG data hierarchy




Notes: Notes:


MPEG-2MPEG-2

• MPEG-2 encodes the active portion of the PAL frame — 720 pixels by 576 lines

• Using 8 bits for each of 8 enc0ding streams required 166 Mbit/s• First stage is to average adjacent lines reducing rate to 124 Mbit/s• Frames have spatial and temporal redundancy

— Adjacent parts of a picture are similar

— 2 dimensional blocks are encoded 8 pixels by 8 lines

The active region of a digital television frame, sampled according to CCIR recommendation 601, is720 pixels by 576 lines for a frame rate of 25 Hz. Using 8 bits for each Y, U or V pixel, theuncompressed bit rates for 4:2:2 and 4:2:0 signals are therefore:

4:2:2: 720 x 576 x 25 x 8 + 360 x 576 x 25 x ( 8 + 8 ) = 166 Mbit/s4:2:0: 720 x 576 x 25 x 8 + 360 x 288 x 25 x ( 8 + 8 ) = 124 Mbit/s

MPEG-2 is capable of compressing the bit rate of standard-definition 4:2:0 video down to about 3-15 Mbit/s. At the lower bit rates in this range, the impairments introduced by the MPEG-2 codingand decoding process become increasingly objectionable. For digital terrestrial television

broadcasting of standard-definition video, a bit rate of around 6 Mbit/s is thought to be a goodcompromise between picture quality and transmission bandwidth efficiency. A bit rate reductionsystem operates by removing redundant information from the signal at the coder prior totransmission and re-inserting it at the decoder. A coder and decoder pair are referred to as a 'codec'.In video signals, two distinct kinds of redundancy can be identified.Spatial and temporal redundancy: Pixel values are not independent, but are correlated with theirneighbours both within the same frame and across frames. So, to some extent, the value of a pixel ispredictable given the values of neighbouring pixels.Psychovisual redundancy: The human eye has a limited response to fine spatial detail, and is lesssensitive to detail near object edges or around shot-changes. Consequently, controlled impairmentsintroduced into the decoded picture by the bit rate reduction process should not be visible to ahuman observer.




Notes: Notes:


MPEG-2MPEG-2

• The pattern of values for pixels is not random im practice

• By using cosine transforms the division of frequency of change in the

pixels can be concentrated into coefficient values — The pattern of coefficients is generally concentrated in lower frequencies• Compression can be achieved by not transmitting the high frequencies

• The quantization of the encoding of coefficients is weighted — Low frequency components are encoding using more accuracy than high

frequency• The encoding uses a form of run length encoding

— Takes advantage of the high instance of zero values

For an 8x8 block of 8 bit pixels, the DCT produces an 8x8 block of 11 bit coefficients (the range ofcoefficient values is larger than the range of pixel values.) The reduction in the number of bitsfollows from the observation that, for typical blocks from natural images, the distribution ofcoefficients is non-uniform. The transform tends to concentrate the energy into the low-frequencycoefficients and many of the other coefficients are near-zero. The bit rate reduction is achieved bynot transmitting the near-zero coefficients and by quantising and coding the remaining coefficientsas described below. The non-uniform coefficient distribution is a result of the spatial redundancypresent in the original image block.Quantisation: The function of the coder is to transmit the DCT block to the decoder, in a bit rateefficient manner, so that it can perform the inverse transform to reconstruct the image. It has beenobserved that the numerical precision of the DCT coefficients may be reduced while stillmaintaining good image quality at the decoder. Quantisation is used to reduce the number ofpossible values to be transmitted, reducing the required number of bits.The degree of quantisation applied to each coefficient is weighted according to the visibility of theresulting quantisation noise to a human observer. In practice, this results in the high-frequencycoefficients being more coarsely quantised than the low-frequency coefficients. Note that thequantisation noise introduced by the coder is not reversible in the decoder, making the coding anddecoding process 'lossy'.Coding: The serialisation and coding of the quantised DCT coefficients exploits the likely clusteringof energy into the low-frequency coefficients and the frequent occurrence of zero-value coefficients.The block is scanned in a diagonal zigzag pattern starting at the DC coefficient to produce a list ofquantised coefficient values, ordered according to the scan pattern.




Notes: Notes:


MPEG-2 EncodingMPEG-2 Encoding

Length of run of zeros

Value of non-zero

coefficient

Variable-lengthCode-word

0 12 0000 0000 1101 00

0 6 0010 0001 0

1 4 0000 0011 000

0 3 0010 10

EOB - 10

12, 6, 6, 0, 4, 3, 0, 0, 0...0

(12), (6), (6), (0, 4), (3) EOB

• Example encoding of a string of coefficients:

• First step is to group them into a string of zeros followed by non-zero

Using the scan pattern common to both MPEG-1 and MPEG-2. MPEG-2 has an additional 'alternate'scan pattern intended for scanning the quantised coefficients resulting from interlaced sourcepictures.To illustrate the variable-length coding process, consider the following example list of valuesproduced by scanning the quantised coefficients from a transformed block:

12, 6, 6, 0, 4, 3, 0, 0, 0...0The first step is to group the values into runs of (zero or more) zeros followed by a non-zero value.Additionally, the final run of zeros is replaced with an end of block (EOB) marker. Using

parentheses to show the groups, this gives:(12), (6), (6), (0, 4), (3) EOB

The second step is to generate the variable length code words corresponding to each group (a run ofzeros followed by a non-zero value) and the EOB marker. Table 1 shows an extract of the DCTcoefficient VLC table common to both MPEG-1 and MPEG-2. MPEG-2 has an additional 'intra'VLC optimised for coding intra blocks (see Section 4). Using the variable length code from Tableand adding spaces and commas for readability, the final coded representation of the example blockis:

0000 0000 1101 00, 0010 0001 0, 0010 0001 0, 0000 0011 000, 0010 10, 10




Notes: Notes:


MPEG-2 Motion CompensationMPEG-2 Motion Compensation

• Temporal redundancy is exploited by predicting each frame

• An earlier reference frame is used and compared with the current

• The decoded pictures are not identical to the source — Small distortions result from the compression encoding — The source therefore constructs a local decode and uses this for reference

• Blocks of picture are matched between reference and new frame

• The 'best' offset is selected on the basis of minimum error between theblock being coded and the prediction

This technique exploits temporal redundancy by attempting to predict the frame to be coded from aprevious 'reference' frame. The prediction cannot be based on a source picture because theprediction has to be repeatable in the decoder, where the source pictures are not available (thedecoded pictures are not identical to the source pictures because the bit rate reduction processintroduces small distortions into the decoded picture.) Consequently, the coder contains a localdecoder which reconstructs pictures exactly as they would be in the decoder, from which predictionscan be formed.The simplest inter-frame prediction of the block being coded is that which takes the co-sited (i.e. thesame spatial position) block from the reference picture. Naturally this makes a good prediction forstationary regions of the image, but is poor in moving areas. A more sophisticated method, known asmotion-compensated inter-frame prediction, is to offset any translational motion which has occurredbetween the block being coded and the reference frame and to use a shifted block from the referenceframe as the prediction.One method of determining the motion that has occurred between the block being coded and thereference frame is a 'block-matching' search in which a large number of trial offsets are tested by thecoder using the luminance component of the picture. The 'best' offset is selected on the basis ofminimum error between the block being coded and the prediction.The bit rate overhead of using motion-compensated prediction is the need to convey the motionvectors required to predict each block to the decoder. For example, using MPEG-2 to compressstandard-definition video to 6 Mbit/s, the motion vector overhead could account for about 2 Mbit/sduring a picture making heavy use of motion-compensated prediction.




Notes: Notes:


MPEG-2: Profiles and LevelsMPEG-2: Profiles and Levels

8, 4, 44, 3Bitrate352 X 288/30Lower

352 X 288/30352 X 288/30Enhancement

Low

25, 10, 1520, 15, 415, 10Bitrate

720 X 576/30352 X 288/30Lower

720 X 576/30720 X 576/30720 X 576/30Enhancement

Main

100, 40, 6080, 60, 2060, 40, 15Bitrate

1920 X 1152/60720 X 576/30720 X 576/30Lower

1920 X 1152/601440 X 1152/601440 X 1152/60Enhancement

High-1440

130, 50, 80100, 80,25Bitrate

1920 X 1151/60960 X 576/30Lower

1920 X 1151/601920 X 1151/60Enhancement

High

Multiview

4:2:0

High

4:2:0;4:2:2

Spatial

4:2:0

SNR

4:2:0

Profiles

Levels




Notes: Notes:


Pro MPEG For Error TolerancePro MPEG For Error Tolerance

• Adding Forward Error recovery to MPEG protocols improves quality

• During transmission on satellite and terrestrial broadcast errors occur

• Over IP networks packets of frames are lost when errors occur — On copper cables error rates of 1 bit in 109 are typical — On fiber cables error rates of 1 bit in 1012 are expected — Gigabit Ethernet frames are 1500 bytes or 12,000 bits long

– Even over fiber this results in 12 errors per second on average!!• Codes of Practice define how to add FEC to MPEG streams

Professional video streaming over IP networks has become a reality and this paperprovides an insight into the workings of Pro-MPEG CoP#3 Forward ErrorCorrection (FEC) in protecting contribution and distribution services. Intransporting real-time media over IP networks either UDP or RTP (Real TimeProtocol) protocols can be used. RTP provides packet sequence ordering over UDP,enabling a receiver to identify out of sequence, discarded or reordered packets so ismore robust than UDP. The Pro-MPEG CoP #3 FEC scheme uses the RTP transportprotocol as a building block for providing packet recovery techniques to ensurereliable real-time media transport. The MPEG Transport Stream (TS) packets mustfirst be packed into IP frames. Since most streams will pass over an Ethernetnetwork at some point, whose MTU (Maximum Transmission Unit) is 1500, the IPframe must be constrained so that fragmentation does not occur. This limits thenumber of TS packets to a maximum of 7 per IP packet. Packing the maximum ofseven 188 byte packets into an IP packet gives optimum packing efficiency at thecost of excessive data loss per a packet. If only a single MPEG packet is placed inan IP packet minimal loss of data occurs on packet loss, at the cost of highertransmission overheads, which in turn will place greater demand on the network.




Notes: Notes:


Pro-MPEG CoP3 Forward Error Correction (FEC)Pro-MPEG CoP3 Forward Error Correction (FEC)

Normal MPEG-2 Transport streams packseveral video frames in an Ethernet Transfer

Pro-MPEG takes consecutivepackets and adds FEC

The generation of the FEC packets is based on the use of a matrix. The matrix sizeis defined by two parameters L and D, L is the spacing between non-consecutivepackets to be used to calculate the FEC packet and D is the depth of the matrix.

Column FEC provides correction for consecutive burst packet loss of up to Lpackets. The FEC packets are generated per a column within the matrix allowingloss of any single media packet within a column or burst of error within a row to be

corrected through the FEC packet. Column FEC is ideal for correcting packet bursterrors and random errors.

Row FEC provides correction of non-consecutive packet loss and can correct anysingle packet loss within a row of media packets. The FEC packets are generatedper a row allowing loss of any single packet to be recovered. Row FEC is ideal forcorrecting random packet errors.

Column FEC is often called 1D FEC due to the FEC only being calculated on 1dimension where Column and Row FEC is referred to as 2D FEC due to the FECbeing calculated on 2 dimensions, column and row.






Notes: Notes:


COP 4 for SMPTE 292MCOP 4 for SMPTE 292M

• RFC3497, RTP Payload Format

• Society of Motion Picture & Television Engineers (SMPTE) 292M Video

• Commercial Code of Practice issued by the Pro-MPEG forum

• Used to carry uncompressed HDTV

The serial digital interface, SMPTE 292M , defines a universal medium ofinterchange for uncompressed High Definition Television (HDTV) between varioustypes of video equipment (cameras, encoders, VTRs, etc.). SMPTE 292M stipulatesthat the source data be in 10 bit words and the total data rate be either 1.485 Gbpsor 1.485/1.001 Gbps. The use of a dedicated serial interconnect is appropriate in astudio environment, but it is desirable to leverage the widespread availability ofhigh bandwidth IP connectivity to allow efficient wide area delivery of SMPTE292M content.

This RFC only addresses the transfer of uncompressed HDTV. Compressed HDTVis a subset of MPEG-2 , which is fully described in document A/53 of theAdvanced Television Standards Committee. The ATSC has also adopted theMPEG-2 transport system (ISO/IEC 13818-1) . Therefore RFC 2250 sufficientlydescribes transport for compressed HDTV over RTP.




Notes: Notes:


Versions of MPEG4Versions of MPEG4

MPEG-4 Version 1 was approved by MPEG in December 1998; version 2 wasfrozen in December 1999. After these two major versions, more tools were added insubsequent amendments that could be qualified as versions, even though they areharder to recognize as such. Recognizing the versions is not too important,however; it is more important to distinguish Profiles. Existing tools and profilesfrom any version are never replaced in subsequent versions; technology is alwaysadded to MPEG?4 in the form of new profiles. Figure 3 below depicts therelationship between the versions. Version 2 is a backward compatible extension ofVersion 1, and version 3 is a backward compatible extension of Version 2 – and soon. The versions of all major parts of the MPEG-4 Standard (Systems, Audio,Video, DMIF) were synchronized; after that, the different parts took their ownpaths.

The Systems layer of Version later versions is backward compatible with all earlierversions. In the area of Systems, Audio and Visual, new versions add Profiles, donot change existing ones. In fact, it is very important to note that existing systemswill always remain compliant, because Profiles will never be changed in retrospect,and neither will the Systems Syntax, at least not in a backward-incompatible way.




Notes: Notes:


MPEG4MPEG4

Media objects may need streaming data, which is conveyed in one or more elementary streams. An object descriptor identifiesall streams associated to one media object. This allows handling hierarchically encoded data as well as the association ofmeta-information about the content (called ‘object content information’) and the intellectual property rights associated with it.Each stream itself i s characterized by a set of descriptors for configuration information, e.g., to determine the required decoderresources and the precision of encoded timing information. Furthermore the descriptors may carry hints to the Quality ofService (QoS) it requests for transmission (e.g., maximum bit rate, bit error rate, priority, etc.)Synchronization of elementary streams is achieved through time stamping of individual access units within elementarystreams. The synchronization layer manages the identification of such access units and the time stamping. Independent of themedia type, this layer allows identification of the type of access unit (e.g., video or audio frames, scene descriptioncommands) in elementary streams, recovery of the media object’s or scene description’s time base, and it enablessynchronization among them. The syntax of this layer is configurable in a large number of ways, allowing use in a broadspectrum of systems.

The synchronized delivery of streaming information from source to destination, exploiting different QoS as available from thenetwork, is specified in terms of the synchronization layer and a delivery layer containing a two-layer multiplexer.The first multiplexing layer is managed according to the DMIF specification, part 6 of the MPEG?4 standard. (DMIF standsfor Delivery Multimedia Integration Framework) This multiplex may be embodied by the MPEG-defined FlexMux tool,which allows grouping of Elementary Streams (ESs) with a low multiplexing overhead. Multiplexing at this layer may beused, for example, to group ES with similar QoS requirements, reduce the number of network connections or the end to enddelay.The “TransMux” (Transport Multiplexing) layer models the layer that offers transport services matching the requested QoS.Only the interface to this layer is specified by MPEG-4 while the concrete mapping of the data packets and control signalingmust be done in collaboration with the bodies that have jurisdiction over the respective transport protocol. Any suitableexisting transport protocol stack such as (RTP)/UDP/IP, (AAL5)/ATM, or MPEG-2’s Transport Stream over a suitable linklayer may become a specific TransMux instance. The choice is left to the end user/service provider, and allows MPEG-4 to beused in a wide variety of operation environments.




Notes: Notes:


MPEG-4 Video EncodingMPEG-4 Video Encoding

• MPEG-4 includes tools for:- — Efficient compression of images and video — Efficient compression of textures for texture mapping on 2-D and 3-D

meshes — Efficient compression of implicit 2-D meshes — Efficient compression of time-varying geometry streams that animate

meshes — Efficient random access to all types of visual objects — Extended manipulation functionality for images and video sequences — Content-based coding of images and video — Content-based scalability of textures, images and video — Spatial, temporal and quality scalability — Error robustness and resilience in error prone environments

The tools for representing natural video in the MPEG-4 visual standard provide standardized coretechnologies allowing efficient storage, transmission and manipulation of textures, images and videodata for multimedia environments. These tools allow the decoding and representation of atomic unitsof image and video content, called “video objects” (VOs). An example of a VO could be a talkingperson (without background), which can then be composed with other AVOs (audio-visual objects)to create a scene. Conventional rectangular imagery is handled as a special case of such objects.In order to achieve this broad goal rather than a solution for a narrow set of applications,functionalities common to several applications are clustered. Therefore, the visual part of theMPEG-4 standard provides solutions in the form of tools and algorithms for:Efficient compression of images and videoEfficient compression of textures for texture mapping on 2-D and 3-D meshesEfficient compression of implicit 2-D meshesEfficient compression of time-varying geometry streams that animate meshesEfficient random access to all types of visual objectsExtended manipulation functionality for images and video sequencesContent-based coding of images and videoContent-based scalability of textures, images and videoSpatial, temporal and quality scalabilityError robustness and resilience in error prone environments




Notes: Notes:


MPEG-4 Video EncodingMPEG-4 Video Encoding

• Scalable Coding of Video Objects

• Robustness in Error Prone Environments

• Resynchronization

• Data Recovery

• Error Concealment

• Fast recovery in real-time coding

• Improved temporal resolution stability with low buffering delay — A special technique of use in real-time encoding situations is Dynamic

Resolution Conversion (DRC) — It is a way to stabilize the transmission buffering delay

The coding of conventional images and video is similar to conventional MPEG-1/2 coding. Itinvolves motion prediction/compensation followed by texture coding. For the content-basedfunctionalities, where the image sequence input may be of arbitrary shape and location, thisapproach is extended by also coding shape and transparency information. Shape may be eitherrepresented by an 8 bit transparency component - which allows the description of transparency ifone VO is composed with other objects - or by a binary mask.The extended MPEG-4 content-based approach can be seen as a logical extension of theconventional MPEG-4 VLBV Core or high bit-rate tools towards input of arbitrary shape.There are several scalable coding schemes in MPEG-4 Visual: spatial scalability, temporalscalability, fine granularity scalability and object-based spatial scalability. Spatial scalabilitysupports changing the spatial resolution. Object-based spatial scalability extends the 'conventional'types of scalability towards arbitrary shape objects, so that it can be used in conjunction with otherobject-based capabilities. Thus, a very flexible content-based scaling of video information can beachieved. This makes it possible to enhance SNR, spatial resolution, shape accuracy, etc, only forobjects of interest or for a particular region, which can be done dynamically at play-time. Finegranularity scalability (FGS) was developed in response to the growing need on a video codingstandard for streaming video over the Internet. FGS and its combination with temporal scalabilityaddresses a variety of challenging problems in delivering video over the Internet. FGS allows thecontent creator to code a video sequence once and to be delivered through channels with a widerange of bitrates. It provides the best user experience under varying channel conditions. Itovercomes the “digital cutoff” problem associated with digital video. In other words, it makescompressed digital video behave similarly to analog video in terms of robustness while maintainingall the advantages of digital video.




Notes: Notes:


VLBV: Very Low Bit-rate VideoVLBV: Very Low Bit-rate Video

MPEG-4 provides error robustness and resilience to allow accessing image or video informationover a wide range of storage and transmission media. In particular, due to the rapid growth of mobilecommunications, it is extremely important that access is available to audio and video information viawireless networks. This implies a need for useful operation of audio and video compressionalgorithms in error-prone environments at low bit-rates (i.e., less than 64 kbit/s).The error resilience tools developed for MPEG-4 can be divided into three major areas:resynchronization, data recovery, and error concealment. It should be noted that these categories arenot unique to MPEG-4, but instead have been used by many researchers working in the area errorresilience for video. It is, however, the tools contained in these categories that are of interest, andwhere MPEG-4 makes its contribution to the problem of error resilience.After synchronization has been reestablished, data recovery tools attempt to recover data that ingeneral would be lost. These tools are not simply error correcting codes, but instead techniques thatencode the data in an error resilient manner. For instance, one particular tool that has been endorsedby the Video Group is Reversible Variable Length Codes (RVLC). In this approach, the variablelength codewords are designed such that they can be read both in the forward as well as the reversedirection.An example illustrating the use of a RVLC is given in Figure 19. Generally, in a situation such asthis, where a burst of errors has corrupted a portion of the data, all data between the twosynchronization points would be lost. However, as shown in the Figure, an RVLC enables some ofthat data to be recovered. It should be noted that the parameters, QP and HEC shown in the Figure,represent the fields reserved in the video packet header for the quantization parameter and theheader extension code, respectively.




Notes: Notes:


Example of Sprite Coding of Video SequenceExample of Sprite Coding of Video Sequence

• The sprite panoramic background image is constructed and sent — This is held in a sprite buffer

• Foreground object is transmitted separately as an arbitrary-shape videoobject

An important advantage of the content-based coding approach MPEG-4 is that the compression efficiency can be significantlyimproved for some video sequences by using appropriate and dedicated object-based motion prediction “tools” for each objectin a scene. A number of motion prediction techniques can be used to allow efficient coding and flexible presentation of theobjects:Standard 8x8 or 16x16 pixel block-based motion estimation and compensation, with up to ¼ pel accuracyGlobal Motion Compensation (GMC) for video objects: Encoding of the global motion for a object using a small number ofparameters. GMC is based on global motion estimation, image warping, motion trajectory coding, and texture coding forprediction errors.Global motion compensation based for static “sprites”. A static sprite is a possibly large sti ll image, describing panoramicbackground. For each consecutive image in a sequence, only 8 global motion parameters describing camera motion are codedto reconstruct the object. These parameters represent the appropriate affine transform of the sprite transmitted in the firstframe.Quarter Pel Motion Compensation enhances the precision of the motion compensation scheme, at the cost of only smallsyntactical and computational overhead. A accurate motion description leads to a smaller prediction error and, hence, to bettervisual quality.Shape-adaptive DCT: In the area of texture coding, the shape-adaptive DCT (SA-DCT) improves the coding efficiency ofarbitrary shaped objects. The SA-DCT algorithm is based on predefined orthonormal sets of one-dimensional DCT basisfunctions.This slidevdepicts the basic concept for coding an MPEG-4 video sequence using a sprite panorama image. It is assumed thatthe foreground object (tennis player, image top right) can be segmented from the background and that the sprite panoramaimage can be extracted from the sequence prior to coding. (A sprite panorama is a still image that describes as a static imagethe content of the background over all frames in the sequence). The large panorama sprite image is transmitted to the receiveronly once as first frame of the sequence to describe the background – the sprite remains is stored in a sprite buffer. In eachconsecutive frame only the camera parameters relevant for the background are transmitted to the receiver. This allows thereceiver to reconstruct the background image for each frame in the sequence based on the sprite. The moving foregroundobject is transmitted separately as an arbitrary-shape video object. The receiver composes both the foreground and backgroundimages to reconstruct each frame (bottom picture in figure below). For low delay applications it i s possible to transmit thesprite in multiple smaller pieces over consecutive frames or to build up the sprite at the decoder progressively.




Notes: Notes:


MPEG-4 SoundMPEG-4 Sound

• Speech coding at bitrates between 2 and 24 kbit/s is supported by usingHarmonic Vector eXcitation Coding (HVXC) for bit rates 2- 4 kbit/s

•

Code Excited Linear Predictive (CELP) coding for an operating bit rate of 4- 24 kbit/s

• For general audio coding at and above 6 kbit/s, transform codingtechniques is used

MPEG-4 standardizes natural audio coding at bitrates ranging from 2 kbit/s up to and above 64kbit/s. When variable rate coding is allowed, coding at less than 2 kbit/s, such as an average bitrateof 1.2 kbit/s, is also supported. The presence of the MPEG-2 AAC standard within the MPEG-4 toolset provides for general compression of audio in the upper bitrate range. For these, the MPEG-4standard defines the bitstream syntax and the decoding processes in terms of a set of tools. In orderto achieve the highest audio quality within the full range of bitrates and at the same time provide theextra functionalities, speech coding techniques and general audio coding techniques are integrated ina common framework:· Speech coding at bitrates between 2 and 24 kbit/s is supported by using Harmonic VectoreXcitation Coding (HVXC) for a recommended operating bitrate of 2 - 4 kbit/s, and Code ExcitedLinear Predictive (CELP) coding for an operating bitrate of 4 - 24 kbit/s. In addition, HVXC canoperate down to an average of around 1.2 kbit/s in its variable bitrate mode. In CELP coding, twosampling rates, 8 and 16 kHz, are used to support narrowband and wideband speech, respectively.The following operating modes have been subject to verification testing: HVXC at 2 and 4 kbit/s,narrowband CELP at 6, 8.3, and 12 kbit/s, and wideband CELP at 18 kbit/s. In addition various ofthe scalable configurations have been verified.· For general audio coding at bitrates at and above 6 kbit/s, transform coding techniques,namely TwinVQ and AAC, are applied. The audio signals in this region typically have samplingfrequencies starting at 8 kHz.To allow optimum coverage of the bitrates and to allow for bitrate and bandwidth scalability, ageneral framework has been defined.






Notes: Notes:


Carriage of MPEG-4 over IPCarriage of MPEG-4 over IP

• MPEG-4 can be carried over RTP/UDP/IP — UDP Delivers multiplexing by using port numbers — RTP adds a time stamp and sequence number to provide synchronization — IP provides addressing and routing

The specifications on the carriage of MPEG-4 contents over IP networks are developed jointly withIETF AVT working group. These include a framework and several RTP payload formatspecifications. The framework is standardized as both a part 8 of MPEG-4, i.e. ISO/IEC 14496-8and informative RFC in IETF. RTP payload format specifications are only standardized as astandard track RFC in IETF.

Framework is an umbrella specification for the carriage and operation of MPEG-4 sessions over IP-based protocols, including RTP, RTSP, and HTTP, among others. It provides a framework for thecarriage of MPEG-4 contents over IP networks and guidelines for designing payload formatspecifications for the detailed mapping of MPEG-4 content into several IP-based protocols. Toassure compatibility between different RTP payload formats, framework defines a conformancepoint as illustrated in the Figure 1. To conform this framework all the payload formats shall providenormative mapping functions to reconstruct logical MPEG-4 SL packets. Framework also definesthe standard MIME types associated with MPEG-4 contents.

Several RTP payload formats are developed under this framework including generic payload formatand FlexMux payload format. Generic RTP payload format specify a homogeneous carriage ofvarious MPEG-4 streams. It defines a simple but efficient mapping between logical MPEG-4 SLpackets and RTP packets. It also supports concatenation of multiple SL packets into one RTPpackets to minimize overheads. FlexMux payload format specifies a carriage of FlexMux packetizedstreams via RTP packets. It includes a payload formats to convey FlexMux descriptors todynamically signal the configuration of FlexMux.




Notes: Notes:


H.264 and MPEG-4 Part 10H.264 and MPEG-4 Part 10

• H.264 Advanced Video Coding (AVC)

• The following terms are used interchangeably:

– H.26L – The Work of the JVT or “JVT CODEC” – JM2.x, JM3.x, JM4.x – The Thing Beyond H.26L – The “AVC” or Advanced Video CODEC

• Proper Terminology going forward: – MPEG-4 Part 10 (Official MPEG Term)

» ISO/IEC 14496-10 AVC – H.264 (Official ITU Term)




Notes: Notes:


The JVT ProjectThe JVT Project

• Started as ITU-T Q.6/SG16 (VCEG - Video Coding Experts Group) “H.26L”standardization activity

•

August 1999: 1st test model (TML-1)• July 2001: MPEG open call for “AVC” technology: H.26L wins

• December 2001: Formation of the Joint Video Team (JVT) between VCEGand MPEG to finalize H.26L as a joint project similar to MPEG-2/H.262)

• July 2002: Final Committee Draft status in MPEG

• Final approval completed in both orgs 2003

After finalising the original H.263 standard for videotelephony in 1995, the ITU-TVideo Coding Experts Group (VCEG) started work on two further developmentareas: a “short-term” effort to add extra features to H.263 (resulting in Version 2 ofthe standard) and a “long-term” effort to develop a new standard for low bitratevisual communications. The long-term effort led to the draft “H.26L” standard,offering significantly better video compression efficiency than previous ITU-Tstandards. In 2001, the ISO Motion Picture Experts Group (MPEG) recognised thepotential benefits of H.26L and the Joint Video Team (JVT) was formed, includingexperts from MPEG and VCEG. JVT’s main task is to develop the draft H.26L“model” into a full International Standard. In fact, the outcome will be twoidentical) standards: ISO MPEG4 Part 10 of MPEG4 and ITU-T H.264. The“official” title of the new standard is Advanced Video Coding (AVC); however, itis widely known by its old working title, H.26L and by its ITU document number,H.264.




Notes: Notes:


Relationship to Other StandardsRelationship to Other Standards

• Same design to be approved in both ITU-T and MPEG

• In ITU-T this will is new & separate standard

— ITU-T Recommendation H.264 — ITU-T systems (H.32x) to be modified to support it• In MPEG this will be a new “part” in the MPEG-4 suite

— Separate codec design from prior MPEG-4 visual — New part 10 called “advanced video coding” (similar to “AAC” position in

MPEG-2 as separate codec) — Not backward compatible with prior standards (incl. the prior MPEG-4 visual

spec. – core technology is different) — MPEG-4 Systems / File Format modifying to support it

• H.222.0 | MPEG-2 Systems will also be modified to support it

• IETF working on RTP payload packetization:

— RTP Payload Format for H.264 Video RFC 3984




Notes: Notes:


JVT Project Technical ObjectivesJVT Project Technical Objectives

• Primary technical objectives: — Significant improvement in coding efficiency: Average bit rate reduction of

50% given fixed fidelity compared to any other video standard — Error robustness & “Network Friendliness” (carry it well on MPEG-2 or RTP

Issues examined in H.263 and MPEG-4 are further improved — Low latency capability (better quality for higher latency) — Exact match decoding — Simple syntax specification

– Targeting simple and clean solutions – Avoiding any excessive quantity of optional features or profile

configurations




Notes: Notes:


0

1

2

3

4

5

6

1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005

M b i t / s

MPEG-2

1stMPEG-2 Encoder

2nd Generation

Encoder

3 rd GenerationEncoder

4th GenerationEncoder

MPEG-2Standard

Frozen(H.262)


Need for H.264: MPEG-2 Has Hit A WallNeed for H.264: MPEG-2 Has Hit A Wall




Notes: Notes:


0

1

2

3

4

5

6

1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005

M b i t / s

MPEG-2

MPEG-4

H.263

1stMPEG-2 Encoder

2nd GenerationEncoder




MPEG-4 in ComparisonMPEG-4 in Comparison




Notes: Notes:


0

1

2

3

4

5

6

1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005

M b i t / s

MPEG-2

MPEG-4

H.26L

H.263

1stMPEG-2 Encoder




5 th GenerationEncoder

H.26L Provides FocusH.26L Provides Focus




Notes: Notes:


0

1

2

3

4

5

6

1994 1995 1996 1997 1998 1999 2000 2001 2002 2003 2004 2005

M b i t / s

MPEG-2

MPEG-4

H.26L

H.263

1stMPEG-2 Encoder




H.264 /MPEG-4 part 10

5 th GenerationEncoder

MPEG-4 “Adopts” H.26LMPEG-4 “Adopts” H.26L








Notes: Notes:


“2-D mesh modeling of video object - By deforming the mesh, the fish can be animatedvery efficiently, and be made to swim. Also, a logo could be projected onto the fish, andmade to move in accordance with the fish”

More MPEG-4 Solutions in Search of ProblemsMore MPEG-4 Solutions in Search of Problems




Notes: Notes:


3.2quality18 kb/s1AAC

3.6quality6 kb/s base,18 kb/s enh.

1Scalable: Twin VQ base andAAC enhancement

3.7quality6 kb/s base,18 kb/s enh.

1Scalable: CELP base andAAC enhancement

4.2quality24 kb/s1AAC

4.1impairment128 kb/s2MPEG-1 Layer III

4.3impairment192 kb/s2MPEG-1 Layer II

4.4impairment96 kb/s2AAC


4.6impairment640 kb/s51995 Backward CompatibleMPEG-2 Layer II


Subjectivequality

Grading scaleTotal bitrate

ChannelsCoding tool

2.8quality6.3 kb/s1G.723

MPEG-4 Audio ToolsMPEG-4 Audio Tools




Notes: Notes:


MPEG-4 Audio ToolsMPEG-4 Audio Tools

1.8quality6 kb/s1HILN

2.8quality16 kb/s1HILN

1.8quality6 kb/s1Twin VQ

2.5quality6 kb/s1Narrowband CELP

3.4quality32 kb/s1AAC Low Delay

4.2quality64 kb/s1G.722

4.4quality64 kb/s1AAC Low Delay

3.0quality64 kb/s2BSAC



2.3quality18.2 kb/s1Wideband CELP

Subjectivequality

Grading scaleTotal bitrate

ChannelsCoding tool




Notes: Notes:


Applications for AVC/H.264Applications for AVC/H.264

• Entertainment Video (1 - 8+ Mbps, higher latency) — Broadcast / Satellite / Cable / DVD / VoD / FS-VDSL / …

• Conversational H.32X Services (usu. <1Mbps, low latency) — H.320 Conversational — 3GPP Conversational H.324/M — H.323 Conversational Internet/unmanaged/best effort IP/RTP — 3GPP Conversational IP/RTP/SIP

• Streaming Services (usu. lower bit rate, higher latency) — 3GPP Streaming IP/RTP/RTSP — Streaming IP/RTP/RTSP (without TCP fallback)

• Other Services — 3GPP Multimedia Messaging Services






Notes: Notes:


CODECSystemPlatform& CPU

MotionImagerySources

Video Sensors,SMPTE Streams,Networked Video,

Video Servers

BitstreamOutput

V

MD

VLSI Chip orChips

Encoder System GUI

DGFX PVAFrequencyShaping &

NoiseReduction

DGFX PVAHi QualityChroma

Processing

Logical System Interconnect (HW or SW)

NAL: MPEG-2 or Other Xport

AVCEncoder

W/ExactMatch

AVC’s Network Adaptation Layer (NAL)Supports a Range of Xport Layer Formats &

Protocols (Similar to SMPTE Abstraction)

10 (& 12) BitEnabled

AVC EncoderAVC Encoder




Notes: Notes:


2728293031323334353637

3839

0 50 100 150 200 250Bit-rate [kbit/s]

Foreman QCIF 10Hz

QualityY-PSNR [dB]

MPEG-2H.263

MPEG-4JVT/H.264/AVC

Comparison to MPEG-2, H.263, MPEG-4Comparison to MPEG-2, H.263, MPEG-4




Notes: Notes:


CIF 30Hz

252627282930313233343536

3738

0 500 1000 1500 2000 2500 3000 3500Bit-rate [kbit/s]

QualityY-PSNR [dB]

MPEG-2H.263

MPEG-4JVT/H.264/AVC

Comparison to MPEG-2, H.263, MPEG-4Comparison to MPEG-2, H.263, MPEG-4

Peek Signalto Noise Ratio




Notes: Notes:


CODEC ComparisonCODEC Comparison

0.10.81.4High detail,

measurable

6-Test Signals

1.22.59.5Slow motion,high detail

5-Oscars ’02

0.92.9Motion, highdetail

4-PhilipsLiquids

23.330.870.0Randommotion

3-Preakness

2.24.210.1Skin tones,water

2-PanasonicGirls

4.67.326.7Fast motion,high detail

1-CollegeFootball

AVC-2AVC-1MPEG-2AttributesSequence

(Normalized speed In Mbps)




Notes: Notes:


AVC/H.264 ProfilesAVC/H.264 Profiles

• High Profile — Highest compression or video quality at a given bit rate — Suitable for good quality entertainment video distribution

• Baseline Profile — Least complexity — Error resilient — Suitable for telephony, conferencing application




Notes: Notes:


AVC/H.264 – LevelsAVC/H.264 – Levels

• Level 1.0: QCIF @ 15frames/sec• Level 1.1: QCIF @ 30 frames/sec, CIF @7.5 frames/sec• Level 1.2: CIF @ 15 frames/sec• Level 2.0: CIF @ 30 frames/sec• Level 2.1: HHR @ 25 or 30 frames/sec• Level 2.2: SDTV @ 15 frames/sec• Level 3.0: SDTV: 720x480x30i, 720x576x25i

— 10 Mbps (max.), up to 5 (max. resolution) reference frames• Level 3.1: HDTV - 1280x720x30p, SVGA (800x600) 50+p• Level 3.2: HDTV - 1280x720x60p• Level 4.0: HDTV (all formats) - 1920x1080x30i, 1280x720x60p, 2kx1kx30p

— 20 Mbps (max.), up to 4 (max. resolution) reference frames• Level 4.1: HDTV - 1920x1080x30i, 1280x720x60p, 2kx1kx30p

— 50 Mbps, up to 4 (max. resolution) reference frames• Level 4.2: S-HDTV - 1920x1080x60p• Level 5.0: S-HDTV/D-Cinema – 2kx1kx72p• Level 5.1: S-HDTV/D-Cinema – 2kx1kx120p, 4kx2kx24p, 4kx2kx30p




Notes: Notes:


Transport of AVC / H.264Transport of AVC / H.264

• Transport of MPEG-4 AVC using MPEG-2 System: ISO/IEC 13818-1 — PDAM (Proposed Draft AMendment) in May 2002 — FPDAM (Final Proposed Draft AMendment) in Dec 2002 — FDAM in July 2003 — Approved AMD

• IP delivery — MPEG-2 TS over UDP/IP, or — RTP over IP




Notes: Notes:


DVB ProjectDVB Project

• ETSI TR 101 200 — Digital Video Broadcasting (DVB);A guideline for the use of DVB

specifications and standards — Originally from DVB.org Blue Books

• Included different delivery technologies originally — Satellite (DVB-S and DVB-S2) — Cable (DVB-C) — Terrestrial television (DVB-T) — Terrestrial television for handhelds (DVB-H)

• Later DVB-IPI added

DVB, short for Digital Video Broadcasting, is a suite of internationally accepted,open standards for digital television maintained by the DVB Project, an industryconsortium with more than 270 members, and published by a Joint TechnicalCommittee (JTC) of European Telecommunications Standards Institute (ETSI),European Committee for Electrotechnical Standardization (CENELEC) andEuropean Broadcasting Union (EBU).

How the several DVB sub-standards interact is described in the DVB Cookbookfrom DVB.org.

One of the fundamental decisions which were taken during the early days of theDVB Project was the selection of MPEG-2 for the source coding of audio and videoand for the creation of programme elementary streams, Transport Streams (TS),etc.; the so-called Systems level. The ISO/IEC 13818 Parts 1, 2, 3 [60], [61], [62]are international standards which describe MPEG-2 Systems, Video and Audio. Allthree are truly generic and can be considered too wide in scope for them to beapplied to DVB directly.




Notes: Notes:


DVB StandardsDVB Standards

• DVB.org

• Industry Group

• Developed DVB standards

• Blue Books

• Have become ETSI

From time to time, DVB publishes documents approved by its Steering Board: theBlueBooks. In practice, these are either commercial requirements documents,policy statements, or technical specifications which are being standardised. In thelatter case, DVB has decided that there is value the in rapid publication of draftspecifications as BlueBooks, pending their formal standardisation.

The BlueBooks available on the DVB.org site are those that have not since beensuperceded by a published standard. All DVB standards, specifications and reportscan be downloaded free of charge from the ETSI website. A086 (DVB-IPI) is notlisted but can be found using a search athttp://www.dvb.org/technology/standards_specifications/internet_protocol/dvbipi/

or as ETSI standard TS 102 034.




Notes: Notes:


DVB-C Standard Over CableDVB-C Standard Over Cable

• DVB-C is Digital video Broadcasting over Cable

• Standards for Cable systems are available at http://www.cablelabs.org

DVB-C is the digital Video broadcast standard for cable operation. Source coding and MPEG-2multiplexing (MUX): video, audio, and data streams are multiplexed into an MPEG-2 PS (MPEG-2Programme Stream). One or more PSs are joined together into a MPEG-2 TS (MPEG-2 TransportStream); this is the basic digital stream which is being transmitted and received by home Set TopBoxes (STB). Allowed bitrates for the transported MPEG-2 depend on a number of modulationparameters: it can range from about 6 to about 64 Mbit/s (see the bottom figure for a completelisting). MUX adaptation and energy dispersal: the MPEG-2 TS is identified as a sequence of datapackets, of fixed length (188 bytes). With a technique called energy dispersal, the byte sequence is

decorrelated. External encoder: a first level of protection is applied to the transmitted data, using anonbinary block code, a Reed-Solomon RS (204, 188) code, allowing the correction of up to amaximum of 8 wrong bytes for each 188-byte packet. External interleaver: convolutionalinterleaving is used to rearrange the transmitted data sequence, such way it becomes more rugged tolong sequences of errors. Byte/m-tuple conversion: data bytes are encoded into bit m-tuples (m = 4,5, 6, 7, or 8). Differential coding: the two most significant bytes in each m-tuple are encoded inorder to give some ruggedness to the signal. QAM Mapper: the bit sequence is mapped into a base-band digital sequence of complex symbols. There are 5 allowed modulation modes: 16-QAM, 32-QAM, 64-QAM, 128-QAM, 256-QAM. Base-band shaping: the QAM signal is filtered with araised-cosine shaped filter, in order to remove mutual signal interference at the receiving side. DACand front-end: the digital signal is transformed into an analog signal, with a digital-to-analogconverter (DAC), and then modulated to radio frequency by the RF front-end.




Notes: Notes:


DVB-HDVB-H

• Digital Video Broadcast for Handhelds — ETSI EN 302 304 V1.1.1 (2004-11)

– Digital Video Broadcasting (DVB);Transmission System for Handheld Terminals(DVB-H) — ETSI TS 102 472 V1.1.1 (2006-06)

– Digital Video Broadcasting (DVB);IP Datacast over DVB-H: Content DeliveryProtocols

DVB-H (Digital Video Broadcasting - Handheld) is a technical specification forbringing broadcast services to handheld receivers. was formally adopted as ETSIstandard EN 302 304 in November 2004. The DVB-H specification (EN 302 304)can be downloaded from the DVB-H Online website[1]. The major competitor ofthis technology is Digital Multimedia Broadcasting (DMB).The conceptual structure of a DVB-H receiver is depicted here. It includes a DVB-H demodulator and a DVB-H terminal. The DVB-H demodulator includes a DVB-

T demodulator, a time-slicing module and a MPE-FEC module. The DVB-Tdemodulator recovers the MPEG-2 Transport Stream packets from the receivedDVB-T (EN 300 744 [1]) RF signal. It offers three transmission modes 8K, 4K and2K with the corresponding Transmitter Parameter Signalling (TPS). Note that the4K mode, the in-depth interleavers and the DVB-H signalling have been definedwhile elaborating the DVB-H standard. The time-slicing module, provided byDVB-H, aims to save receiver power consumption while enabling to performsmooth and seamless frequency handhover. The MPE-FEC module, provided byDVB-H, offers over the physical layer transmission, a complementary forward errorcorrection allowing the receiver to cope with particularly difficult receivingsituations.




Notes: Notes:


DVB-TDVB-T

• Digital Video Broadcasting over Terrestrial

• ETSI EN 300 744

— Framing structure, channel coding and modulation for digital terrestrialtelevision• ETSI EN 301 958

— Interaction channel for Digital Terrestrial Television (RCT) incorporatingMultiple Access OFDM

• ETSI TR 101 190 — Implementation guidelines for DVB terrestrial services;Transmission aspects

VB-T stands for Digital Video Broadcasting - Terrestrial and it is the DVBEuropean consortium standard for the broadcast transmission of digital terrestrialtelevision. This system transmits a compressed digital audio/video stream, usingOFDM modulation with concatenated channel coding (i.e. COFDM). The adoptedsource coding methods are MPEG-2 and, more recently, H.264.

In January 2006, a study group named TM-T2_SM (Study Mission on SecondGeneration DVB-T) has been established by DVB Group for investigation ofadvanced modulation schemes that could be adopted by a second generation digitalterrestrial television standard .




Notes: Notes:


DVB-IPIDVB-IPI

• Digital Video Broadcasting over IP Interfaces

• ETSI TR 102 033

— Architectural Framework for the Delivery of DVB Services over IP-basedNetworks• ETSI TR 102 034 (Formerly DVB.org Blue Book A086)

— Transport of MPEG-2 Based DVB Services over IP Based Networks• ETSI TS 102 813

— Transport of DVB Services over IP-based Networks: IEEE 1394 HomeNetwork Segment

• ETSI TS 102 814 — Transport of DVB Services over IP-based Networks: Ethernet Home

Network Segment

DVB-IPI is the set of standards for delivery of DVB over IP interfaces. It evolvedfrom the DVB.org Blue Book A086 standard and now forms the basis for mostIPTV distribution systems.

TR 102 034 is the original document which concentrates on the Transport of DVBover IP.




Notes: Notes:


DVB-IPI Architecture TR 102 033DVB-IPI Architecture TR 102 033

• The Architecture defines 4 domains

Content Provider: the entity who owns or is licensed to sell content or contentassets. Although the Service Provider is the primary source for the client at Home, adirect logical information flow may be set up between Content Provider and Homeclient e.g. for rights management and protection.Service Provider: the entity providing a service to the client. Different types ofservice provider may be relevant for DVB services on IP, e.g. simple InternetService Providers (ISPs) and Content Service Providers (CSPs). In the context of

DVB services on IP, the CSP acquires/licenses content from Content Providers andpackages this into a service. In this sense the service provider is not necessarilytransparent to the application and content information flow.Delivery Network: the entity connecting clients and service providers. The deliverysystem usually is composed of access networks and core or backbone networks,using a variety of network technologies. The delivery network is transparent to theIP traffic, although there may be timing and packet loss issues relevant for A/Vcontent streamed on IP.Home: the domain where the A/V services are consumed. In the home a singleterminal may be used for service consumption, but also a network of terminals andrelated devices may be present for this purpose




Notes: Notes:


DVB-IPI Home Reference ModelDVB-IPI Home Reference Model

1) A home network can be simultaneously connected to multiple and heterogeneousdelivery networks. As an example, in a typical scenario ADSL and DVB-S are bothavailable at the home. Load balancing may be possible between the differentdelivery networks in order to optimize the utilization and throughput of thenetworks and to minimize the delay.2) End users can choose the service provider. As an example, the ISPs and the CSPsmay be independent from each other.

3) Different end users in the same home network can select different serviceproviders.4) Access to the content is independent from the underlying hardware.As anexample, terminals with different capabilities (e.g. CPU power, display size,storage capacity) may be allowed to access to the same content through the use oftranscoding resources, or through the use of device specific resources.5) Roaming of end users between delivery networks should be possible. As anexample, the personal environment of a (SOHO) user stored on a home servershould be accessible from different external locations. Adequate security aspectsneed to be taken into account.




Notes: Notes:


DVB-IPIDVB-IPI

• TS 102 034 species a specific subset of Internet Protocols to be used — Real-Time Streaming Protocol (RTSP) and Real Time Protocol are used — These transport delivery of broadcast TV and audio

This is a logical diagram of the high-level protocols on the IPI-1 interface, specifiedin the present document for enabling DVB services over IP-based networks. Theorganization of this protocol stack is according to the ISO/OSI layering convention.The top layer of this stack signifies the service offering intended by the ServiceProvider. This consists of programs, information about programs, multicast- and/orunicast IP addresses; in short, the essential items needed to enable a DVB serviceover an IP network. The present TS 102 034 document specifies the protocolsrequired for transport of elements of the service offering via IP networking, inprinciple independent of the physical layers below the IP networking layer.However, for use in future DVB Home Networking, the present document alsospecifies the Ethernet and IEEE 1394 Home Network Segments as physical layers.They are shown in their correct place, at the bottom of the diagram.The HNED is an IP compliant device; on its IPI-1 interface it supports therequirements laid down in RFC 1122. HTTP, TCP, UDP and IP are available to theHNED as networking and transport protocols.S




Notes: Notes:




IP Delivery

Encoding Methods

Chapter Summary





Notes: Notes:



Now you have completed this chapter you can

• Examine component parts of a TV distribution networks

• Explore how the various systems options

• Identify the key interfaces

• Predict how the technology will evolve in the near future

• Examine the encoding and compression standards




Notes: Notes:

Engineering Reliable IP ServicesEngineering Reliable IP ServicesEngineering Reliable IP Services






Notes: Notes:


Course MaterialsCourse Materials

• Course Notes — Copies of all slides and supplemental presentation material




Notes: Notes:


Course ContentsCourse Contents

Introduction and Overview

Chapter 1 Internet Protocol Suite Delivery of Multimedia Service

Chapter 2 AddressingChapter 3 Routing

Chapter 4 Managing Devices With SNMP

Chapter 5 Course Summary




Notes: Notes:


Course Contents (Continued)Course Contents (Continued)

Appendix A Networks Glossary




Notes: Notes:


Course ScheduleCourse Schedule

Each day, the course will follow this schedule:

Start class 9:30 a.m.

Morning break 10:30 a.m. (approximately)

Lunch

Resume class

Afternoon break(s) As needed

Adjourn 5:00 p.m.




Notes: Notes:


LogisticsLogistics

• Restrooms/toilets

• Drinking fountains, coffee and soft drink machines, snacks

• Restaurants

• Messages/phones

• Security

• Emergency measures

• Use of equipment after class hours (if applicable)

• Other important items




Notes: Notes:


Course InstructorCourse Instructor

• Background and education


• Experience




Notes: Notes:


Attendee IntroductionsAttendee Introductions

• Your name

• Organization name


• Experience in:- — Telecommunications — TCP/IP networking

• Expectations




Notes: Notes:

Internet Protocol Suite Delivery of

Multimedia Services

Internet Protocol Suite Delivery ofInternet Protocol Suite Delivery of

Multimedia ServicesMultimedia Services

Chapter 3Chapter 3

In this chapter we will examine all the protocols that were used in order to connectand operate the VoIP calls we used in demonstration 1 and captured indemonstration2. You can work from the LANwatch capture that you made of a realcall or from the handout of a previous capture.




Notes: Notes:



When you have completed this chapter you will be able to

• Describe the key protocols used for voice over IP

• Discuss addressing and routing in IP networks

• Explore the operation of applications within IP networks

• Characterize the behavior of TCP/IP networks

• Compare some alternative WAN technologies




Notes: Notes:


Internet Protocol Suite Delivery of Multimedia ServicesInternet Protocol Suite Delivery of Multimedia Services

TCP/IP Architecture

Network Links

IP

TCP and UDP

Applications

Chapter Summary




Notes: Notes:


Can We Put Voice on Our Data Networks?Can We Put Voice on Our Data Networks?

• IP networks are packet networks, which provoke several questions: — Can we transmit voice over packets? — What protocols would we use? — What is the impact on our data network?

• Voice requires reliable, timely delivery; i.e., it is a “real-time” application — Can this be done on “best-effort” data networks? — What protocols can deal with this requirement?

• Voice networks need high reliability and availability — Can data network routing protocols cope with outages quickly enough?

• To connect a VoIP device, we need — IP addresses — Physical connectivity

– Where and how should we connect devices?

Telephone companies (Telcos) are optimized for voice. (Shannon’s Law, ErlangTables) Lans and Wans are optimized for throughput (RSVP, Tag Switching).Voice to Data ratios on telcos are tipping in favor of Data. Doesn’t it make sense toput voice on the data network rather than the other way around? Isn’t this justanother way of saying that all networks are migrating (or will migrate) to data as

the voice/data ratio becomes distorted in favor of data?




Notes: Notes:


Sources of Network StandardsSources of Network Standards

• There are two major standards groups of importance

• International Telecommunications Union (ITU)

• Internet Engineering Task Force (IETF)

There are two important sources of standards that are vendor independent.Standards are said to be open if they are:-

1. Built to standards

2. Vendor Independent

3. Widely available




Notes: Notes:


International Telecommunications Union (ITU)International Telecommunications Union (ITU)

• The only telecommunications body with United Nations Charter

• Interested in International Interconnection

— National standards bodies adapt and extend standards• European Telecommunications Standards Institute (ETSI) evolve from

these

• Recognize standards from numbers — A for administrative — E for numbering plans — G for trunk and CODEC standards — H for multimedia standards (VoIP) — Q for Signaling — V for data over analog telephony — X for digital data standards — and others

• See http://www.itu.org

ITU standards are internationally recognized. The ITU is the ONLY internationalbody with a charter from the UN to deliver standards.

Historically they were produced every four years, being voted on at a largeconference held every four years in a warm location. From 1992 onwards thereviews of standards have been as and when required allowing for much fasterevolution of standards.

A critical difficulty is the need under the charter for there to be agreement frommajor nations. This has caused some standards to be held up because of nationalinterests.

Standards must be purchased or subscribed to. Downloading a single standard canbe expensive for an individual.




Notes: Notes:


Internet Engineering Task Force (IETF)Internet Engineering Task Force (IETF)

• Produces standards upon which Internet is based

• Called Requests For Comment (RFC)

— Most RFCs are proposals for standards — Only small proportion finally accepted• Available free via the Internet

— Information available from Internet Assigned Numbers Authority – http://www.iana.org/

— Downloadable from http://www.ietf.org/rfc.html — List of current standards in STD 1 — http://rfc.net/std1.html — Currently RFC3600

The IETF is an unpaid voluntary body that is heavily dominated in practice byAmerican companies and Universities. However it tends to develop standardsmuch faster than the ITU.

Ideas for new standards are produced as RFCs and these may or may not eventuallybecome standards.

All standards and drafts are freely available over the Internet.




Notes: Notes:


Internet Protocol Suite StructureInternet Protocol Suite Structure

• IP does not care what the lower layers are: LAN or WAN — WAN can be frame relay, ATM, Point-to-Point Protocol (PPP)

TCP/IP

Application

Transport

Internet

Network interface

Hardware

Application

Transport

Internet

Network interface

Hardware

7. Application

6. Presentation5. Session

4. Transport

3. Network

2. Data Link

1. Physical

OSI Model

Internet Model

OSI = Open Systems Interconnection

We need to point out here that the top three layers of the OSI model are theapplication running in the PCs. A similar application is running less visibly in therouters to achieve the connection and operation.




Notes: Notes:


OSI ModelOSI Model

Provides applicationservices

Provides datarepresentation

Provides checkpointing,activity manageme nt

Provides end-to-enddata integrity

Rou te s and re lays

Manages communicationbe tween ad jacent nodes

Transmits bit streamover physical medium

•

•

•

•

•

•

•

• ISO 7498 and ITU-T X.200

Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

The OSI model has 7 layers as this is the way the human brain works.

The average person can hold in their mind only about 7 (plus or minus 2) ideas atany one time. Therefore we divide the functions of communications into 7 groups.

The ISO standard for the model is very abstract and complex. We shall thereforetranslate this into simple, easy to grasp rules of thumb ; roughly what each layerdoes in simple terms.




Notes: Notes:


OSI ModelOSI Model








•

•

•

•

•

•

•Moves Bits


Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Layer 1 moves bits. Anything to do with moving bits is layer 1. Electrical levelsthat represent bits, the cables through which the bits move, the plugs and socketsthat join the cables, the timing of the bits.- all these are layer 1.

Layer 1 is governed by the laws of physics and the laws of nature — one law ofnature above all others — Murphy’s First Law. If things can go wrong they will;

they will go wrong most in layer 1.

So now we need a layer of firmware to detect errors. This is layer 2.




Notes: Notes:


OSI ModelOSI Model









•

•

•

•

•

•

•Moves Bits

Detects Errors

Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Layer 2 detects errors. Anything to do with error detection is layer 2. We maydivide the bit stream up into groups of bits called frames and calculate an errorcheck which is sent at the end of the frame. This is known as a Frame CheckSequence or Cyclic Redundancy Check.

While layer 2 is responsible for framing and error detection it is not required to

correct the errors it finds. Some layer 2 protocols do but most don’t.

I could construct a network of nodes interconnected by layer 1 cables, each linkrunning a layer 2 detecting errors. However to deliver the data to the correctdestination I need Routing which is the job of layer 3.




Notes: Notes:


OSI ModelOSI Model









•

•

•

•

•

•

•Moves Bits

Detects Errors

Routing

Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Layer 3 does routing. How did you get here today? I came by centralized staticrouter — I came by train. You may have come by distributed dynamic router —taxi, or even by distributed static router — bus.

There are many forms of routing, some have advantages over others. We will lookat some of these later but in our case notice they all worked. However have you

ever jumped on the wrong train, or missed your stop, or even found that your trainwas cancelled. Yes, the network layer can make mistakes. Not deliberate mistakes,but failures in the network can cause data to be lost.

To fix this we need a transport layer.




Notes: Notes:


OSI ModelOSI Model









•

•

•

•

•

•

•Moves Bits

Detects Errors

Routing

End to End ErrorRecovery

Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Layer 4, the Transport layer, runs from end-to-end and corrects errors the lowerlayers leave behind. It is this layer that guarantees the correct delivery of the data.Notice that it is in the ends not in the middle of the network.

On the Internet when you are surfing the Transport layer runs in your desktopcomputer and in the web server itself — it does not run in the router within the

Internet.




Notes: Notes:


OSI ModelOSI Model








•

•

•

•

•

•

•Moves Bits

Detects Errors

Routing


Checkpoints andActivities


Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Layers 5, 6 and 7 add services to the network. The first group of these is added bylayer 5 — the Session layer. This layer adds checkpoints and activities.

Checkpoints are points in a conversation that we can go back to. If you have eversurfed the Internet you have used the session layer without realizing it. When youclick on a web link the system takes you to a new web page but remembers whereyou came from within the session layer. At any time you can click the back button

on the web browser and return to exactly the point where you clicked the link. Thisis the purpose of checkpoints.

However some systems are so simple that they have only a single transaction -perhaps taking money from an automatic teller machine. This will start an activityat the start of the transaction (when you insert your plastic card) and then requestinput from you. If you press the return-card button, the system will abort theactivity. If on the other hand you complete the transaction it will end-activity andupdate all the databases. Yes, databases depend upon the session layer.




Notes: Notes:


OSI ModelOSI Model








•

•

•

•

•

•

•Moves Bits

Detects Errors

Routing



Converts bits to Objects


Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

The Presentation Layer, layer 6, converts bits to objects. Characters are objects, soa code set is Presentation Layer. Voices are objects so bits to voices and voices tobits in a CDEC is Presentation Layer too. Pictures to bits in MPEG is alsoPresentation Layer. Have you ever seen Star-Trek where there is a very cleverPresentation Layer in Captain Kirk’s Transporter which converts bits to people andpeople back into bits.




Notes: Notes:


OSI ModelOSI Model








•

•

•

•

•

•

•Moves Bits

Detects Errors

Routing



Converts bits to Objects

Application and itsServices


Physical

Data Link

Network

Transport

Session

Presentation

Application

1

2

3

4

5

6

7

Finally layer 7 is where the application and all the other functions run. Indeed youthe user are in layer 7 too.




Notes: Notes:


Internet Protocol Suite StructureInternet Protocol Suite Structure

• IP does not care what the lower layers are: LAN or WAN — WAN can be frame relay, ATM, Point-to-Point Protocol (PPP)

TCP/IP

Application

Transport

Internet

Network interface

Hardware

Application

Transport

Internet

Network interface

Hardware

7. Application

6. Presentation5. Session

4. Transport

3. Network

2. Data Link

1. Physical

OSI Model

Internet Model

OSI = Open Systems Interconnection

We need to point out here that the top three layers of the OSI model are theapplication running in the PCs. A similar application is running less visibly in therouters to achieve the connection and operation.




Notes: Notes:



TCP/IP Architecture

Network Links

IP

TCP and UDP

Applications

Chapter Summary




Notes: Notes:

In reality Internetworking involves the interconnection of LANs using WANsformed from routers.


Network InterconnectionNetwork Interconnection

• Internetworking issues: — Addressing — Routing, path through the network — Packet size and format — Access technology — Shared protocols

– Errors, flow, timing

• These issues are resolved by an internetworking protocol (Layer 3)

LAN A

LAN B

Host

Host




Notes: Notes:


Data Link and Physical Layer in the LANData Link and Physical Layer in the LAN

• IP can run over any kind of LAN

Application

Transport

Internet

Network interface

Hardware

Application

Transport

Internet

Network interface

Hardware

LAN or WAN link

TCP/IP

The LANs and the physical links that join routers together form the lowest twolayers of the OSI model and of the Internet model.




Notes: Notes:


Data and Computer NetworksData and Computer Networks

• LANs permit computers to interconnect locally — Standardized by IEEE 802 committee

– Includes wired and wireless versions — Use 48-bit addresses that are not routable — Address Resolution Protocol maps LAN addresses to IP addresses

• Limited in physical dimensions

• LANs limited to one site can be interconnected — Effectiveness of interconnection depends upon data rate of access

We need to explain why LAN addresses are not easy to use for voice addressing asthey are not routable. The first 3 bytes in the addresses are a code that identifies themanufacturer of the interface, but although we know who made the device we haveno idea where it is. This makes 802 addresses fine for devices phyically on thename LAN, that is on the same site, but not much use on their own when we have

many potential sites.IP addresses start with a Network address which is location fixed and so we can usethis to get to the right network and finally the device host address to reach the finalend point.






Notes: Notes:


Public network

Type Bit rates DistanceHDSL 1.5–2.0 Mbit/s,

symmetric4.5 kmof 2- or 3-pair UTP

SDSL 1.5–2.0 Mbit/s,symmetric

3 km of1-pair UTP

ADSL 1.5–9 Mbit/s down,16–640 kbit/s up

2.5–5 km of 1-pairUTP

VDSL 13–52 Mbit/sdown,1.5–2.3 Mbit/s up

0.3–1.5 km of1-pair UTP

xDSLxDSL

• Digital Subscriber Loop technologies provide access over telephone loops — Often known as the last mile — HDSL (high-data-rate DSL) — SDSL (single-line DSL) — ADSL (asymmetric DSL)

– High bit rate downstream – Low upstream

— VDSL (very high-data-rate DSL)

Digital subscriber loop technologies have evolved to allow high speed transmission over the coppersubscriber loops installed originally to carry analog phone conversations.

The loop generally pass from the subscriber premises to the local exchange. In more than 95% ofcases this distance is less than 5 km.

Generally speaking the higher the data rate used, the higher will be the bandwidth of frequenciesrequired to carry the signals, and thus the higher will be the highest frequency. Generally speakinghigh frequencies suffer most loss and are affected most by noise. The more complex and expensivethe electronics used, the higher will be the data rate achieved, but eventually the rate will be limitedby the physics of the problem.

Most loops are just a single pair of wires for sending data in both directions. It is not easy to use thesame frequencies in both directions at the same time and so part of the bandwidth is normallyallocated to each direction. Some xDSL technologies use different amounts of bandwidth in eachdirection and so are Asymmetric. Others may use two pairs of wires or use equal bandwidths and besymmetric.




Notes: Notes:



TCP/IP Architecture

Network Links

IP

TCP and UDP

Applications

Chapter Summary




Notes: Notes:


Internet Protocol (IP)Internet Protocol (IP)

• Basic concepts: — IP used by hosts, routers, gateways — Does not need to know how to get to the final destination (endpoint)

– Just to the next network (host, next hop) — Does not need to know about all of the networks on the route

– Just the locally connected networks• IP packet contains destination address and data

— That’s all that’s needed to reach destination• Best-effort delivery

— No guarantees

The job of a router is to forward a packet to the next “best hop” en route to itsdestination.




Notes: Notes:


Internet Protocol SuiteInternet Protocol Suite

• Internet protocol suite — Includes applications

– More than 100 now defined — Operates at Layer 3 and higher

ICMPARP

UDP

Etc.SNMPFTPSMTP RIPPOP Etc.

Physical

Data link

IP

TCP

HTTP Application

Transport

Internet

Network interface

Hardware

RIP = Routing Information ProtocolSNMP = Simple Network Management Protocol

This diagram shows some protocols run over TCP and some over UDP. Oursignaling is going to run over TCP and our voices over UDP.




Notes: Notes:


EncapsulationEncapsulation

• Layer 4 interface is with processes on the host — For example, SMTP or POP

• Layer 3 interface is with other hosts via address — For example, IP address

• Layer 2 depends on the type of physical network used — For example, frame relay, ATM, ISDN, Ethernet

— May include additional addressing (SVC, PVC, etc.)

Ethernet IP TCP E-mail

E-mail

TCP E-mail

IP TCP E-mail

Me YouE-mail

Ethernet IP TCP E-mail

IP TCP E-mail

TCP E-mail

PVC = permanent virtual circuitSVC = switched virtual circuit

Notice that IP is not used alone. Each layer of the protocol stack passes its datawithin the protocol data units of the layer below. This is known as encapsulation.

As data passes down the stack, the units of transfer grow. At the receiving end thetransfers are unwrapped.




Notes: Notes:


IPv4 DatagramIPv4 Datagram

TOS = type of serviceTTL = time to live

0 314 bytes

20 bytesProt Checksum

Fragoff

IHL TOS

Data

Options

Destination address

Source address

TTL

Datagram ID

Ver Length

MF

DF0

To see exactly how IP functions it is possible to capture traffic that is running overa LAN or serial PC interface. Using Ethereal which can be downloaded fromwww.ethereal.com and installed on almost any PC it is possible to view the fieldswithin the IP header.




Notes: Notes:


IPv4 Datagram (continued)IPv4 Datagram (continued)

• Datagram fields important to VoIP — Type of service (TOS)

– Used in VoIP QoS – First 3 bits give precedence

— Datagram ID provides a unique number for the packet with the TTL — TTL: time to live in seconds (hops)

– Routers reduce by 1 or number of seconds held — Flag field

– MF: Indicates last fragment (MF=0 is last fragment ) – DF: Permission to fragment—Don’t Fragment ( DF bit)

— Fragoff: Fragment offset – Measured in units of 8 octets

— Prot: Next protocol – 6=TCP, 17=UDP

•

Total overhead is at least 20 bytes for the IP packet header

The header of IP version 4 packets is generally 20 bytes long. The version numberand the Internet Header Length (IHL) confirm the version of IP and the header sizemeasured in 32 bit words. The Type of Service (TOS) field is not often used tocarry data but can be used to identify the kind of data being carried. Recently thisfield has been renamed the differentiated services code point and can be used in

VoIP systems to identify voice packets. Routers can use this field to give priorityto voice packets over data.

The length field identifies the total size of the datagram including the header. As itis 16 bits long packets are limited to 64kbytes in length.

Th fields in the next 32 bit word are used for fragmentation indicated the datagramnumber, a unique field value that is present in every fragment of a datagram.

The Time To Live (TTL) is used to prevent packets going into loops within theInternet and is reduced by 1 in each router.




Notes: Notes:


Where The Internet Protocol RunsWhere The Internet Protocol Runs

Router

Router

Router

Router

Router

Wireless Network

Switched Network

LAN

Point-To-PointNetwork

IP is the most widely used protocol in the world and represents in excess of 85% ofthe market. Indeed it is increasing in use with both voice and video distributionnow available over IP. IP was developed originally by the US DOD from theARPAnet project. ARPAnet was an experimental packet network developedbetween 1963 and 1969 to be used to launch nuclear missiles. It was deployedbetween 1969 and 1972 when the defense networks were split into the specialnetworks for secret military use and the non-classified parts which were titled theInternet. Between 1972 and 1976 the Internet Protocol Suite was extended and IPimproved over several versions until version 4 (known as IPv4) was reached whichwe now use. Development has continued to improve capabilities and address spacewith Version 6 being completed in 1996. However the migration from version 4 toversion 6 will be very long and painful — so painful that some have doubted it willever happen.




Notes: Notes:


Where The Internet Protocol RunsWhere The Internet Protocol Runs

• The internet protocol runs in every host and every router — Host: a device that communicate over the internetwork — Router: a device that joins one or more networks onto the internetwork

• Does not run in devices that form the networks themselves — Not in devices below OSI layer 3

– e. g. – Not in Ethernet switches – Not in Bridges – Not in Modems – Not in Network Termination Units – Not in any device that is Layer 2 or layer 1

IPv4 runs in every router and every computer that needs to connect over theInternet within the layer 3 of its protocol stack. Hubs, bridges, switches, modemsand other layer 1 or layer 2 devices are invisible to IP.

It is an Internetworking protocol which means that it can be used not just within anetwork but between networks. Indeed it was intended that it should be possible to

interconnect machines attached to any kind of network with IP. This has proved tobe the case.






Notes: Notes:


Physical Medium IndependencePhysical Medium Independence

• Each network technology may have its own characteristics — Different hardware — Different API for access — Different timing dependency

• IP provides a standard logical interface for exchange of data packets

IP Different NetworkInterfaces

Single CommonNetwork Interface

NetworkType 1 NetworkType 2 NetworkType 3

All internetworking protocols that join network technologies together must provideindependence from the layer 1 and 2 used. The physical cables and channels usedon different physical technologies bring with them limitations. IP removes theselimitations.

In practice this is achieved by configuring low level driver software that supportsthe physical technology below and interfaces to one of a range of driver interfaces

to IP. NDIS (Network Driver Interface Specification) and ODI (Open DriverInterface) are two popular ones for PCs.




Notes: Notes:


Logical AddressingLogical Addressing

• Each network technology has its own addressing system

• We require interoperation between any group of devices

• IP introduces a single unique logical addressing scheme — Each device is given a logical address in addition to its physical network

address — All IP addresses form part of the same single address space

48-bit Ethernet

14-decimal-digit X.25

IP addressmapping 32-bit IP address Internetwork

address

E.164 15 digit

ATM 20 byte NSAP

Different network technologies use different address lengths. X.25 a 14 digitaddress, the phone service E.164 a 15 digit address, Ethernet a 48 bit address and soon. All are different addresses and even different sizes. Devices on these networkswill retain their original addresses but will be allocated another logical address byIP.




Notes: Notes:


Independence of MTU sizeIndependence of MTU size

• Each technology has a different maximum transmission unit size — The optimum MTU is determined by the error performance — The more reliable the physical transmission the optimum MTU — High error rates require small packets

– More chance of error so less data can be sent before error occurs

• Some actual MTUs for technologies widely used

Network MTU (bytes) Max frame sizeEthernet 1,500 1,518IEEE 802.5 (4 Mbit/s) 4,440 4,500IEEE 802.5 (16 Mbit/s) 17,940 18,000FDDI 4,352 4,500X.25 4096 4102

Different networking technologies have different maximum frame and packet sizes.IP can support packets up to 64 kbytes in length and will fragment packets asnecessary to fit within the network maximum sizes. There are limits however. Thesmallest IP header is 20 bytes long and each segment other than the last mustcontain a multiple of 8 bytes.

There is a recommendation that networks should support at least 576 byte packetsas a minimum. In practice most do, although radio networks may need limitationssmaller than this.




Notes: Notes:


Fragmentation and ReassemblyFragmentation and Reassembly

• It is the task of the router to match the MTU sizes between networks

• Packets too large for delivery over the output network are fragmented

• Destination host reassembles — Packets may be fragmented several times between source and destination

Source Destination

Routers are responsible for fragmentation and will fragment datagrams down tosizes small enough to pass through output networks. The destination of the datamust then reassemble the original datagram from the fragments. Reassembly canbe a difficult and time consuming process so is best avoided if possible.




Notes: Notes:


Internetwork Datagram ServiceInternetwork Datagram Service

• IP always offers a datagram service — Best Efforts but no guarantee

Source Destination

Virtual Circuit Virtual CircuitDatagram

IP is a datagram service. This means that while it will try to deliver data as quicklyas possible there is no guarantee. Indeed in the event of congestion or othernetwork limitations, routers will discard datagrams without warning. End systemsmust therefore use retransmission timers in the Transport layer (layer 4) toovercome datagrams lost.




Notes: Notes:



TCP/IP Architecture

Network Links

IP

TCP and UDP

Applications

Chapter Summary




Notes: Notes:


Transmission Control Protocol (TCP)Transmission Control Protocol (TCP)

• Provides “reliable,” in-sequence stream transport — Useful for transfer of large volumes of data such as file transfer — Connection oriented—virtual circuit — When connection is established, data transfer starts

– Protocols verify correct reception – Buffered traffic flow – Full-duplex connection

• Accounts for more than 90 percent of Internet traffic

• “Reliability” is achieved through retransmission — When packets are lost or errors occur, retransmission provides “reliability”

– Suitable for data traffic and signaling for voice: Ensures correctinformation

– Not suitable for voice traffic: Voice information must be continuous – By the time data is retransmitted, it is too late!

TCP is used to overcome the short coming of IP. IP provides delivery of datawithout any guarantees. Data may be corrupted, lost, duplicated or even (intheory) delivered to the wrong address.

TCP provides sequence numbers and timers that allow data to be delivered onceand only once, in order and with a high level of guarantee. In order to achieve this

data is held in buffers by the sender until an acknowledgement is received and maybe resent repeatedly after a time-out if no acknowledgment is received. Thisrequires significant amounts of processing capacity, memory and takes time.

TCP may be used in VoIP to carry the signaling used to dial and connect calls, butadds too much delay to carry the voice itself.




Notes: Notes:


TCP Segment HeaderTCP Segment Header

• Port is a destination for the message• Local host process can communicate with the port• A pair of IP addresses and port numbers for a connection forms a socket

— Complete specification for anassociation —a conversation•

Adds at least 20 bytes overhead per packet

PaddingUrgent pointer

Reserved Code bits Window

Destination port

Options (normally absent)Data

Options (if any)Checksum

HLENAcknowledgment number

Sequence numberSource port

0 4 10 16 24 31

Units of transfer for TCP are called Segments. Conversations are identifiedbetween source and destination using both addresses in the IP header together withboth port numbers in the TCP header. This is known as a Full Association.

Sequence numbers and acknowledgement numbers are used to ensure that alltransfers are received in order and acknowledged. If a segment is lost then a timer

in the sender will expire and cause the segment to be resent. This takes additionalprocessing and delay but ensures all data is received.

Because the additional retransmission and processing takes time it causes delay. Itis therefore not practical to use TCP to carry voice in most cases.




Notes: Notes:


User Datagram Protocol (UDP)User Datagram Protocol (UDP)

• Used for unreliable delivery— i.e., not acknowledged — Application must handle errors

– Loss of packet, duplication, delay – Out-of-sequence, loss of physical connectivity

— These functions add processing overhead in applications but… — Reduce processing in the transport layer

– Much less than TCP• Accounts for less than 5 percent of Internet traffic currently

• Used for transaction processing; selected for voice transport

When the overhead and delay of TCP is not required or cannot be tolerated UDP isused. This identifies the full association but does not guarantee delivery. Wegenerally prefer to have lower delay rather than retransmission of lost data. Data isgenerally lost when the network is overloaded and so by careful sizing we canavoid overload.




Notes: Notes:


UDP HeaderUDP Header

• Source and destination are ports

• Length is total length of packet

• Checksum is for the header and data — Checksum is optional — All zeros if not used

Checksum

Destination port

User data …

Length

Source port

0 15 31

The port numbers and IP addresses together identify the conversation. Thechecksum can be used to verify that no data has been lost or corrupted in thesegment but in practice the layer 2 protocol will have already confirmed accuratetransfer.




Notes: Notes:



TCP/IP Architecture

Network Links

IP

TCP and UDP

Applications

Chapter Summary




Notes: Notes:


Applications Use PortsApplications Use Ports

• Hosts are multitasking computers running multiple applications — Communication takes place between the applications running on the hosts — Linkage is not direct; use software code called a port

– Allows operating system to direct packets to correct application• Server ports are normally well-known ports—less than 1024

— HTTP 80 SMTP 25 — FTP 21, 20 DNS 53 — Telnet 23 NNTP 119

• Ports used by VoIP are generally client ports—greater than 1024 — End-to-end VoIP call setup uses destination port 1720

– UDP ports dynamically assigned and both ends > 1024

Different conversations are identified by using the IP addresses and port numbers ofboth ends of a conversation. The change in any one of these four values, or in thelayer 4 protocol (TCP or UDP) represents a different conversation.

Client server operations are carried using low numbers less than 1024 in the portnumber of the server.

As VoIP does not normally run between client and server, but runs client to client,low port numbers are not used. However well known values are used to carrysignaling for H.323 (1720 to connect calls) and for SIP (5060).




Notes: Notes:


Datagram DeliveryDatagram Delivery

• Ports assigned — Fixed

– “Well-known ports” assigned at and below 1024 — Dynamic

– Assigned by applications: above 1024 up to 65536 (216)• RFC 1700 (assigned numbers)—defined “well-known ports” in 1992

— Updated list available fromhttp://www.iana.net

TCP UDP

IP

UDP applicationTCP application

Defined by “Protocol / Next” field

Defined byport number

I was going to put a copy of all the registered port numbers in an appendix butwhen I looked the list was enormously long now. You can find this on the Internetat

http://www.iana.org/assignments/port-numbers.

Signaling ports are registered numbers but not less than 1024 so not well known inthe correct sense. VoIP conversations are normally NOT client server in thetraditional sense but peer to peer so both ends are "client" ports greater than 1024.




Notes: Notes:


Real-Time Transport Protocol (RTP)Real-Time Transport Protocol (RTP)

• TCP/IP protocol suite includes protocols for real-time applications — Real-Time Transport Protocol (RTP) — Real-Time Control Protocol (RTCP)

• RTP provides — Timestamping, sequence number

– For playback timing and synchronization — Setting up real-time applications

– Audio and video• RTCP provides

— Reporting on achieved results — Delay, packet loss statistics

• Defined in RFC 1889 originally — Copied by H.323 systems

TCP delivers too much delay to carry voice but we do need to ensure thatdatagrams of voice are played by the receiver in the order that the sender recordedthem and in the correct timing. RTP is used to achieve this.




Notes: Notes:


Real-Time Applications on Packet NetworksReal-Time Applications on Packet Networks

• To be intelligible, our speech must be played out with the same timingrelationship between words as the original — Received packets may not all arrive with exactly the same delay

– This is called jitter• Real-time Transport Protocol marks the voice samples with a timestamp

— That timestamp is used to play out the packet in sequence – With the correct relative time relationship

You’re right This is an IP telephony course

SentReceived

Individual packets are produced by the sender when there is sound to send. Whenthere is silence no packets are sent. In each packet a sequence number and timerindicate to the receiver the order in which packets must be decoded and played aswell at the time at which playing should start. When the receiver has no packets toplay at a particular time the receiver plays “silence”. In reality there is never any

real silence on a voice call so the sender must define the sound level below whichno data will be sent. To account for this some CODEC definitions include“comfort noise”, a low level sound which is used instead of true silence when thereare no packets to play. This proves to be less disturbing to listeners in real life thanpure silence.




Notes: Notes:


RTPRTP

• Real-time Transport Protocol—Adds minimum of 12 bytes — V: Version—RFC 1889 currently 2 — P: Padding— =1 if packet contains padding — X: Extension bit—if 1, then there is an extension header — CC: CSRC count—the number of CSRC identifiers following — M: Marker—profile may use this bit to define frame boundaries — PT: Payload type—defines the type of encoding

• Sequence number increments by 1 for each RTP data packet

V=2 CCXP M PT Sequence number

Timestamp

Synchronization source (SSRC) identifier

Contributing source (CSRC) identifiersCSRC = contributing source reference code




Notes: Notes:


Payload TypesPayload Types

• RTP does not define payload types — Defined by the application or an RTP profile

• For VoIP, the payload types are defined by the multimedia conferencingstandards (H.323 and H.225) — Most widely used types:

– Payload type Codec – 0 PCM µ-Law – 8 PCM A-Law – 9 G.722 audio codec – 4 G.723 audio codec – 15 G.728 audio codec – 18 G.729 audio codec – 34 H.263 video codec – 31 H.261 video codec

— These codecs are examined in more detail later

H.323 defines that voice should be carried according to H.225 standards which areidentical in format to RTP. The document does not define all the codecs given inthe RTP RFC but just a subset.




Notes: Notes:


Payload Types Defined in RFC 1890Payload Types Defined in RFC 1890

encoding audio/video clock rate channelsPT name (A/V) (Hz) (audio)

0 PCMU A 8000 11 1016 A 8000 12 G721 A 8000 1

13 GSM A 800015 DVI4 A 800016 DVI4 A 1600017 LPC A 800018 PCMA A 800019 G722 A 8000210 L16 A 44100111 L16 A 44100

14 MPA A 90000115 G728 A 8000

25 CelB V 9000026 JPEG V 9000028 nv V 9000031 H261 V 9000032 MPV V 9000033 MP2T AV 9000034--71 unassigned ?72--76 reserved N/A N/A 77--95 unassigned ?96--127 dynamic ?

This is the list of codecs from the RFC rather than H.323, and shows that qualitymuch higher than that over normal phone systems is possible.

The list has grown since the RFC and the latest list can be found at:-

http://www.iana.org/assignments/rtp-parameters

For example types 10 and 11 indicate CD quality sound (music) in either mono orstereo. Also video codecs are defined too.MP4 is in a dynamic range, which meansthat there is not a defined number for mpeg4. See RFC 3016. ISMA uses a differentRFC for audio transmission (or a RFC draft, actually).




Notes: Notes:


Real-Time Transport Control Protocol (RTCP)Real-Time Transport Control Protocol (RTCP)

• Real-Time Transport Control Protocol is used with RTP — Provides

– Monitoring and feedback of real-time parameters related to quality – Packets lost, cumulative packets lost – Interarrival jitter calculated as new = old + (delta-old)/16

– Calculated as each packet arrives (regardless of sequence) – Transport Level ID for the source of the RTP packets – Optional session control

– Minimal capabilities – – Start session, bye

• RTCP provides an option for encryption to ensure privacy

• There is no provision in the standard for any action that may be taken if theresults are unacceptable — RTCP is only a reporting mechanism

RTCP packets are so rare in our network that they are hard to find. You may onlyget one in a minute of what has been captured in a voice call. The highest rate isone every 5 seconds but without RTCP packets it is not possible to report to sourceupon loss and jitter changes. Probably it would not be possible to adjust the jitterbuffering in a receiver more often than RTCP reports. Changes within the network

loading that cause different end to end delays to be observed can cause packet lossuntil the next RTCP exchange. The infrequent exchanges in the classroom meanthat configuration changes that cause big differences in delay will result in losses inthe speech.




Notes: Notes:


RTCPRTCP

• RTCP has four functions — Primary function of RTCP is to provide feedback on quality — Carries the canonical name (CNAME) of the source

– This may or may not be displayed to the participants — Controls the rate at which the first two types are sent — Carries session control information

Most VoIP devices do not carry the identity in the CNAME but NetMeeting does.




Notes: Notes:


Example Multimedia Web ApplicationsExample Multimedia Web Applications

• Internet Radio

• Internet TV : see http://wwitv.com

• Streaming over the Internet






Notes: Notes:




• Describe the key protocols used for voice over IP

• Discuss addressing and routing in IP networks

• Explore the operation of applications within IP networks

• Characterize the behavior of TCP/IP networks

• Compare some alternative WAN technologies






Notes: Notes:




• Describe how addressing works at layer 2

• Examine IEEE 802 addressing

• Identify how LANs are interconnected at layer 2

• Examine Link level aggregation




Notes: Notes:


Layer 2 AddressingLayer 2 Addressing

IEEE 802 Addressing

Bridging and Switching

Spanning Tree

Aggregation

Chapter Summary




Notes: Notes:


IEEE 802 StandardsIEEE 802 Standards

• Institute of Electrical and Electronic Engineers produces LAN standards — Available fromhttp://standards.ieee.org/getieee802/

10BASE5

10BASE2

10BASE-T

100BASE-T

1 Gbps LANs

802.2 Logical Link Control

1 Mbps

5 Mbps

10 Mbps

802.4Token Bus

4 Mbps

16 Mbps

802.5TokenRing

DQDB155 Mbps

802.6MAN

802.3Ethernet • • • 8

0 2

. 1 o v e r a

l l

s t a n

d a r d s

Type 1Type 2

Others




Notes: Notes:


General Aspects of LANsGeneral Aspects of LANs

• Several common frame format aspects — Destination and source address fields

– 48-bit addresses — Variable-length “payload” data size

– Maximum size differs – 32-bit error-check fields

( 6 b yt es )Destination

address . . . Payload data C RC

0 = IEEE Admin1 = Local admin

0 = Individual1 = Group

Address is sent low order bits first in each byteE.g., Locally admin, group address 1100 0000 - - -

Ethernet Hex Address 0 3

Assignedto the vendor

(3 bytes)

Assignedby the vendor

(3 bytes)

Addressing on all 802 standard network technologies is similar. Each deploys a 48byte address comprising a 3 byte field identifying the Vendor and a 3 byte fieldmaking each address unique. The Vendor code was originally call a Manufacturercode and has recently changed its name again to an Organizational Unit Identifier.

Bytes are transmitted lowest bit in each byte first. The first two bits transmitted arereserved and used to identify the addressing scheme used and group addresses

deployed when using multicasting.




Notes: Notes:


Speed, Size and Distance LimitationsSpeed, Size and Distance Limitations

• If we remove the CSMA/CD then the speed and distance can increase

• Distance will be limited by the media and layer one characteristics

• By buffering in the hub collisions can be removed — The device changes from a hub or repeater to a switch or bridge

• However by interconnecting switches very large layer 2 networks result

• Also Ethernet is not routable — The address does not indicate where a station sits — Switches must flood broadcasts and unknown destination addresses — Switches must also build up tables of source addresses on each interface

FCSType orlengthSourceaddressDestinationaddress101010111010...1010 “Data” Pad

Preamble(7 bytes)

Startdelimiter (6 bytes) (6 bytes) (2 bytes)

Frame checksequence(4 bytes)

Min 64Max 1,518

bytes

If we buffer packets in hubs and repeaters then we can preserve the packets whileother send. By giving each device a separate interface that is buffered in bothdirections a switch is created and by filtering packets to remove those that a devicedoes not need to see the speed and distance limitations can be removed. Also muchfaster and more secure networks result.




Notes: Notes:


10BASE510 Mbps

Baseband500 M beforeneeding a repeater

– Or, the letter “T” after“BASE” standing for“twisted pair” theMedia Type

802.3 Variations802.3 Variations

• The IEEE 802.3 variations are often expressed in the form

Abbreviation Meaning

T Half duplex twisted pair, cat 3 or cat 5 (at 10 Mbit/s) 100m

TX Full Duplex Twisted Pair- two pairs of cat 5 100m

FX Fiber optic pair up to 400 m

LX Long wavelength fiber (1270-1355nm) typically SMF 10/125

SX Short wavelength fiber (770-860nm) typically MMF 50 or 62.5/125

LH Long Haul fiber (1310 or 1550nm) SMF 9/125

CX Shielded twisted pair 25 m or less

Notations now identify the speed, the kind of encoding (baseband or broadband)and the media used.




Notes: Notes:



IEEE 802 Addressing


Spanning Tree

Aggregation

Chapter Summary




Notes: Notes:


Dividing Heavily Loaded SegmentsDividing Heavily Loaded Segments

Heavy traffic!

Lighter traffic!

• Total LAN traffic on a segment may be high

• Bridges allow heavy traffic to be isolated

• Improves overall performance but adds delay between segments

However powerful a network becomes there will be limits on its capacity. Bydividing this into two parts where most of the traffic stays local to each half and oncommunication between devices on the two halves needs to pass between, greateroverall copacity is created.

The interconnecting device is called a bridge and a group of many bridges in asingle box is called a switch.






Notes: Notes:


Switch

Multiple Simultaneous PathsMultiple Simultaneous Paths

• Using multiple simultaneous paths means that backplane bandwidth is nolonger shared

• May be non-blocking every station can transmit at the same time — Can also be full-duplex• Each interface becomes a collision domain with full bandwidth

Each interface to a switch is normally full duplex and by ensuring that the backbonepath is greater than the sum of the interface speeds it is possible to build full duplexnon-blocking switches.




Notes: Notes:


802.1D Concepts802.1D Concepts

• Unique identification of a bridge — A unique 48-bit Universally Administered MAC Address, termed the

Bridge Address is assigned to each Bridge — The Bridge Address may be the individual MAC Address of a Bridge

Port typically the lowest numbered Bridge Port (Port 1)

To work each bridge bust be assigned a unique identification. This is formed eitherby configuration or using the address of one of it ports, typically the lowest.






Notes: Notes:



IEEE 802 Addressing


Spanning Tree

Aggregation

Chapter Summary




Notes: Notes:


The Rapid Spanning Tree ProtocolThe Rapid Spanning Tree Protocol

• Consider this example: First devices forward Bridge Protocol Data Units

In this example the devices have been assigned identifications based upon theirlowest port address. When a topology change is identified they output on every porta multicast packet giving details of their identity and listen for similar packet fromtheir neighbors. Identifications are compared and when a device receives a better(lower) identification it stops sending its own and forwards that received updatingthe Root Path Cost.

The Root Path Cost is the cost of the path from the root bridge, in reality if allinterfaces are the same speed it is a hop count.






Notes: Notes:


Bridge Protocol Data UnitsBridge Protocol Data Units

• Th BPDU Type can be either a topology change or a RTST

• The Root Identifier is encoded in Octets 6 through 13

— This is made up from the address of the root and its priority• The Root Path Cost is encoded in Octets 14 through 17

• The Bridge Identifier is encoded in Octets 18 through 25

• The Port Identifier is encoded in Octets 26 and 27




Notes: Notes:


RST Parameter ValuesRST Parameter Values

Parameter Recommended or Permitted CompatibilityDefault value Range Range

Migrate Time 3.0 — —Bridge Hello Time 2.0 — 1.0–2.0Bridge Max Age 20.0 6.0–40.0 6.0–40.0Bridge Forward Delay 15.0 4.0–30.0 4.0–30.0Transmit Hold Count 6 1–10 1–10

• RSTP Timer and Transmit Hold Count parameter values

• Bridge and Port Identifier Priority valuesParameter Recommended Range

or default valueBridge Priority 32 768 0–61 440 in steps of 4096Port Priority 128 0–240 in steps of 16




Notes: Notes:


Link Speed Recommended Recommended Rangevalue range

<=100 Kb/s 200 000 000 20 000 000–200 000 000 1–200 000 0001 Mb/s 20 000 000 2 000 000–200 000 000 1–200 000 00010 Mb/s 2 000 000 200 000–20 000 000 1–200 000 000100 Mb/s 200 000 20 000–2 000 000 1–200 000 0001 Gb/s 20 000 2 000–200 000 1–200 000 00010 Gb/s 2 000 200–20 000 1–200 000 000100 Gb/s 200 20–2 000 1–200 000 0001 Tb/s 20 2–200 1–200 000 00010 Tb/s 2 1–20 1–200 000 000

Port Path Cost valuesPort Path Cost values






Notes: Notes:


The Rapid Spanning Tree ProtocolThe Rapid Spanning Tree Protocol

Identity

Root Path Cost

Path to root Discarding Interface

Forwarding Interface

ForwardingEdge Interface

Here 111 has become the root.

Bridges that receive more than one copy of the information from the root comparethe root path cost values and select the one with the lowest value as the interface touse to reach the root. Other interfaces over which copies of the root identity werereceived with higher costs are “turned off” by discarding packets received. Packetsare then forwarded to/from the root port from/to all other ports.




Notes: Notes:


Effective TopologyEffective Topology

• The effective topology is thus reduced

In effect the network becomes a tree.




Notes: Notes:


Typical Ring Backbone TopologyTypical Ring Backbone Topology

• In a typical ring backbone one port becomes non-forwarding

A typical topology selected for reliable operation is a ring. This results in a treebeing built while all interfaces function. When one fails a topology change resultsand a new tree is built continuing service with the failed interface becoming theedge of the tree.




Notes: Notes:



IEEE 802 Addressing


Spanning Tree

Aggregation

Chapter Summary




Notes: Notes:


IEEE 802.3 Link AggregationIEEE 802.3 Link Aggregation

• When connecting 802.3 switches together 802.1d forms a spanning tree

• This means that there is only a single active link carrying traffic between

any two switches• As load increases this does not scale well as congestion can occur

• Adding additional links will not help if these are turned off by 802.1d!

• The solution to this is Link Aggregation

• Aggregation can be useful to provide: — Improved reliability without spanning tree reconfiguration — Increased throughput beyond the capacity of a single link

In this aggregated network we have 5 1 Gbit/s interconnections between twoswitches. Without aggregation 802.1d would turn off 4 of the 5 links deliveringonly 1 Gbiit/s capacity. If one link or indeed even if 4 links failed the service wouldcontinue but 802.1d might take a small number of seconds to switch cables.

With aggregation data is shared between the links and so we can achieve up to say5 times 1 Gbit/s as well as maitaining service without interruption over individual

link failures.






Notes: Notes:


0x01Type:88-09

01-80-C2-00-00-02

Link Aggregation Control PDU (LACPDU)Link Aggregation Control PDU (LACPDU)

LACPDUs are basic IEEE 802.3® frames; they arenot tagged. The LACPDU structure above has the following fielddefinitions:a) Destination Address (DA). The DA in LACPDUs is the Slow_Protocols_Multicast address. Its use and encoding arespecified in Annex 43B.b) Source Address (SA). The SA in LACPDUs carries the individual MAC address associated with the port through which theLACPDU is transmitted.c) Length/Type. LACPDUs are always Type encoded, and carry the Slow_Protocols_Type field value. The use and encodingof this type is specified in Annex 43B.d) Subtype. The Subtype .eld identi.es the specific Slow Protocol being encapsulated. LACPDUs carry the Subtype value0x01.

e) Version Number. This identities the LACP version; implementations conformant to this version of the standard carry thevalue 0x01.f) TLV_type = Actor Information. This field indicates the nature of the information carried in this TLVtuple. Actorinformation is identified by the value 0x01.g) Actor_Information_Length. This .eld indicates the length (in octets) of this TLV-tuple, Actor information uses a lengthvalue of 20 (0x14).h) Actor_System_Priority. The priority assigned to this System (by management or administration policy), encoded as anunsigned integer.i) Actor_System. The Actor’s System ID, encoded as a MAC address.

j) Actor_Key. The operational Key value assigned to the port by the Actor, encoded as an unsigned integer.k) Actor_Port_Priority. The priority assigned to this port by the Actor (the System sending the PDU;assigned by management or administration policy), encoded as an unsigned integer.l) Actor_Port. The port number assigned to the port by the Actor (the System sending the PDU), encoded as an unsignedinteger.m) Actor_State. The Actor’s state variables for the port, encoded as individual bits within a single octet.




Notes: Notes:



Priority and System identifiertogether identify the entity

Port Priority and Porttogether identify the portprefered in an aggregated group

All ports with same keyare in the same group

1) LACP_Activity is encoded in bit 0. This .ag indicates the Activity control value with regard to this link. Active LACP isencoded as a 1; Passive LACP is encoded as a 0.

2) LACP_Timeout is encoded in bit 1. This .ag indicates the Timeout control value with regard to this link. Short Timeout isencoded as a 1; Long Timeout i s encoded as a 0.

3) Aggregation is encoded in bit 2. If TRUE (encoded as a 1), this .ag indicates that the System considers this link to beAggregatable; i.e., a potential candidate for aggregation. If FALSE (encoded as a 0), the link is considered to be Individual;i.e., this link can be operated only as an individual link.

4) Synchronization is encoded in bit 3. If TRUE (encoded as a 1), the System considers this link to be IN_SYNC; i.e., it hasbeen allocated to the correct Link Aggregation Group, the group has been associated with a compatible Aggregator, and the

identity of the Link Aggregation Group is consistent with the System ID and operational Key information transmitted. IfFALSE (encoded as a 0), then this link is currently OUT_OF_SYNC; i.e., it is not in the right Aggregation.

5) Collecting is encoded in bit 4. TRUE (encoded as a 1) means collection of incoming frames on this link is de.nitely enabled;i.e., collection is currently enabled and is not expected to be disabled in the absence of administ rative changes or changes inreceived protocol information. Its value is otherwise FALSE (encoded as a 0);

6) Distributing is encoded in bit 5. FALSE (encoded as a 0) means distribution of outgoing frames on this link is de.nitelydisabled; i.e., istribution is currently disabled and is not expected to be

enabled in the absence of administrative changes or changes in received protocol information. Its value is otherwise TRUE(encoded as a 1);

7) Defaulted is encoded in bit 6. If TRUE (encoded as a 1), this .ag indicates that the Actor’s Receive machine is usingDefaulted operational Partner information, administratively con.gured for the Partner. If FALSE (encoded as a 0), theoperational Partner information in use has

been received in a LACPDU;8) Expired is encoded in bit 7. If TRUE (encoded as a 1), this .ag indicates that the Actor’s Receive machine is in theEXPIRED state; if ALSE (encoded as a 0), this .ag indicates that the Actor’s Receive machine is not in the EXPIRED state.




Notes: Notes:



Partner at the other end of the linkcan be identified if required

n) Reserved. These 3 octets are reserved for use in future extensions to the protocol. They shall be ignored on receipt and shallbe transmitted as zeroes to claim compliance with Version 1 of this protocol.o) TLV_type = Partner Information. This .eld indicates the nature of the information carried in this TLV-tuple. Partnerinformation is identi.ed by the integer value 0x02.p) Partner_Information_Length. This .eld indicates the length (in octets) of this TLV-tuple, Partner information uses a lengthvalue of 20 (0x14).q) Partner_System_Priority. The priority assigned to the Partner System (by management or administration policy), encoded asan unsigned integer.r) Partner_System. The Partner’s System ID, encoded as a MAC address.s) Partner_Key. The operational Key value assigned to the port associated with this link by the Partner, encoded as an unsigned

integer.t) Partner_Port_Priority. The priority assigned to this port by the Partner (by management or administration policy), encodedas an unsigned integer.u) Partner_Port. The port number associated with this link assigned to the port by the Partner, encoded as an unsigned integer.v) Partner_State. The Actor’s view of the Partner’s state variables, depicted in Figure 43–8 and encoded as individual bitswithin a single octet, as de.ned for Actor_State.w) Reserved. These 3 octets are reserved for use in future extensions to the protocol. They shall be ignored on receipt and shallbe transmitted as zeroes to claim compliance with Version 1 of this protocol.x) TLV_type = Collector Information. This .eld indicates the nature of the information carried in this TLV-tuple. Collectorinformation is identi.ed by the integer value 0x03.CSMA/CD IEEE Std 802.3-2002®, Section Threey) Collector_Information_Length. This .eld indicates the length (in octets) of this TLV-tuple. Collector information uses alength value of 16 (0x10).z) CollectorMaxDelay. This .eld contains the value of CollectorMaxDelay (43.2.3.1.1) of the station transmitting theLACPDU, encoded as an unsigned integer number of tens of microseconds. The range of values for this parameter is 0 to 65535 tens of microseconds (0.65535 seconds).




Notes: Notes:


Link and Aggregator System IdentifiersLink and Aggregator System Identifiers

• Example

In order to allow for convenient transcription and interpretation by human networkpersonnel, this standard provides a convention for representing compound LAGIDs. Using this format

a) All fields are written as hexadecimal numbers, 2 digits per octet, in canonicalformat.

b) Octets are presented in order, from left to right. Within .elds carrying numerical

signiifcance (e.g.,priority values), the most significant octet is presented first, and the least signi.cantoctet last.

c) Within .elds that carry MAC addresses, successive octets are separated by dashes(-), in accordance

with the hexadecimal representation for MAC addresses defined in IEEE Std 802-1990.

d) Parameters of the LAG ID are separated by commas.




Notes: Notes:


Aggregator GroupingAggregator Grouping

• Links with the same key are candidates for membership of a link aggigatorgroup

•

In configuration it is possible to specify individual ports or any in a group — Any can be specified by using zero in the port number• The number of active links from a group can be configured

— Ports are then activated in sequence of their priority and port number — Low numbers win

• As groups are formed by specifying both ends, if links are miss connectedthey appear as different groups — There are 4 groups here

A B

1 4445

555

112

222

By labelling each element in the group at both ends and passing this informationbetween the adjacent switches it is possible for the switches to detect and confirmthe correct cable attachment for each element in each group. Where a cable isincorrectly connected by mistake, the switches at each end can detect this and eitherinform the management system, report errors on the switch console control interfaceor intelligently reconfigure the operation of the switches algorithmically.




Notes: Notes:



IEEE 802 Addressing


Spanning Tree

Aggregation

Chapter Summary






Notes: Notes:

Layer 3 AddressingLayer 3 AddressingLayer 3 Addressing

Chapter 5Chapter 5




Notes: Notes:



In this chapter, we will

• Review how addressing works at layer 2 and layer 3

• Examine addressing and routing in IP networks

• Explore the operation of MAC addresses

• Resolve addresses between layer 2 and layer3






Notes: Notes:


Routable Address StructureRoutable Address Structure

• To deliver a packet, a router must know where to deliver it

• IP addresses are divided into two fields

— Network—where to deliver the packet (usually a LAN) — Host—which machine at that location• The division of IP addresses is undertaken by a network mask

• Mask (netmask or subnet mask) contains binary 1s in the network portion — Used to indicate the division of network and host

Network Host

192 . 168 . 1 .10

255.255.255 .0Mask

With 2^32 possible destinations it might be necessary somewhere within theInternet to hold a table of possible addresses that was 2^32 rows long. This wouldbe too large to be practical. To make this problem more manageable, addresses aregrouped together into networks where every device on the same network has thesame value in the first part of their addresses. This is known as the network addressor the prefix.

he length of the prefix can vary from one part of the Internet to another and so weneed to define how long this is within the routing table of any device. This is donebe using either binary mask with binary 1s set in the network portion and zerosfollowing this. Another method used is to indicate the length of the networkportion in bits. In this example I could say:-

address 192.168.1.10 with mask 255.255.255.0

or I could say 192.168.1.10/24

Notice that 255.255.255.0 contains 3 bytes full of 1s making a total of 24 bits.




Notes: Notes:


Net Prefix NotationNet Prefix Notation

• Net prefix notation is a shorthand way of describing address and mask

• 192.168.1.10/24 is equivalent to

— Address = 192.168.1.10 — Mask = 255.255.255.0• Value after the “/” gives the number of “1s” in the mask

• Trailing zeros in the address can be removed — 122.15.0.0/16 can be written as 122.15/16 — 0.0.0.0/0 can be written as 0/0

Net prefix notation is the name given to the method of expressing addresses andmasks in the compact form such as 192.168.1.10/24.




Notes: Notes:


Networks and SubnetworksNetworks and Subnetworks

• Networks are identified by addresses — Each network must have unique ID — Each host must have unique ID and same prefix

• Networks and subnetworks are joined by routers

LAN ALAN B

192.168.0.1 192.168.0.2 192.168.0.10

Mask = 255.255.255.0192.168.1.1 192.168.1.2 192.168.1.10

Mask = 255.255.255.0

192.168.1.254

192.168.0.254

192.168.4.1

192.168.4.2

Within any one organization a specific range of addresses will be used. This rangewill be allocated to the organization either directly by the Internet administrativeauthorities, or via their Internet Service Provider.

Inside the organization however it might still be necessary to further sub-divide thenetwork address in order to refer to different LANs within a building. This isknown as sub-networking.




Notes: Notes:


Addressing for IPAddressing for IP

• IP phones need several pieces of information to work — An IP address — The mask for its network/subnetwork — The address of a router on its subnetwork

– Often called a Default Gateway

• IP addresses must be unique to connect to the Internet — Need a numbering plan — Address assignment can be

– Static—by administrator – Dynamic; e.g., Dynamic Host Configuration Protocol (DHCP)

For any device to be able to route data to other parts of the Internet clearly it musthave an IP address. However it also needs to know other things too. It must knowits network mask so that it can distinguish between local devices on the same LAN(the same sub-network), and the address of the interface on a router attached to itsLAN so that it can send data to other networks. It may also need other information

such as details of the server which can map names to IP addresses for Email andWeb access.

These values can be set manually although this can be time consuming and errorprone. More usually the details are sent to the device when it is first turned onusing DHCP. When the device starts up it sends out a single broadcast frame thatevery device on the LAN will examine. One device, the DHCP server, will respondto this and forward the required information including the IP address it should use.




Notes: Notes:



Routable Addresses

Address Classes (Historic)

Issuing Addresses and Resolving them

Subnetworking

Chapter Summary




Notes: Notes:


Interconnecting to the InternetInterconnecting to the Internet

• Choice between public and private (RFC 1918) IP address — If the device has a private address it cannot connect directly to the Internet

– Network Address Translation (NAT) can translate addresses – Allows a limited set of addresses to be shared by a larger community – May improve immunity to attack from hackers

• Address structure — Historically, the division of addresses was fixed in structure

– Classes A, B, and C — Today we use classless addressing, which allows flexible division

• Today we use classless addressing—before, we used address classes

There are now millions of devices attached to the Internet and it is through by somepeople that soon we will run out of IP addresses. Organizations may find that theyare limited in the number of addresses they can use. In 1992 the Internetcommunity recognized that this will become a problem. Many organizations hadrequested addresses in case they wished in the future to use Internet services buthad never connected. RFC 1918 allocated 3 ranges of addresses that will never beused on the Internet so that private networks could use these addresses withoutusing up address capacity. Also firewalls are normally deployed to protectorganizations from attack by hackers. These devices can further protect theirorganizations by hiding the real addresses behind the firewall.

Devices are allocated a private address from RFC 1918 and the firewall convertsthis to a valid address dynamically. This is called Network Address Translation. Itis then possible for several devices to share the same address.




Notes: Notes:


Addresses Classes (Historic)Addresses Classes (Historic)

• Special addresses — Local host: 127.0.0.1 — Broadcast:

– Local 255.255.255.255 – Directed; e.g., 192.168.1.255

• Private networks (RFC 1918) — 10.X.X.X Class A — 172.16-30.X.X Class B — 192.168.0-255.X Class C

N

N

N

0xxxxxxxx

10xxxxxxx

110xxxxxx

N

N N

H H H

H H

H

1–126

128–191

192–223

A

B

C

Before 1992 the length of the network portion of addresses was fixed at one of 3sizes. This was changed in 1992 when classless addressing was introduced andmasks became variable in length.




Notes: Notes:


Internet Protocol (IP)Internet Protocol (IP)

• Role of IP is to communicate between hosts — Routable protocol

– Define both network and host address – Other routable protocols

– Novell IPX, Banyan VINES, AppleTalk, OSI IP• Unreliable, best-effort, connectionless delivery

— Delivery of packet is not guaranteed — Reliability is the responsibility of the application

• Reliability is the responsibility of the upper-layer protocols — May be TCP — May be the application

IP evolved over the 1960s and early 1970s until 1976 when the current version,version 4, was produced.

It is a routable datagram protocol that delivers data without any guarantees.






Notes: Notes:


Mapping IP Addresses to Link AddressesMapping IP Addresses to Link Addresses

• Imagine two machines connected to the same link, a LAN perhaps

• Machine A wishes to send an IP datagram to machine B

• It can construct the datagram but must place this inside the link protocol

• How can it find out machine B’s link address?

• Address Resolution Protocol (ARP) provides the answer

Machine AIP Address: 192.168.10.1

Machine BIP Address: 192.168.10.2




Notes: Notes:


Address Resolution ProtocolAddress Resolution Protocol

• RFC 826 ARP provides a means of mapping IP addresses to link addresses

• Machine A broadcasts an ARP request to all the machines on the link

— ARP request includes the IP address of the required machine B• All machines on the link compare the requested IP address with their own

— Only machine B finds a match and responds

Machine AIP Address: 192.168.10.1

Machine BIP Address: 192.168.10.2

Broadcast ARP Request

Broadcast ARP Request806Source A

Dest A ARP Reply806Source B




Notes: Notes:


ARP CacheARP Cache

• Once a matching IP to physical link address mappings can be stored — Generally stored for a few minutes in a RAM arp cache — Long enough to avoid repeated arp requests — Not so long that if the link address or IP address changes it still remains

– Link address might change because of• Contents of your ARP cache can be displayed using the arp command

C:\WINDOWS>arp -a

Interface: 213.122.23.149 on Interface 0x1000002Internet Address Physical Address Type62.248.16.48 20-53-52-43-00-00 dynamic

Interface: 192.168.7.2 on Interface 0x2000003Internet Address Physical Address Type192.168.7.1 00-a0-cc-7b-18-66 dynamic192.168.7.11 00-20-18-71-f0-ea dynamic192.168.7.32 00-40-05-d0-85-6d dynamic

192.168.7.191 00-03-47-0c-ae-40 dynamic




Notes: Notes:


Configuring IP AddressesConfiguring IP Addresses

• A device can obtain its IP address in one of a number of ways — By manual configuration

– A very reliable human configures it for storage on its backing store – This can make changing IP addresses very difficult – Error prone - if the human makes mistakes!

— By using BOOTP - RFC 951 and 1084 – Fixed IP address for all time

– Provides details of router and other information required — By using DHCP

IP UDP BOOTP request BOOTPserver

IP UDP BOOTP reply

Dest IP address = 255.255.255.255Source IP address = 0.0.0.0




Notes: Notes:


Dynamic Host Configuration Protocol (DHCP)Dynamic Host Configuration Protocol (DHCP)

• DHCP extends BOOTP to add a lease time

• Used in most Microsoft environments

• A request to the server yields an offer of an address with a lease time — The lease time limits the time the IP address is used — Can be renewed if server agrees — Allows more machines to exist than IP addresses

– Only active devices need addresses• Very widely used to allocate addresses within windows networks




Notes: Notes:


Can find details with IPCONFIGCan find details with IPCONFIG




Notes: Notes:



Routable Addresses



Subnetworking

Chapter Summary




Notes: Notes:


Dividing Networks - SubnetworksDividing Networks - Subnetworks

• May be administratively convenient or even necessary to divide networks — Insufficient network addresses for each LAN to have a network — Routing between small groups of workstations on LANs — Easier administration

139.21.1.1

139.21.2.1

Rest of the Internet

139.21.1.2

139.21.1.3

139.21.2.2

139.21.2.3

All Traffic to139.21.0.0




Notes: Notes:


Subnetting Classful AddressesSubnetting Classful Addresses

• Before classless addressing network Classes A, B and C were allocated — Needed to divide these so we called the result a subnetwork — Mask used to define the division between subnetwork and host

– Called a subnet mask – Today called a Netmask

Network Subnet Host

1s in mask 0s in mask

IP Address




Notes: Notes:


Division of NetworkDivision of Network

• The division between network (or subnetwork) and host at bit location — Number of host addresses is always a power of 2 — Zero host address identifies the subnet - not used for host — All 1s host address used for broadcast - not used for host — Result is that number of usable host addresses is 2 less than power of 2

• Valid masks — 255.255.255.252 /30 2 valid hosts — 255.255.255.248 /29 6 valid hosts — 255.255.255.240 /28 14 valid hosts — 255.255.255.224 /27 30 valid hosts — 255.255.255.192 /26 62 valid hosts — 255.255.255.128 /25 126 valid hosts — 255.255.255.0 /24 254 valid hosts — 255.255.254.0 /23 510 valid hosts

— 255.255.252.0 /22 1022 valid hosts — 255.255.248.0 /21 2046 valid hosts and so on




Notes: Notes:


How Many Subnets Are Needed?How Many Subnets Are Needed?

• Routers route between networks or subnetworks

• Each interface needs to be on a different network or subnetwork

• How many subnetworks do I need here?

LAN B

LAN A

LAN C




Notes: Notes:


Point to Point LinksPoint to Point Links

• Point to point links count as subnetworks — Routers route between subnets so point to point links must be a subnets — Point to point links have only two addresses

– One for each end – Ideal mask is 255.255.255.252 as just 2 valid addresses – Some router vendors allow unnumbered links

– No need for IP addressbut difficult to test link remotely — Most popular routing protocol - RIP - requires all masks are the same

– Point to point links use up many addresses




Notes: Notes:


Balancing Subnet and Host NeedsBalancing Subnet and Host Needs

• Every Host that is connected to a subnet needs a unique IP address

• Every point to point link, LAN and other network needs a subnet address

• Zero Subnet and all 1s subnet are more difficult to use — Zero subnet address and the network address are the same

– Example – 139.21.0.0/16 is a network address (class B) – 139.21.0.0/24 is a subnet address but its address is the same – Some routers do not allow the use of zero subnet

Rest of the InternetAll Traffic to139.21.0.0

139.21.2.0

139.21.1.0

?




Notes: Notes:


SMIS Inc NetworkSMIS Inc Network

• SMIS Inc has 6 sites each with a router and a LAN

1 2 3

4 5 6




Notes: Notes:


SMIS Inc NetworkSMIS Inc Network

• Each site has the following numbers of computers currently — 1 225 — 2 250 — 3 100 — 4 250 — 5 210 — 6 216

• We have been allocated network address 139.21.0.0/16

• You have been appointed to advise on address allocations

• Answer the following questions — How many subnets do we need — What subnet masks should be used on each subnet — How should the network addresses be defined for each router interface

— How should the addresses be assigned to each host




Notes: Notes:



Routable Addresses



Subnetworking

Chapter Summary




Notes: Notes:



• In this chapter we have

• Reviewed how addressing works at layer 2 and layer 3

• Examined addressing and routing in IP networks

• Explored the operation of MAC addresses

• Resolved addresses between layer 2 and layer3




Notes: Notes:

RoutingRoutingRouting

Chapter 6Chapter 6




Notes: Notes:



• In this chapter we will

• Study the operation of distributed dynamic routing

• Review the major routing protocols

• Examine how distance vector link state and policy based routing functions




Notes: Notes:


RoutingRouting

Distributed Dynamic Routing Principles

RIP

OSPF

BGP4

Chapter Summary




Notes: Notes:


Dynamic Routing PrinciplesDynamic Routing Principles

D

A

B C

C• Consider this network

• Each site has one LAN

IP uses distributed dynamic routing. Each router must maintain its own tables ofrouting information and make decisions about where to send data.

We will take this simple example of 4 routers each connected to a local LAN toillustrate how routing can function.




Notes: Notes:



D

A

B C

AB

C

01

1

ABCD

1101

CD

10

ABC

101

• Routers build tables — Networks they see

Each router first constructs a table of destinations networks it knows. In ourexample we will use hops to indicate the metric used. Zero means no hops to thedestination - it is local. One hop means that we must pass the data to one morerouter for delivery. Initially routers will only know the existence of routers towhich they directly attach. Notice that router A at the bottom is connected directly

to B and C but not directly to D.




Notes: Notes:



D

A

B C

AB

CD

01

1

10

1

11

01

ABCD

1101

011

101

AB

CD

10

11

01

ABCD

101

011

1101

B C

A B

C

A C

10

D

• Exchange tables with — Immediate neighbors

So that indirect connection can be used each router sends a copy of its own routinginformation to each of its direct neighbors. Notice now that A has information forboth B and C. From the information received from router C it can discover theexistence of router D.




Notes: Notes:



D

A

B C

AB

CD

01

12

10

12

11

01

ABCD

1101

0112

1012

AB

CD

22

10

11

01

ABCD

1012

0112

1101

B C

A B

C

A C

2210

D

• Update Tables

When all of the information is exchanged A will discover two possible routes to D.One via B amd one via C. The route via B is 2 hops while that via C only one so itwill prefer that via C and will add one to this to construct its own metric of 2.




Notes: Notes:



D

A

B C

AB

CD

01

12

10

12

11

01

ABCD

1101

0112

1012

AB

CD

22

10

11

01

ABCD

1012

0112

1101

B C

A B

C

A C

2210

D

• Failed link — Remove information — Update links

However if the link from A to C fails the information A has received from C is nolonger useful and is removed. However a route still exists via B.




Notes: Notes:



D

A

B C

AB

CD

01

23

10

12

ABCD

2101

1012

AB

CD

32

10

21

01

ABCD

1012

0123

1101

B

B

C

A C

3210

D

• Updated Information — Routes around failure

The route via B will then be taken as the preferred route for A to use to reach D anddata is sent via B.




Notes: Notes:



D

A

B C

AB

CD

01

23

10

12

ABCD

2101

1012

AB

CD

32

10

21

01

ABCD

1012

0123

1101

B

B

C

A C

3210

D

• Further Failure — Parts of Network Isolated — Further information

If however the B to C link also fails then clearly there will be not possible route toD from A. However the router will not immediately notice this.




Notes: Notes:



D

A

B C

AB

CD

01

45

10

34

ABCD

4301

AB

CD

32

10

43

01

ABCD

1034

0123

B

C

A

3210

D

• Tables Updated

A will continue to receive routes from B and B continue to send them back to A.As time passes the count of hops from A to D will grow from 2 to 3 and then from 3to 4 until the hop count grows large.




Notes: Notes:



D

A

B C

AB

CD

01

67

10

56

ABCD

6501

AB

CD

54

10

43

01

ABCD

1056

0145

B

C

A

5410

D

• Tables Updated again — Hop counts too large — Only 4 hops in network — Nodes unreachable

When the hop count reaches a value larger that the maximum possible count, in ourexample there are only 4 links so a hop count could never exceed 4, the routers willnotice that the routes are unreachable.

Unreachable routes can then be deleted.




Notes: Notes:



D

A

B C

AB

01

10

CD

01

CD

10

01

AB

10

01

B

C

A

10

D

• Tables reduced — Two distinct networks

Notice now the internetwork has divided into two distinct parts.




Notes: Notes:


RoutingRouting


RIP

OSPF

BGP4

Chapter Summary




Notes: Notes:


Distance Vector RoutingDistance Vector Routing

• This is known as distance vector routing - routing by rumor

• First used Routing Information Protocol (RIP)

— Uses metric of Hops — Simple to implement but treats all links as equivalent — Updates routes every 30 seconds — Maximum hop count is 15 - 16 means unreachable

– Maximum of 16 hops from destination — Ages routes not refreshed after 6 updates — All networks have the same mask

– Does not support Variable Length Subnet Masks (VLSM) — Takes long time to update distant routes

– Maximum time is 16 x 30 seconds = 8 minutes — Takes time tocount to infinity (16 hops)

• RIP version 2 overcomes many of the problems

— Not widely supported yet and other options are better

The example we have taken is small but shows how RIP works. RIP was the firstrouting protocol ever used on the Internet but is now only used for small privatenetworks of 50 subnets ot less.




Notes: Notes:


RoutingRouting


RIP

OSPF

BGP4

Chapter Summary




Notes: Notes:


Link State RoutingLink State Routing

• Link state routing is routing using a network map

• Routers pass information about state of links to neighbors

• Neighbors flood the network so every router gets information

• Each router builds a map of the whole network

• Changes in link states are sent immediately and flooded — Fast update of like state changes - typically in seconds — Only updates sent so much less load after initial update

• Most widely used link state routing algorithm is OSPF

Most large private networks would run very inefficiently if they used RIP. As theinter-network grows the routing tables grow too. With 1000 subnets the tableswould be at least 1000 rows long and must be exchanged every 30 seconds. InsteadLink State routing is normally used. This exchanges routes only when their statuschanges. If there are no changes then there is no traffic.




Notes: Notes:


R1 R4

R2 R3

Net 1

Net 4

Net 2

Net 5Net 3

Topological Map

R4R1

R2 R3

Open Shortest Path First (OSPF)Open Shortest Path First (OSPF)

• Every router has a complete topological map of the network — Routers correspond to nodes in a graph

– Networks are arcs or links between nodes• – Routers are neighbors if they share the same link

Net 1

Net 2

Net 5

Net 4

Net 3

OSPF is the most widely used routing protocol for medium and large organizations.Most ISPs use this internally.




Notes: Notes:


Net 2

Net 3

Net 4

Net 1

R1

R2 R3

R4

Net 5

CRASH Mylinks to Net 3 andNet 4 areup! Sorry,my link toNet 5 isdown.

Mylinks to Net 2 andNet 3 areup!

OSPF Routing UpdatesOSPF Routing Updates

• Routers periodically send (multicast) status of each one of their links — Link status messages are used to update topological map database

– Every router sees the same link-status message – Each calculates shortest path first to all networks

— Route path cost can be a function of what ever manager requires – Hops, bandwidth, delay, protocol priority, usage cost, etc.

It first exchanges details so that every router knows not just what its neighbors cando but gets a copy of details from every router. This takes some time, in practice afew seconds. However this data is sent only at startup. After the initial exchanges,only updates are sent and these are initiated only when the status of a link changes.




Notes: Notes:


What OSPF Can DeliverWhat OSPF Can Deliver

• Type of service routing — Possible to have different routing table for each TOS

• Load balancing — Traffic can be divided between routes of equal metric

• Flexible route path cost — Route path cost can be based on what ever is required

• Unnumbered networks — Point to point links can be configured with taking and IP address

• Multicast IP — Overhead reduced by using multicast addresses

• Fast Convergence — New routes found within seconds after failures

• Hierarchical routing possible using OSPF Areas

OSPF can deliver all of these functions in addition to normal routing of traffic.




Notes: Notes:


RoutingRouting


RIP

OSPF

BGP4

Chapter Summary




Notes: Notes:


Routing LevelsRouting Levels

• Routing within the Internet can be broken down into 3 levels

• First level of routing hierarchy: your PC talks to a router

— Hosts make routing decisions for each IP packet they send out – Is destination directly connected? – Is destination in host’s routing table? – Is there a default router?

— Host-specific routes are checked first• Second level of routing hierarchy: routers within an AS

— Routers communicate within AS with generic Interior Gateway Protocols — Normally this is OSPF or in small networks RIP/EIGRP

• Third level of routing hierarchy: AS to AS — Routers communicate between ASs with generic Exterior Gateway

Protocols — In reality this is BGP4 today

Routing takes place at several levels. At the PC we route data to and from theinterface of the PC and out to the nearest router. Once we reach the first ISP theISP’s routers will follow routes that are advertised by other ISPs that it has abusiness arrangement with.

For commercial reasons, routers within the internet will not follow necessarily the

“shortest” route in all cases. In some cases economics may result in a longer andslower router being used.




Notes: Notes:


Example ASExample AS

• Details of Autonomous System numbers is held at www.ripe.net

• Example•

aut-num : AS12576• as-name: ORANGE-PCS• descr: Orange PCS Limited• import: from AS702 action pref=100; accept ANY• import: from AS2856 action pref=100; accept ANY• export: to AS702 announce AS12576• export: to AS2856 announce AS12576• admin-c: ORA1-RIPE• tech-c: ORA1-RIPE• mnt-by: AS12576-MNT• changed: [email protected] 19990730• changed: [email protected] 20020905• changed: [email protected] 20020905• changed: [email protected] 20040624• source: RIPE

This is an example of a registration from an Autonomous System. Within EuropeRIPE acts as a non-profit organization to register addresses and AS identifications.ISPs attach to switches at general switching points known as Internet Exchanges.

At these points they exchange routes, importing routes from other ISPs (AS) andexporting their own routes.




Notes: Notes:


Example ASExample AS

• Details of Autonomous System numbers is held at www.ripe.net

• Example• aut-num : AS2856• as-name: BT-UK-AS• descr: BTnet UK Regional network• import: from AS286 action pref=11; accept AS-KQ• export: to AS286 announce AS-BTGB• import: from AS702 action pref=11; accept AS-UUNETEUUK• export: to AS702 announce AS-BTGB• import: from AS786 action pref=11; accept AS-JANETPLUS• export: to AS786 announce AS-BTGB

• . . . . .

• import: from AS12576 action pref=10; accept AS12576• export: to AS12576 announce AS-BTGB

Some ISPs specialize in interconnecting the Internet exchanges of the worldforming the high bandwidth backbone connections. Other specialize in providingaccess services to users.






Notes: Notes:


European Internet ExchangesEuropean Internet Exchanges

• European exchanges http://www.dix.dk/euro/

This map shows the major exchanges around Europe. The European InternetExchange Association provides information on the Internet exchanges in Europealthough there is competition between them.




Notes: Notes:


European ExchangesEuropean Exchanges

• European Exchanges are members of European Internet ExchangeAssociation — A few exchanges outside Europe are also associate members

• There are now 38 members

• See http://www.euro-ix.net/about/memberlist.shtml

• Can you spot those in your country?

There are now 38 Internet Exchanges around Europe that are members of theEuropean Internet Exchange Association.




Notes: Notes:


European Internet Exchange AssociationEuropean Internet Exchange Association








Notes: Notes:


Amsterdam Exchange Daily Statistics May 2006Amsterdam Exchange Daily Statistics May 2006

• Exchange with the most members is in Amsterdam — Largest number of ISP peering in Europe — 2nd largest volume but growing fast

Amsterdam has the largest number of ISPs connected, though not the highestthroughput in Europe.

This diagram at

http://www.ams-ix.net/technical/stats/

shows the traffic statistics in May 2006.




Notes: Notes:


LINX Performance: May 2006LINX Performance: May 2006

• LINX in London has higher throughput and is largest in volume terms

• LINX statistics https://www.linx.net

Detailed statisitics of Internet traffic through LINX can be found athttps://www.linx.net/www_public/our_network/traffic_statistics_old/view?searchterm=stats




Notes: Notes:


LINX Yearly 1 Day Average To May 2006LINX Yearly 1 Day Average To May 2006

When LINX stated publishing Annual summaries of their statistics in the Year 2000the traffic was doubling every six months and the daily total stood at below 3Gbit/s. Today it is growing proportionately more slowly but has reached nearly 100Gbit/s.




Notes: Notes:


LINX Growth Over 5 yearsLINX Growth Over 5 years

• There has been a 10 fold increase over the last 4 years — Before this year on year growth was running at 300%

• Can you predict the traffic for 2015?20052004200320022001 2006

Notice how the traffic continues to grow year on year. This is NOT the total traffic,it represents only the traffic between ISPs though London. Data starting and endingwithin the same ISP will not be included.

Notice that traffic continues to increase year on year.




Notes: Notes:


Number of Computers on the InternetNumber of Computers on the Internet

1981 2131982 2351983 5621984 1,0241985 1,9611986 5,0891987 28,1741988 33,0001989 159,0001990 313,0001991 617,0001992 1,136,0001993 2,056,0001994 3,864,0001995 6,642,0001996 16,729,0001997 26,053,0001998 36.739,0001999 56,218,0002000 93,047,7852001 125,888,1972002 162,128,4932003 171,638,2972004 285,139,1072005 317,646,084

The number of computers attached to the Internet increases also.

The key question that is now being asked is when will we run out of IP addresses?




Notes: Notes:


Growth in the InternetGrowth in the Internet

• The Internet has grown tremendously since 1990 — 32 bit address space was not allocated well initially

• Demand for addresses was slowing because of firewall and proxy use — Rate has started to increase again — can you predict when we will run out

of the 4,000,000,000 addresses?

Twice each year the ISC publishes a survey of the number of host addressesactually used in packets passing through the core of the Internet. This can be foundat:-

http://www.isc.org/index.pl?/ops/ds/

While this is large and growing, we have still only used 315 out of the more than4000 million addresses or about 8%.

When you have spent 8% of your monthly pay, have you run out of money? If not,then at which point do you consider yourself out of cash - 90% say?

So how long do you think it will take before we use another 3000 millionaddresses?

In the past we have found more and more ways to reduce the demand for addressesand this can continue for some years yet.




Notes: Notes:


Routing Between ISPsRouting Between ISPs

• OSPF provides a good solution for large Autonomous Systems — Can cope with hundreds of networks and routers — Can be scaled using areas

• Between ISPs it is not enough to just find the shortest route — Commercial decisions mean that shortest is not always best — ISP may gain revenue from its users — Does not want to carry transit traffic unless there is a commercial reason

My ISP Your ISP

Transit ISPs

Not this wayThankyou!

OSPF is no use however between commercial entities such as ISPs. ISPs are notinterested in the shortest or best link if this cost them income. Business andpolitical decisions need policy based routing.




Notes: Notes:


Border Gateway Protocol (BGP)Border Gateway Protocol (BGP)

• BGP enables a manager to decide the policy for carrying traffic — Current version is version 4 (BGP4) — Route from end to end is defined so that every ISP (AS) knows total route — If the route passes through an ISP which is not supported it can be rejected — ISPs can enter into commercial agreements to support traffic passage

• Enables a manager to provide quality of service to supported customers

• Routes may be summarized to reduce routing tables

• Scalable to large number of Autonomous Systems

Border Gateway protocol version 4 is now required between ISPs. Most ISPsinterconnect via Internet Exchanges which are layer 2 switches that join manyrouters together from different ISPs. These route according to policies that reflecttheir commercial interests. Broadly speaking the policy is “ you pay me and I willgive you a route to carry the data”.




Notes: Notes:


Routing Survival KitRouting Survival Kit

• What you need to know to make routing work:-

• Devices must have unique IP address

• Prefix length (subnet mask) must be valid and consistent with address

• Default router must be defined on the same subnet

• Routing table must be valid: Use netstat -r to check — In a routershow ip route

• Must have route to destination: use tracert to check it exists — Check routing tables in devices where route is incorrect




Notes: Notes:


Trace RouteTrace Route




Notes: Notes:


Routing TableRouting Table




Notes: Notes:


PathpingPathping

Pathping provides a full tracerout to the destination and then undertakes a ping testprogressively over the path. This enables details of packet loss to be determined sothat particular error prone links or nodes may be identified. Most losses actuallycome from congestion causing queue buffers to overflow. There is no way to tellthe actual cause of the packet los but it is possible at least to tie down the suspectlinks and routers.




Notes: Notes:


RoutingRouting


RIP

OSPF

BGP4

Chapter Summary




Notes: Notes:



• In this chapter we have

• Studied the operation of distributed dynamic routing

• Reviewed the major routing protocols

• Examined how distance vector link state and policy based routingfunctions




Notes: Notes:

Multicasting and Stream DeliveryMulticasting and Stream DeliveryMulticasting and Stream Delivery

Chapter 7Chapter 7




Notes: Notes:




• Capture multicast streams

• Identify different multicast address standards

• Analyze IGMP and PIM

• Differentiate between IGMP versions 1, 2 and 3

• Troubleshoot multicasting services




Notes: Notes:


Multicasting and Stream DeliveryMulticasting and Stream Delivery

Multicast ConceptsMulticast Addressing

IGMP

PIM Sparse Mode Configuration

Analysis of Multicast Exchanges

Troubleshooting Multicast Problems

Chapter Summary




Notes: Notes:


Unicast vs MulticastUnicast vs Multicast

Unicast

Multicast

• Unicast transmission sends multiple copies of data, one copy for

each receiver

– Ex: host transmits 3 copies of data and network forwards each to 3separatereceivers

– Ex: host can only send to one receiver at a time

• Multicast transmission sends a single copy of data to multiple

receivers

– Ex: host transmits 1 copy of data and network replicates at last possible hop foreach receiver, each packet exists only one time on any given net work

– Ex: host can send to multiple receivers simultaneously




Notes: Notes:


Multicast AdvantagesMulticast Advantages

As the number of viewers receiving the service increases, multicast deliverybecomes dramatically more efficient. It is really not feasible to deliver broadcasttelevision to a nation via Unicast, however via multicast it would be feasible todeliver dozens or perhaps a hundred channels that way.




Notes: Notes:


Multicast DisadvantagesMulticast Disadvantages

• Best Effort Delivery: Drops are to be expected — Multicast applications should not expect reliable delivery of data and should

be designed accordingly — Reliable Multicast is still an area for much research

• No Congestion Avoidance: Lack of TCP windowing and “slow-start”mechanisms can result in network congestion — Multicast applications should attempt to detect and avoid congestion

conditions• Duplicates: Some multicast protocol mechanisms (e.g. Asserts, Registers

and Shortest-Path Tree Transitions) result in the occasional generation ofduplicate packets — Multicast applications should expect occasional duplicate packets

• Out-of-Sequence Packets: Various network events can result in packetsarriving out of sequence.

— Multicast applications should be designed to handle packets that arrive insome other sequence than they were sent by the source

There are however disadvantages with multicast. By its very nature the trafficcannot be acknowledge so the service is less reliable.




Notes: Notes:




IGMP




Chapter Summary




Notes: Notes:


IP Multicast Service ModelIP Multicast Service Model

• RFC 1112 (Host Ext. for Multicast Support)

• Each multicast group identified by a class-D IP address

• Members of the group could be present anywhere in the Internet

• Members join and leave the group and indicate this to the routers

• Senders and receivers are distinct: — i.e., a sender need not be a member

• Routers listen to all multicast addresses and use multicast routingprotocols to manage groups

RFC 1112 is the Internet Group Management Protocol (IGMP) . It allows hosts to join a group that receives multicast packets. Users can dynamically register(join/leave multicast groups) based on the applications they execute It uses IPdatagrams to transmit data

Addressing

Class D IP addresses (224-239) are dynamically allocated. Multicast IP addresses

represent receiver groups, not individual receivers.Group Membership

Receivers can be densely or sparsely distributed throughout the Internet. Receiverscan dynamically join/leave a multicast session at any time using IGMP to managegroup membership within the routers. Senders are not necessarily included in themulticast group they are sending to. Many applications have the characteristic ofreceivers also becoming senders eg

RTCP streams from IP/TV clients






Notes: Notes:


Address Usage

224.0.0.1 All systems on this subnet

224.0.0.2 All routers on this subnet

224.0.0.5 OSPF routers

224.0.0.6 OSPF designated routers

224.0.0.12Dynamic Host Configuration Protocol (DHCP)server/relay agent

Reserved Link Local AddressesReserved Link Local Addresses

The IANA has reserved addresses in the range 224.0.0.0/24 to be used by networkprotocols on a local network segment. Packets with these addresses should never beforwarded by a router. Packets with link local destination addresses are typicallysent with a time-to-live (TTL) value of 1 and are not forwarded by a router.

Network protocols use these addresses for automatic router discovery and tocommunicate important routing information. For example, Open Shortest Path First

(OSPF) uses the IP addresses 224.0.0.5 and 224.0.0.6 to exchange link-stateinformation. This lists some well-known link local IP addresses.




Notes: Notes:


Globally Scoped AddressesGlobally Scoped Addresses

• IANA has assigned multicast addresses globally to a number of protocols

• They are in the 224.0.0.0/16 range - several hundred in all!

• Routers should NOT forward streams sent to these addresses

• Examples:-

224.0.0.0 Base Address (Reserved) [RFC1112,JBP]224.0.0.1 All Systems on this Subnet [RFC1112,JBP]224.0.0.2 All Routers on this Subnet [JBP]224.0.0.3 Unassigned [JBP]224.0.0.4 DVMRP Routers [RFC1075,JBP]224.0.0.5 OSPFIGP OSPFIGP All Routers [RFC2328,JXM1]224.0.0.6 OSPFIGP OSPFIGP Designated Routers [RFC2328,JXM1]224.0.0.7 ST Routers [RFC1190,KS14]224.0.0.8 ST Hosts [RFC1190,KS14]

224.0.0.9 RIP2 Routers [RFC1723,GSM11] 224.0.0.10 IGRP Routers [Farinacci]

Addresses in the range from 224.0.1.0 through 238.255.255.255 are called globallyscoped addresses. These addresses are used to multicast data between organizationsand across the Internet.

Some of these addresses have been reserved for use by multicast applicationsthrough IANA. For example, IP address 224.0.1.1 has been reserved for NetworkTime Protocol (NTP).

IP addresses reserved for IP multicast are defined in RFC 1112, Host Extensions forIP Multicasting. More information about reserved IP multicast addresses can befound at the following location:http://www.iana.org/assignments/multicast-addresses




Notes: Notes:


Source Specific Multicast (SSM)Source Specific Multicast (SSM)

• Addresses in the range 232/8 are reserved for source specific Multicast

• Useful when several sources use the same multicast destination for

different applications• End systems can distinguish application protocols but network efficiency

suffers

• By requesting streams based on the full (S, G) identity this is avoided

• The receiver application sends a join a particular source by using theINCLUDE mode in IGMPv3 — The multicast router can now send the request directly to the source rather

than send the request to a common RP as in PIM sparse mode

If two applications with different sources and receivers use the same IP multicastgroup address, receivers of both applications will receive traffic from the senders ofboth the applications. Even though the receivers, if programmed appropriately, canfilter out the unwanted traffic, this situation still would likely generate noticeablelevels of unwanted network traffic.

In an SSM-enhanced multicast network, the router closest to the receiver will "see"

a request from the receiving application to join to a particular multicast source. Thereceiver application then can signal its intention to join a particular source by usingthe INCLUDE mode in IGMPv3. The multicast router can now send the requestdirectly to the source rather than send the request to a common RP as in PIM sparsemode. At this point, the source can send data directly to the receiver using theshortest path. In SSM, routing of multicast traffic is entirely accomplished withsource trees. There are no shared trees and therefore an RP is not required.

SSM also solves IP multicast address collision issues associated with one-to-manytype applications. Routers running in SSM mode will route data streams based onthe full (S, G) address, S making the entry unique.




Notes: Notes:


GLOP AddressesGLOP Addresses

• GLOP is not an acronym

• It is a means of assigning addresses in 233/8

• They are based on an autonomous system number

• For a given ASN number, converted into two octets (say X and Y)

• The GLOP space is therefore 233.X.Y/24

RFC 2770, GLOP Addressing in 233/8, proposes that the 233.0.0.0/8 address rangebe reserved for statically defined addresses by organizations that already have anAS number reserved. This practice is called GLOP addressing. The AS number ofthe domain is embedded into the second and third octets of the 233.0.0.0/8 addressrange. For example, the AS 62010 is written in hexadecimal format as F23A.Separating the two octets F2 and 3A results in 242 and 58 in decimal format. Thesevalues result in a subnet of 233.242.58.0/24 that would be globally reserved for AS62010 to use.






Notes: Notes:


Layer 2 Multicast AddressesLayer 2 Multicast Addresses

• Multicast MAC addresses have the lowest order bit in the first byte set

• If they are locally administered then the next lowest is set too

XXXXXX11 XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX XXXXXXXX

Broadcast or multicast

Locally administered address

In IP multicast, several hosts need to be able to receive a single data stream with acommon destination MAC address. Some means had to be devised so that multiplehosts could receive the same packet and still be able to differentiate betweenseveral multicast groups. One method to accomplish this is to map IP multicastClass D addresses directly to a MAC address. Today, using this method, NICs canreceive packets destined to many different MAC addresses—their own unicast,broadcast, and a range of multicast addresses.The IEEE LAN specifications madeprovisions for the transmission of broadcast and multicast packets. In the 802.3standard, bit 0 of the first octet is used to indicate a broadcast or multicast frame.

The IANA owns a block of Ethernet MAC addresses that start with 01:00:5E inhexadecimal format. Half of this block is allocated for multicast addresses. Therange from 0100.5e00.0000 through 0100.5e7f.ffff is the available range ofEthernet MAC addresses for IP multicast.This allocation allows for 23 bits in theEthernet address to correspond to the IP multicast group address. The mappingplaces the lower 23 bits of the IP multicast group address into these available 23bits in the Ethernet address




Notes: Notes:


Layer 2 Multicast AddressesLayer 2 Multicast Addresses

• IANA has devised a mechanism for generating multicast MAC addresses

• IANA has allocated to it the prefix 01-00-5e

• The lower 23 bits of the IP multicast group address are used in the lowest23 bits — This is not perfect as 32 multicast addresses map to each MAC address — Care should be taken when selecting addresses to keep the lowest 3 bytes

unique

Because the upper five bits of the IP multicast address are dropped in this mapping,the resulting address is not unique. In fact, 32 different multicast group IDs map tothe same Ethernet address . Network administrators should consider this fact whenassigning IP multicast addresses.

For example, 224.1.1.1 and 225.1.1.1 map to the same multicast MAC address on aLayer 2 switch. If one user subscribed to Group A (as designated by 224.1.1.1) and

the other users subscribed to Group B (as designated by 225.1.1.1), they would bothreceive both A and B streams. This situation limits the effectiveness of thismulticast deployment.




Notes: Notes:


Multicasting At Layer 2Multicasting At Layer 2

• There are 3 methods of controlling multicasting at layer 2 — Cisco Group Management Protocol (CGMP) — IGMP Snooping — Router-Port Group Management Protocol (RGMP)

– Used on routed segments that contain only routers• CGMP and IGMP Snooping are used on subnets that include end users or

receiver clients

The default behavior for a Layer 2 switch is to forward all multicast traffic to everyport that belongs to the destination LAN on the switch. This behavior reduces theefficiency of the switch, whose purpose is to limit traffic to the ports that need toreceive the data.

Three methods efficiently handle IP multicast in a Layer 2 switchingenvironment—Cisco Group Management Protocol (CGMP), IGMP Snooping, and

Router-Port Group Management Protocol (RGMP). CGMP and IGMP Snooping areused on subnets that include end users or receiver clients. RGMP is used on routedsegments that contain only routers, such as in a collapsed backbone.






Notes: Notes:


Requests for Multicasts StreamsRequests for Multicasts Streams

The receivers indicate their interest by sending an Internet Group ManagementProtocol (IGMP) host report to the routers in the network. The routers are thenresponsible for delivering the data from the source to the receivers. The routers useProtocol Independent Multicast (PIM) to dynamically create a multicast distributiontree. The video data stream will then be delivered only to the network segments thatare in the path between the source and the receivers. This process is furtherexplained in the following sections.

Multicast is based on the concept of a group. A multicast group is an arbitrarygroup of receivers that expresses an interest in receiving a particular data stream.This group has no physical or geographical boundaries—the hosts can be locatedanywhere on the Internet or any private internetwork. Hosts that are interested inreceiving data flowing to a particular group must join the group using IGMP .




Notes: Notes:


Multicast ConceptsMulticast Concepts

IP multicast is a bandwidth-conserving technology that reduces traffic bysimultaneously delivering a single stream of information to potentially thousands ofcorporate recipients and homes.IP multicast delivers application source traffic to multiple receivers withoutburdening the source or the receivers while using a minimum of networkbandwidth. Multicast packets are replicated in the network at the point where pathsdiverge by routers enabled with Protocol Independent Multicast (PIM) and othersupporting multicast protocols, resulting in the most efficient delivery of data tomultiple receivers. Many alternatives to IP multicast require the source to sendmore than one copy of the data. Some, such as application-level multicast, requirethe source to send an individual copy to each receiver. Even low-bandwidthapplications can benefit from using IP multicast when there are thousands ofreceivers. High-bandwidth applications, such as MPEG video, may require a largeportion of the available network bandwidth for a single stream. In theseapplications, IP multicast is the only way to send to more than one receiversimultaneously. The figure shows how IP multicast is used to deliver data from onesource to many interested recipients




Notes: Notes:


Internet Group Management Protocol (IGMP)Internet Group Management Protocol (IGMP)

• IGMP is used to dynamically register individual hosts in a multicast group — Normally works on a particular LAN

• There are 3 versions of IGMP• Version 1 has only 2 message types

— Membership query — Membership report

• Version 2 has 4 message types — Membership query — Version 1 membership report — Version 2 membership report — Leave group

• Version 3 adds a further Version 3 membership report

IGMP is used to dynamically register individual hosts in a multicast group on aparticular LAN. Hosts identify group memberships by sending IGMP messages totheir local multicast router. Under IGMP, routers listen to IGMP messages andperiodically send out queries to discover which groups are active or inactive on aparticular subnet.In version 1 IGMP is used to dynamically register individual hosts in a multicastgroup on a particular LAN. Hosts identify group memberships by sending IGMP

messages to their local multicast router. Under IGMP, routers listen to IGMPmessages and periodically send out queries to discover which groups are active orinactive on a particular subnet.IGMPv1 has been superceded by IGMP Version 2 (IGMPv2), which is now thecurrent standard. IGMPv2 is backward compatible with IGMPv1. RFC 2236,Internet Group Management Protocol, Version 2, describes the specification forIGMPv2The main difference is that there is a leave group message. With this message, thehosts can actively communicate to the local multicast router that they intend toleave the group.




Notes: Notes:


IGMP Version 3 RFC 3376IGMP Version 3 RFC 3376

• IGMPv3 query message enables more reliable group control• 0x11 Membership Query• 0x22 Version 3 Membership Report• 0x12 Version 1 Membership Report [RFC-1112]• 0x16 Version 2 Membership Report [RFC-2236]• 0x17 Version 2 Leave Group [RFC-2236]

Query Request Query Report

Must also supportthese for compatabilitywith versions 1 and 2

IGMP Version 3 (IGMPv3) is the next step in the evolution of IGMP. IGMPv3 addssupport for "source filtering," which enables a multicast receiver host to signal to arouter the groups from which it wants to receive multicast traffic, and from whichsources this traffic is expected. This membership information enables Cisco IOSsoftware to forward traffic from only those sources from which receivers requestedthe traffic.

IGMPv3 is an emerging standard. The latest versions of Windows, Macintosh, andUNIX operating systems all support IGMPv3. At the time this document was beingwritten, application developers were in the process of porting their applications tothe IGMPv3 API.




Notes: Notes:


IGMPv3 ModesIGMPv3 Modes

• IGMPv3 can operate on one of two modes — INCLUDE mode—the receiver announces membership to a host group

– It provides a list of source addresses known as the INCLUDE list — EXCLUDE mode—the receiver announces membership to a multicast

group

– It provides a list of source addresses known as the EXCLUDE list

– These are addresses from which it does not want to receive traffic

– To receive traffic from all sources an empty EXCLUDE list is used

– This is the behavior of IGMPv2

IGMPv3 supports applications that explicitly signal sources from which they wantto receive traffic. With IGMPv3, receivers signal membership to a multicast hostgroup in the following two modes:

INCLUDE mode—In this mode, the receiver announces membership to a hostgroup and provides a list of source addresses (the INCLUDE list) from which itwants to receive traffic.

EXCLUDE mode—In this mode, the receiver announces membership to a multicastgroup and provides a list of source addresses (the EXCLUDE list) from which itdoes not want to receive traffic. The host will receive traffic only from sourceswhose IP addresses are not listed in the EXCLUDE list. To receive traffic from allsources, which is the behavior of IGMPv2, a host uses EXCLUDE modemembership with an empty EXCLUDE list.The current specification for IGMPv3 can be found in the Internet EngineeringTask Force (IETF) draft titled Internet Group Management Protocol, Version 3 onthe IETF website.




Notes: Notes:


IGMP State Maintained by Multicast RoutersIGMP State Maintained by Multicast Routers

• Multicast routers implementing IGMPv3 keep state per group per attachednetwork

•

state conceptually consists of a set of records of the form: — (multicast address, group timer, filter-mode, (source records))• Each source record is of the form:

— (source address, source timer)• If all sources within a given group are desired, an empty source record list

is kept with filter-mode set to EXCLUDE

Multicast routers send Host Membership Query messages (hereinafter calledQueries) to discover which host groups have members on their attached localnetworks. Queries are addressed to the all-hosts group (address 224.0.0.1), andcarry an IP time-to-live of 1.

Hosts respond to a Query by generating Host Membership Reports reporting eachhost group to which they belong on the network interface from which the Query

was received.




Notes: Notes:


Router Filter-ModeRouter Filter-Mode

• IGMPv3 routers keep a filter-mode per group per attached network

• When a group record is received, the router filter-mode for that group is

updated to cover all the requested sources• When a router filter-mode for a group is EXCLUDE, the source record list

contains two types of sources — the set which represents conflicts in the desired reception state — the set of sources which hosts have requested to not be forwarded

• When a router filter-mode for a group is INCLUDE, the source record list isthe list of sources desired for the group

In order to avoid an "implosion" of concurrent Reports and to reduce the totalnumber of Reports transmitted, two techniques are used:When a host receives a Query, rather than sending Reports immediately, it starts areport delay timer for each of its group memberships on the network interface of theincoming Query. Each timer is set to a different, randomly-chosen value betweenzero and D seconds. When a timer expires, a Report is generated for thecorresponding host group. Thus, Reports are spread out over a D second intervalinstead of all occurring at once.A Report is sent with an IP destination address equal to the host group addressbeing reported, and with an IP time-to-live of 1, so that other members of the samegroup on the same network can overhear the Report. If a host hears a Report for agroup to which it belongs on that network, the host stops its own timer for thatgroup and does not generate a Report for that group. Thus, in the normal case, onlyone Report will be generated for each group present on the network, by the memberhost whose delay timer expires first. Note that the multicast routers receive all IPmulticast datagrams, and therefore need not be addressed explicitly. Further notethat the routers need not know which hosts belong to a group, only that at least one

host belongs to a group on a particular network.




Notes: Notes:


IGMP StatesIGMP States

A host may be in one of three possible states, with respect to any single IPhost group on any single network interface

•

Non-Member state — The host does not belong to the group on the interface. This is the initialstate for all memberships on all network interfaces; it requires no storage inthe host.

• Delaying Member state — The host belongs to the group on the interface and has a report delay timer

running for that membership.• Idle Member state

— The host belongs to the group on the interface and does not have a reportdelay timer running for that membership

Multicast routers send Queries periodically to refresh their knowledge ofmemberships present on a particular network. If no Reports are received for aparticular group after some number of Queries, the routers assume that that grouphas no local members and that they need not forward remotely-originatedmulticasts for that group onto the local network. Queries are normally sentinfrequently (no more than once a minute) so as to keep the IGMP overhead onhosts and networks very low. However, when a multicast router starts up, it mayissue several closely-spaced Queries in order to build up its knowledge of localmemberships quickly.

When a host joins a new group, it should immediately transmit a Report for thatgroup, rather than waiting for a Query, in case it is the first member of that groupon the network. To cover the possibility of the initial Report being lost or damaged,it is recommended that it be repeated once or twice after short delays. A simple wayto accomplish this is to act as if a Query had been received for that group only,setting the group's random report delay timer. Note that, on a network with nomulticast routers present, the only IGMP traffic is the one or more Reports sentwhenever a host joins a new group.




Notes: Notes:


Events Causing State TransitionsEvents Causing State Transitions

• "join group" occurs when the host decides to join the group on theinterface.

•

"leave group" occurs when the host decides to leave the group on theinterface.

• "query received" occurs when the host receives a valid IGMP HostMembership Query message.

• "report received" occurs when the host receives a valid IGMP HostMembership Report message.

• "timer expired" occurs when the report delay timer for the group on theinterface expires. It may occur only in the Delaying Member state

Hosts respond to a Query by generating Host Membership Reports (hereinaftercalled Reports), reporting each host group to which they belong on the networkinterface from which the Query was received. In order to avoid an "implosion" ofconcurrent Reports and to reduce the total number of Reports transmitted, twotechniques are used:

When a host receives a Query, rather than sending Reports immediately, it

starts a report delay timer for each of its group memberships on the networkinterface of the incoming Query. Each timer is set to a different, randomly-chosen value between zero and D seconds. When a timer expires, a Report isgenerated for the corresponding host group. Thus, Reports are spread out overa D second interval instead of all occurring at once.




Notes: Notes:


IGMP ActionsIGMP Actions

• "send report" for the group on the interface

• "start timer" for the group on the interface, using a random delay value

between 0 and D seconds — The maximum report delay D is 10 seconds• "stop timer" for the group on the interface

A Report is sent with an IP destination address equal to the host group addressbeing reported, and with an IP time-to-live of 1, so that other members of thesame group on the same network can overhear the Report. If a host hears aReport for a group to which it belongs on that network, the host stops its owntimer for that group and does not generate a Report for that group. Thus, inthe normal case, only one Report will be generated for each group present onthe network, by the member host whose delay timer expires first. Note that themulticast routers receive all IP multicast datagrams, and therefore need not beaddressed explicitly. Further note that the routers need not know which hostsbelong to a group, only that at least one host belongs to a group on a particularnetwork.




Notes: Notes:


Queries and reportsQueries and reports

• Query Message — Must be at least 8 octets long and have a correct IGMP checksum — Have an IP destination address of 224.0.0.1 — A single Query applies to all memberships on the interface from which the

Query is received — It is ignored for memberships in the Non-Member or Delaying Member state

• Report Message — Must be at least 8 octets long and have a correct IGMP checksum — Contain the same IP host group address in its IP destination field and its

IGMP group address field — A Report applies only to the membership in the group identified by the

Report, on the interface from which the Report is received — It is ignored for memberships in the Non-Member or Idle Member state




Notes: Notes:


IGMP State TrsnsitionsIGMP State Trsnsitions

Non-Member

Idle-MemberDelaying-Member

Leave Group(stop timer) Leave Group

Join Group(send reportstart timer)

Query received(start timer)

Report received(stop timer)

Timer expired(send report)

Multicast routers send Queries periodically to refresh their knowledge ofmemberships present on a particular network. If no Reports are received for aparticular group after some number of Queries, the routers assume that that grouphas no local members and that they need not forward remotely-originatedmulticasts for that group onto the local network. Queries are normally sentinfrequently, no more than once a minute, so as to keep the IGMP overhead onhosts and networks very low. However, when a multicast router starts up, it mayissue several closely-spaced Queries in order to build up its knowledge of localmemberships quickly.When a host joins a new group, it should immediately transmit a Report for thatgroup, rather than waiting for a Query, in case it is the first member of that groupon the network. To cover the possibility of the initial Report being lost or damaged,it is recommended that it be repeated once or twice after short delays. (A simpleway to accomplish this is to act as if a Query had been received for that group only,setting the group's random report delay timer. The state transition diagram belowthis approach.On a network with no multicast routers present, the only IGMP traffic is the one or

more Reports sent whenever a host joins a new group.




Notes: Notes:


Manipulating IGMPManipulating IGMP

access-group IGMP group access group

helper-address IGMP helper address

join-group IGMP join multicast group

last-member-query-interval IGMP last member query interval

querier-timeout IGMP previous querier timeout

query-interval IGMP host query interval

query-max-response-time IGMP max query response value

static-group IGMP static multicast group

unidirectional-link IGMP unidirectional link multicast routing

version IGMP version

These are the qualifiers that can be applied to “ip igmp” commands to manipulateIGMP on Cisco devices..




Notes: Notes:


Limiting the Joining of GroupsLimiting the Joining of Groups

• To filter multicast groups allowed on an interface, use the followingcommand in interface configuration mode: —ip igmp access-group access-list-number

• Multicast routers send host-query messages periodically —This is used to refresh their knowledge of memberships present

• At least one router must be present to produce these queries —Low cost switch vendors often use IGMP spoofing —This will propagate IGMP information in response to queries

• To modify this interval, use the following command in interfaceconfiguration mode: —ip igmp query-interval seconds

Multicast routers send IGMP host-query messages to discover which multicastgroups are present on attached networks. These messages are sent to the all-systemsgroup address of 224.0.0.1 with a TTL of 1.

Multicast routers send host-query messages periodically to refresh their knowledgeof memberships present on their networks. If, after some number of queries, theCisco IOS software discovers that no local hosts are members of a multicast group,

the software stops forwarding onto the local network multicast packets from remoteorigins for that group and sends a prune message upstream toward the source.

Multicast routers elect a PIM designated router for the LAN (subnet). This is therouter with the highest IP address. The designated router is responsible for sendingIGMP host-query messages to all hosts on the LAN. In sparse mode, the designatedrouter also sends PIM register and PIM join messages toward the RP router.




Notes: Notes:


IGMP Versions 1 and 2IGMP Versions 1 and 2

• To control which version of IGMP the router uses, use the followingcommand in interface configuration mode: —ip igmp version {2 | 1}

• To change the query timeout in version 2, use the following command ininterface configuration mode: —ip igmp query-timeout seconds

• To change the maximum query response time, use the followingcommand in interface configuration mode: —ip igmp query-max-response-time seconds

By default, the router uses IGMP Version 2, which allows such features as theIGMP query timeout and the maximum query response time.All routers on the subnet must support the same version. The router does notautomatically detect Version 1 routers and switch to Version 1 as did earlierreleases of the Cisco IOS software. However, a mix of IGMP Version 1 andVersion 2 hosts on the subnet is acceptable. IGMP Version 2 routers will alwayswork correctly in the presence of IGMP Version 1 hosts.You can specify the period of time before the router takes over as the querier for theinterface, after the previous querier has stopped doing so. By default, the routerwaits 2 times the query interval controlled by the ip igmp query-interval command.After that time, if the router has received no queries, it becomes the querier. Thisfeature requires IGMP Version 2. By default, the maximum query response timeadvertised in IGMP queries is 10 seconds. If the router is using IGMP Version 2,you can change this value. The maximum query response time allows a router toquickly detect that there are no more directly connected group members on a LAN.Decreasing the value allows the router to prune groups faster.




Notes: Notes:


Capture of IGMP – Client Starts FirstCapture of IGMP – Client Starts First

In this example we have captured a multicast stream and filtered on IGMP. Westarted VLC which made two IGMPv3 requests in packet 37 and 38. The multicaststream ran until at packet 27365 when the router at 10.0.0.254 sent a membershipreport to 224.0.1.40 using IGMPv2. From this point on the traffic will downgrade toIGMPv2. At 27367 the router sends an IGMPv2 query to 224.0.0.1 – all systems onthe LAN. At packet 27384 the receiver sends a membership report to239.255.255.250 and then immediately after to 255.1.2.3. Following this there arerepeated queries to




Notes: Notes:


Server and Router Start FirstServer and Router Start First

Notice here where the server starts first and the router sends IGMPv2 messagesbefore the client starts, the client starts in IGMPv2. Notice at packet 52 the client

joins the group for 225.1.2.3 by sending a membership report. It repeats this atpackets 197 and 507. We can time the difference between these requests




Notes: Notes:


Setting Time ReferenceSetting Time Reference

By right clicking on packet 52 we can set a time reference so that we can measuretime starting from this point.






Notes: Notes:


IGMP SnoopingIGMP Snooping

• IGMP Snooping is a function of layer 2 switches

• It causes them to look for IGMP membership reports

— These are sent in response to the router IGMP queries• The layer 2 switch delivers multicasts matching the group address

• At least one layer 3 device must exist to send the queries

• The shorter the query interval the more responsive the switch can be

IGMP Snooping is a function of layer 2 switches that causes them to look for IGMPmembership reports that are sent in response to the router IGMP queries.




Notes: Notes:




IGMP




Chapter Summary




Notes: Notes:


Multicast Distribution TreesMulticast Distribution Trees

• Shortest Path or Source Distribution Tree

Shortest Path Trees — aka Source Trees

– A Shortest path or source distribution tree is a minimal spanning tree with thelowest cost from the source to all leaves of the tree.

– We forward packets on the Shortest Path Tree according to both the SourceAddress that the packets originated from and the Group address G that the packetsare addressed to. For this reason we refer to the forwarding state on the SPT by the

notation (S,G) (pronounced “S comma G”).where:

• “S” is the IP address of the source.

• “G” is the multicast group address

– Example 1:

• The shortest path between Source 1 and Receiver 1 is via Routers A and C, andshortest path to Receiver 2 is one additional hop via Router E.




Notes: Notes:



• Shortest Path or Source Distribution Tree

– Every SPT is routed at the source. This means that for every source sending to a

group, there is a corresponding Shortest Path Tree.

– Example 2:

The shortest path between Source 2 and Receiver 1 is via Routers D, F and C,

and shortest path to Receiver 2 is one additional hop via Router E.




Notes: Notes:



• Shared Distribution Tree

Shared Distribution Trees

– Shared distribution tree whose root is a shared point in the network down which

multicast data flows to reach the receivers in the network. In PIM-SM, this shared

point is called the Rendezvous Point (RP).

– Multicast traffic is forwarded down the Shared Tree according to just the Group

address G that the packets are addressed to, regardless of source address. For thisreason we refer to the forwarding state on the shared tree by the notation (*,G)

(pronounced “star comma G”)

where:

• “*” means any source

• “G” is the group address




Notes: Notes:



• Shared Distribution Tree

Shared Distribution Trees

– Before traffic can be sent down the Shared Tree it must somehow be sent to theRoot of the Tree.

– In classic PIM-SM, this is accomplished by the RP joining the Shortest Path Treeback to each source so that the traffic can flow to the RP and from there down theshared tree. In order to trigger the RP to take this action, it must somehow be

notified when a source goes active in the network.• In PIM -SM, this is accomplished by first-hop routers (i.e. the router directlyconnected to an active source) sending a special Register message to the RP toinform it of the active source.

– In the example above, the RP has been informed of Sources 1 and 2 being activeand has subsequently joined the SPT to these sources.




Notes: Notes:


Characteristics of Distribution TreesCharacteristics of Distribution Trees

• Source or Shortest Path trees — Uses more memory O(S x G) but you get optimal paths from source to all

receivers; minimizes delay• Shared trees

— Uses less memory O(G) but you may get sub-optimal paths from source toall receivers; may introduce extra delay

Source or Shortest Path Tree Characteristics

Provides optimal path (shortest distance and minimized delay) from source to allreceivers, but requires more memory to maintain

Shared Tree Characteristics

Provides sub-optimal path (may not be shortest distance and may introduce extradelay) from source to all receivers, but requires less memory to maintain






Notes: Notes:


RPF Check FailureRPF Check Failure

Multicast Forwarding: RPF Check Fails

Ex: Router can only accept multicast data from Source 151.10.3.21 on interface S1

... multicast data is silently dropped because it arrived on an interface not specifiedin the RPF check (S0)






Notes: Notes:


TTL ThresholdsTTL Thresholds

• What is a TTL Threshold? — A “TTL Threshold” may be set on a multicast router interface to limit the

forwarding of multicast traffic to outgoing packets with TTLs greater than theThreshold

• The TTL Threshold Check — 1) All incoming IP packets first have their TTL decremented byone. If <=

Zero, they are dropped. — 2) If a multicast packet is to be forwarded out an interface with a non-zero

TTL Threshold; then it’s TTL is checked against the TTL Threshold. If thepacket’s TTL is < the specified threshold, it is not forwarded out theinterface.

TTL-Thresholds

Non-Zero, Multicast, TTL-Thresholds may be set on any multicast capableinterface.

IP multicast packets whose TTLs (after being decremented by one by normal routerpacket processing) are less than the TTL -Threshold on an outgoing interface, willbe not be forwarded out that interface.

Zero Multicast TTL implies NO threshold has been set.TTL-Threshold Application

Frequently used to set up multicast boundaries to prevent unwanted multicast trafficfrom entering/exiting the network.




Notes: Notes:


TTL Thresholds ExampleTTL Thresholds Example

TTL-Threshold Example

In the above example, the interfaces have been configured with the following TTL -Thresholds:

S1: TTL -Threshold = 16E0: TTL -Threshold = 0 (none)S2: TTL -Threshold = 64

An incoming Multicast packet is received on interface S0 with a TTL of 24.The TTL is decremented to 23 by the normal router IP packet processing.

The outgoing interface list for this Group contains interfaces S1, E0 & S2.

The TTL-Threshold check is performed on each outgoing interface as follows:

S1: TTL (23) > TTL -Threshold (16). FORWARDE0: TTL (23) > TTL -Threshold (0). FORWARDS2: TTL (23) < TTL -Threshold (64). DROP




Notes: Notes:


TTL Threshold BoundariesTTL Threshold Boundaries

TTL-Threshold Boundaries

TTL-Thresholds may be used as boundaries around portions of a network toprevent the entry/exit of unwanted multicast traffic. This requires multicastapplications to transmit their multicast traffic with an initial TTL value set so as tonot cross the TTL -Threshold boundaries.

In the example above, the Engineering or Marketing departments can prevent

department related multicast traffic from leaving their network by using a TTL of15 for their multicast sessions. Similarly, Company ABC can prevent privatemulticast traffic from leaving their network by using a TTL of 127 for theirmulticast sessions.




Notes: Notes:


Administrative BoundariesAdministrative Boundaries

Administrative Boundaries

Administratively-scoped multicast address ranges may also be used as boundariesaround portions of a network to prevent the entry/exit of unwanted multicast traffic.

This requires multicast applications to transmit their multicast traffic with a groupaddress that falls within the Administrative address range so that it will not crossthe Administrative boundaries.

In the example above, the entire Administratively-Scoped address range,(239.0.0.0/8) is being blocked from entering or leaving the router via interfaceSerial0. This is often done at the border of a network where it connects to theInternet so that potentially sensitive company Administratively-Scoped multicasttraffic can leave the network. (Nor can it enter the network from the outside.)




Notes: Notes:


Administrative BoundariesAdministrative Boundaries

Administratively-scoped multicast address ranges generally used in more than onelocation.

In the example above, the Administratively-Scoped address range, (239.128.0.0/16)is being used by both the LA campus and the NYC campus. Multicast trafficoriginated in these address ranges will remain within each respective campus andnot onto the WAN that exists between the two campuses. This is often sort of

configuration is often used so that each campus can source high-rate multicasts onthe local campus and not worry about it being accidentally “leaked” into the WANand causing congestion on the slower WAN links.

In addition to the 239.128.0.0/16 range, the entire company network has aAdministrative boundary for the 239.129.0.0/16 multicast range. This is so thatmulticasts in these ranges do not leak into the Internet.

The Admin. -Scoped address range (239..0.0/8) is similar to the 10.0.0.0 unicastaddress range in that it is reserved and is not assigned for use in the Internet.




Notes: Notes:


Types of Multicast ProtocolsTypes of Multicast Protocols

• Dense-mode — Uses “Push” Model — Traffic Flooded throughout network — Pruned back where it is unwanted — Flood & Prune behavior (typically every 3 minutes)

• Sparse-mode — Uses “Pull” Model — Traffic sent only to where it is requested — Explicit Join behavior

Dense-mode multicast protocols

Initially flood/broadcast multicast data to entire network, then prune back paths thatdon't have interested receivers

Sparse-mode multicast protocols

Assumes no receivers are interested unless they explicitly ask for it




Notes: Notes:


Dense ModeDense Mode

S1 S2 S3

R1 R2 R3 R4 R5 R6

In dense mode, every stream is delivered to every interface of every router unlesspruned off. If there is a small number of streams, one say, and this needs to reachevery user with an AS say the dense mode is a good choice. Typically this mightbe for a rare event, the board of directory’s announcement annually of profits, atelevised broadcast of the first person to step on the moon – or next time Mars. Ifstreams are not being viewed dense mode can waste a great deal of capacity andslow down normal operation.




Notes: Notes:


Sparse ModeSparse Mode

S1 S2 S3

R1 R2 R3 R4 R5 R6

RP1

RP2RP3

In sparse mode streams are delivered only when requested and only to interfaceswhere there are subscribers requesting service. We could support thousands ofstreams provided nobody viewed them! We can probably support small numbers ofhundreds of channels to an MSAN with perhaps one channel to each subscriber onan MSAN. If we assume that a TV stream requires 5 Mbit/s, an access speed ofbetter than 10 Mbit/s is needed to the subscriber and for a Gbit/s backhaul on anMSAN we could support as 100 channels at 50% load.




Notes: Notes:


Multicast ProtocolsMulticast Protocols

• Currently, there are 4 multicast routing protocols:

• DVMRP

— DVMRPv3 (Internet-draft) — DVMRPv1 (RFC1075) is obsolete and was never used.• MOSPF (RFC 1584) “Proposed Standard”

• PIM-DM (Internet-draft)

• PIM-SM (RFC 2362) “Proposed Standard”

DVMRPv1 is obsolete and was never used. DVMRPv2 is an old “Internet-Draft”and is the current implementation used through-out the Mbone. DVMRPv3 is thecurrent “Internet-Draft” although it has not been completely implemented by mostvendors.

MOSPF is currently at “Proposed Standard” status. However, most members of theIETF IDMR working group doubt that MOSPF will scale to any degree and are

therefore uncomfortable with declaring MOSPF as a standard for IP Multicasting.(Even the author of MOSPF, J. Moy, has been quoted in an RFC that, “more workneeds to be done to determine the scalability of MOSPF.”)

PIM-DM is in Internet Draft form and work continues to move into an RFC.

CBT is also in Internet Draft form and while it has been through three different andincompatible revisions, it is not enjoying significant usage nor is it a primary focusof the IETF IDMR working group.

PIM-SM moved to “Proposed Standard” in early 2000. Much of the effort in theIETF towards a working multicast protocol is focused on PIM -SM.




Notes: Notes:


Dense-Mode ProtocolsDense-Mode Protocols

• DVMRP - Distance Vector Multicast Routing Protocol

• MOSPF - Multicast OSPF

• PIM DM - Protocol Independent Multicasting (Dense Mode)




Notes: Notes:


PIM-SM (RFC 2362)PIM-SM (RFC 2362)

• Supports both source and shared trees — Assumes no hosts want multicast traffic unless they specifically ask for it

• Uses a Rendezvous Point (RP) — Senders and Receivers “rendezvous” at this point to learn of each others

existence.• Senders are “registered” with RP by their first-hop router.

• Receivers are “joined” to the Shared Tree (rooted at the RP) by their localDesignated Router (DR).

• Appropriate for… — Wide scale deployment for both densely and sparsely populated groups in

the enterprise — Optimal choice for all production networks regardless of size and

membership density.

Utilizes a rendezvous point (RP) to coordinate forwarding from source to receivers

Regardless of location/number of receivers, senders register with RP and send asingle copy of multicast data through it to registered receivers

Regardless of location/number of sources, group members register to receive dataand always receive it through the RP

Appropriate for Wide scale deployment for both densely and sparsely populatedgroups in the EnterpriseOptimal choice for all production networks regardless of size and membershipdensity.




Notes: Notes:


PIM-SM Shared Tree JoinsPIM-SM Shared Tree Joins

In this example, there is an active receiver (attached to leaf router at the bottom ofthe drawing) has joined multicast group “G”.

The leaf router knows the IP address of the Rendezvous Point (RP ) for group Gand when it sends a (*,G) Join for this group towards the RP.

This (*, G) Join travels hop-by-hop to the RP building a branch of the Shared Treethat extends from the RP to the last-hop router directly connected to the receiver.

At this point, group “G” traffic can flow down the Shared Tree to the receiver.




Notes: Notes:


PIM-SM Sender RegistrationPIM-SM Sender Registration

As soon as an active source for group G sends a packet the leaf router that isattached to this source is responsible for “Registering” this source with the RP andrequesting the RP to build a tree back to that router.

The source router encapsulates the multicast data from the source in a special PIMSM message called the Register message and unicasts that data to the RP.

When the RP receives the Register message it does two things. It de-encapsulates

the multicast data packet inside of the Register messageand forwards it down the Shared Tree. The RP also sends an (S,G) Join backtowards the source network S to create a branch of an (S, G) Shortest-Path Tree.This results in (S, G) state being created in all the router along the SPT, includingthe RP.




Notes: Notes:



As soon as the SPT is built from the Source router to the RP, multicast trafficbegins to flow natively from source S to the RP.

Once the RP begins receiving data natively (i.e. down the SPT) from source S itsends a ‘Register Stop’ to the source’s first hop router to inform it that it can stopsending the unicast Register messages.




Notes: Notes:



At this point, multicast traffic from the source is flowing down the SPT to the RPand from there, down the Shared Tree to the receiver.




Notes: Notes:


PIM-SM SPT SwitchoverPIM-SM SPT Switchover

PIM-SM has the capability for last-hop routers (i.e. routers with directly connectedmembers) to switch to the Shortest-Path Tree and bypass the RP if the traffic rate isabove a set threshold called the “SPT-Threshold”.

The default value of the SPT-Threshold” in Cisco routers is zero. This means thatthe default behavior for PIM-SM leaf routers attached to active receivers is toimmediately join the SPT to the source as soon as the first packet arrives via the

(*,G) shared tree.In the above example, the last-hop router (at the bottom of the drawing) sends an(S, G) Join message toward the source to join the SPT and bypass the RP.This (S,G) Join messages travels hop-by-hop to the first-hop router (i.e. the routerconnected directly to the source) thereby creating another branch of the SPT. Thisalso creates (S, G) state in all the routers along this branch of the SPT.

Finally, special (S, G)RP-bit Prune messages are sent up the Shared Tree to pruneoff this (S,G) traffic from the Shared Tree. If this were not done, (S, G) trafficwould continue flowing down the Shared Tree resulting in duplicate (S, G) packetsarriving at the receiver.




Notes: Notes:



At this point, (S, G) traffic is now flowing directly from the first -hop router to thelast-hop router and from there to the receiver.

Note: The RP will normally send (S, G) Prunes back toward the source to shutoffthe flow of now unnecessary (S, G) traffic to the RP IFF it has received an (S,G)RP-bit Prune on all interfaces on the Shared Tree. (This step has been omittedfrom the example above.)

As a result of this SPT-Switchover mechanism, PIM SM also supports theconstruction and use of SPT (S,G) trees but in a much more economical fashionthan PIM DM in terms of forwarding state.




Notes: Notes:



At this point, the RP no longer needs the flow of (S, G) traffic since all branches ofthe Shared Tree (in this case there is only one) have pruned off the flow of (S, G)traffic.

As a result, the RP will send (S, G) Prunes back toward the source to shutoff theflow of the now unnecessary (S, G) traffic to the RP

Note: This will occur IFF the RP has received an (S, G)RP-bit Prune on all

interfaces on the Shared Tree.




Notes: Notes:



As a result of the SPT-Switchover, (S, G) traffic is now only flowing from the first-hop router to the last-hop router and from there to the receiver.

Notice that traffic is no longer flowing to the RP.

As a result of this SPT-Switchover mechanism, it is clear that PIM SM alsosupports the construction and use of SPT (S,G) trees but in a much moreeconomical fashion than PIM DM in terms of forwarding state.




Notes: Notes:


PIM-SM Frequently Forgotten FactPIM-SM Frequently Forgotten Fact

“The default behavior of PIM-SM is that routers with directly connected members will join the Shortest Path Tree as soon as they detect a new multicast source.”

Unless configured otherwise, the default behaviour of Cisco routers running PIM-SM is for last-hop routers to immediately switch to the SPT for any new source.




Notes: Notes:


PIM-SM — EvaluationPIM-SM — Evaluation

• Effective for sparse or dense distribution of multicast receivers

• Advantages:

— Traffic only sent down “joined” branches — Can switch to optimal source-trees for high traffic sources dynamically — Unicast routing protocol-independent — Basis for inter-domain multicast routing

• When used with MBGP and MSDP

Evaluation: PIM Sparse-mode

Can be used for sparse of dense distribution of multicast receivers (no necessity toflood)

Advantages

Traffic sent only to registered receivers that have explicity joined the multicastgroup

RP can be switched to optimal shortest-path-tree when high-traffic sources areforwarding to a sparsely distributed receiver group

Inter-operates with DVMRP

Potential issues

Requires RP during initial setup of distribution tree (can switch to shortest-path treeonce RP is established and determined sub-optimal)




Notes: Notes:


“Virtually all production networks should be configured to run PIM in Sparse mode!”

ConclusionConclusion




Notes: Notes:




IGMP




Chapter Summary




Notes: Notes:




IGMP




Chapter Summary




Notes: Notes:


PIM Version 2 ConfigurationPIM Version 2 Configuration

• Protocol-Independent Multicast (PIM) Version 2 features

• Single active RP exists per multicast group with multiple backup RPs

—Multiple active RPs for the same group in PIM Version 1• A bootstrap router (BSR) provides a fault-tolerant with automated RP

discovery and distribution mechanism —Thus, routers dynamically learn the group-to-RP mappings

• Sparse mode and dense mode are properties of a group, as opposed toan interface. —Sparse-dense mode is recommended, as opposed to either sparse mode or

dense mode only.• PIM Join and Prune messages have more flexible encodings for multiple

address families

• A more flexible Hello packet format replaces the Query packet

• Register messages to an RP indicate source: border router or adesignated router






Notes: Notes:


BSRs and RPsBSRs and RPs

• Bootstrap routers (BSR) hold list of multicast groups that are reachable

• Different BSRs are elected on each side of PIM Domain Border

• To define a PIM Domain Border use ip pim border

• To prevent Auto-RP messages from entering the PIM domain use AccessControl List —access-list access-list-number {deny | permit} source [source-wildcard] —ip multicast boundary access-list-number

• Configure Candidate BSRs —ip pim bsr-candidate type number hash-mask-length [priority]

• Configure Candidate RPs —ip pim rp-candidate type number ttl group-list access-list-number




Notes: Notes:


Example BSR ConfigurationExample BSR Configuration

!ip multicast-routing!interface Ethernet0

ip address 171.69.62.35 255.255.255.240!interface Ethernet1

ip address 172.21.24.18 255.255.255.248ip pim sparse-dense-mode

!interface Ethernet2


!router ospf 1

network 172.21.24.8 0.0.0.7 area 1network 172.21.24.16 0.0.0.7 area 1

!ip pim bsr-candidate Ethernet2 30 10

ip pim rp-candidate Ethernet2 group-list 5access-list 5 permit 239.255.2.0 0.0.0.255




Notes: Notes:


Border Router Configuration ExampleBorder Router Configuration Example

!ip multicast-routing!!

interface Ethernet0ip address 171.69.62.35 255.255.255.240


ip address 172.21.24.18 255.255.255.248ip pim sparse-dense-modeip pim borderip multicast boundary 1



!access-list 1 deny 239.0.0.0 0.255.255.255access-list 1 deny 224.0.1.39 0.255.255.255

access-list 1 deny 224.0.1.40 0.255.255.255access-list 1 permit 224.0.0.0 15.255.255.255




Notes: Notes:


Using Auto-RPUsing Auto-RP

• Auto-RP is a feature that automates the distribution of group-to-RPmappings in a PIM network. Benefits: —Easy to use multiple RPs within a network to serve different group ranges. —Allows load splitting among different RPs and arrangement of RPs

according to the location of group participants —Avoids inconsistent, manual RP configurations that can cause connectivity

problems• Multiple RPs can be used to serve different group ranges or serve as hot

backups of each other

• To make Auto RP work, a router must be designated as an RP-mappingagent, which receives the RP-announcement messages from the RPs andarbitrates conflicts

• The RP-mapping agent then sends the consistent group-to-RP mappingsto all other routers

• All routers automatically discover which RP to use for the groups theysupport.




Notes: Notes:


Using Auto-RPUsing Auto-RP

• Sparse-mode environments need a default RP; sparse-dense-modeenvironments do not.

•

If you have sparse-dense mode configured everywhere, you do not needto choose a default RP

• Adding Auto-RP to a sparse-mode cloud requires a default RP

• Use that RP for the global groups, for example, 224.x.x.x

• To designate that a router is the RP, use the following command in globalconfiguration mode: —ip pim send-rp-announce type number scope ttl group-list access-list-

number• Example

ip pim send-rp-announce ethernet0 scope 16 group-list 1

access-list 1 permit 239.0.0.0 0.255.255.255




Notes: Notes:


Assign the RP Mapping AgentAssign the RP Mapping Agent

• The RP mapping agent is the router that sends the authoritative Discoverypackets telling other routers which group-to-RP mapping to use. Such arole is necessary in the event of conflicts (such as overlapping group-to-RP ranges)

• Find a router whose connectivity is not likely to be interrupted and assignit the role of RP-mapping agent

• All routers within ttl number of hops from the source router receive theAuto-RP Discovery messages

• To assign the role of RP mapping agent in that router, use the followingcommand in global configuration mode: —ip pim send-rp-discovery scope ttl

• Verify the Group-to-RP Mapping using —show ip pim rp [group-name | group-address] [mapping]




Notes: Notes:


ip pim accept-rp default RP address 1access-list 1 permit 224.0.1.39access-list 1 permit 224.0.1.40

Prevent Join Messages to False RPsPrevent Join Messages to False RPs

• The addresses from which Auto-RP announcements are accepted can belimited

• To filter incoming RP announcement messages, use the followingcommand in global configuration mode: —ip pim rp-announce-filter rp-list access-list-number group-list access-list-

number




Notes: Notes:


Monitoring the Multicast Routing TableMonitoring the Multicast Routing Table

R99#show ip mrouteIP Multicast Routing TableFlags: D - Dense, S - Sparse, C - Connected, L - Local, P - Pruned

R - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPT

M - MSDP created entry, X - Proxy Join Timer RunningA - Candidate for MSDP Advertisement

Outgoing interface flags: H - Hardware switchedTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode

(*, 239.255.255.250), 00:23:39/00:02:32, RP 0.0.0.0, flags: DJCIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:

Ethernet0, Forward/Sparse, 00:23:39/00:02:25Ethernet1, Forward/Sparse, 00:14:47/00:02:32

(*, 224.0.1.39), 00:21:35/00:00:00, RP 0.0.0.0, flags: DJCLIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:

Loopback0, Forward/Sparse, 00:21:35/00:02:23Ethernet0, Forward/Sparse, 00:21:35/00:02:31Ethernet1, Forward/Sparse, 00:14:47/00:02:29

This shows how to display the multicast routing table. Generally this will be quirelong and stretch over many pages.




Notes: Notes:


Monitoring the Multicast Routing TableMonitoring the Multicast Routing Table

(*, 224.0.1.40), 00:24:03/00:00:00, RP 0.0.0.0, flags: DJCLIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:

Ethernet0, Forward/Sparse, 00:24:04/00:02:23

(*, 225.1.2.3), 00:01:36/00:02:59, RP 0.0.0.0, flags: DJCIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:


(192.168.0.31, 225.1.2.3), 00:01:36/00:02:59, flags: CTIncoming interface: Ethernet0, RPF nbr 0.0.0.0Outgoing interface list:


Here 192.168.0.31 is the source for the stream 225.1.2.3 which is arriving oninterface Ethernet 0.




Notes: Notes:


Monitoring IGMP GroupsMonitoring IGMP Groups

R99#show ip igmp groupsIGMP Connected Group MembershipGroup Address Interface Uptime Expires Last Reporter

239.255.255.250 Ethernet1 00:13:27 00:02:46 192.168.1.250239.255.255.250 Ethernet0 00:22:19 00:02:44 192.168.0.18224.0.1.39 Loopback0 00:20:15 never 10.1.1.1224.0.1.39 Ethernet1 00:20:15 never 192.168.1.99224.0.1.39 Ethernet0 00:20:15 never 192.168.0.99224.0.1.40 Ethernet0 00:22:43 never 192.168.0.99225.1.2.3 Ethernet1 00:00:07 00:02:53 192.168.1.250R99#

“Show ip igmp groups” displays the groups that the router is a member of.




Notes: Notes:


SenderReceiver

Ae0/0 e0/1 e3/1 e3/2

R75a R72a

RPF FailureRPF Failure

• Multicast packets are coming into e0/0 of 75a from a server whose ipaddress is 1.1.1.1 and sending to group 224.1.1.1. This is know as an (S,G)or (1.1.1.1, 224.1.1.1)

• Problem: Hosts who are directly connected to R75a are getting themulticast feed. But hosts (HostA), who are directly connected to R72a, arenot getting the stream




Notes: Notes:


Look at what is going on in R75aLook at what is going on in R75a

ip22-R75a#sh ip mroute 224.1.1.1IP Multicast Routing TableFlags: D - Dense, S - Sparse, C - Connected, L - Local, P - PrunedR - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPTM - MSDP created entry, X - Proxy Join Timer RunningA - Advertised via MSDPTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode

(*, 224.1.1.1), 00:01:23/00:02:59, RP 0.0.0.0, flags: DIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:Ethernet0/1, Forward/Sparse-Dense, 00:01:23/00:00:00

(1.1.1.1, 224.1.1.1), 00:01:23/00:03:00, flags: TAIncoming interface: Ethernet0/0, RPF nbr 0.0.0.0Outgoing interface list:Ethernet0/1, Forward/Sparse-Dense, 00:01:23/00:00:00

This is telling us that the multicast packets are being sourced from a server whose

address is 1.1.1.1 which is sending to a multicast group of 224.1.1.1

Since it's dense mode we can basically ignore the *,G entry and focus on the S,Gentry. It's telling us that the multicast packets are being sourced from a serverwhose address is 1.1.1.1 which is sending to a multicast group of 224.1.1.1. Thepackets are coming in the ethernet0/0 interface and being forwarded out theethernet 0/1 interface. This is perfect. So, since we know that packets are beingforwarded out ethernet 0/1 to the LAN which R72a is connected, let's take a look atwhat R72a is doing with the packets.




Notes: Notes:


Look at what is going on in R72aLook at what is going on in R72a


(*, 224.1.1.1), 00:05:36/00:02:19, RP 0.0.0.0, flags: DJCIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:Ethernet3/1, Forward/Sparse-Dense, 00:05:36/00:00:00Ethernet3/2, Forward/Sparse-Dense, 00:05:37/00:00:00

The necessary S,G (1.1.1.1, 224.1.1.1) is not even listed.

Let's look to see if it's showing the upstream router (75a) as a pim neighbor:

The necessary S,G (1.1.1.1, 224.1.1.1) is not even listed. It's definitely notforwarding the packets. So what's going on. At least we found the trouble spot, nowwe'll have to probe deeper on this router.

Let's look to see if it's showing the upstream router (R75a) as a pim neighbor:




Notes: Notes:


Examining the PIM NeighborsExamining the PIM Neighbors

ip22-R72a#sh ip pim neighPIM Neighbor TableNeighbor Address Interface Uptime Expires Ver Mode2.1.1.1 Ethernet3/1 2d00h 00:01:15 v2

ip22-R72a#sh ip rpf 1.1.1.1RPF information for ? (1.1.1.1)RPF interface: Ethernet3/3RPF neighbor: ? (4.1.1.2)RPF route/mask: 1.1.1.1/32RPF type: unicast (static)RPF recursion count: 1Doing distance-preferred lookups across tables

SenderReceiver

Ae0/0 e0/1 e3/1 e3/2

R75a R72a

It is showing the RPFInterface as e3/3 but

we expect it to be e3/1

It's showing the rpf interface being e3/3 but it should be e 3/1 as the incominginterface. Let's further confirm with some debug, although we'll need to be carefulwith this as it will be busy:




Notes: Notes:


Confirm The Problem With Debug and SolveConfirm The Problem With Debug and Solve

ip22-R72a#debug ip mpacket*Jan 14 09:45:32.972: IP: s=1.1.1.1 (Ethernet3/1) d=224.2.127.254 len 60, not RPF interface*Jan 14 09:45:33.020: IP: s=1.1.1.1 (Ethernet3/1) d=224.2.127.254 len 60, not RPF interface*Jan 14 09:45:33.072: IP: s=1.1.1.1 (Ethernet3/1) d=224.2.127.254 len 60, not RPF interface*Jan 14 09:45:33.120: IP: s=1.1.1.1 (Ethernet3/1) d=224.2.127.254 len 60, not RPF interface

Add a static rout to solve the problem

ip22-R72a(config)#ip mroute 1.1.1.1 255.255.255.255 2.1.1.1

Sure enough, The packets are coming in on ethernet3/1 as we want but they arebeing dropped because that's not the interface the unicast routing table is using forrpf check. We can either change the unicast routing to satisfy this requirement orwe can add a static mroute to force multicast to RPF out a particular interfaceregardless of what the unicast routing table states. We'll add a static mroute:

ip22-72a(config)#ip mroute 1.1.1.1 255.255.255.255 2.1.1.1

This static multicast route states that to get to the address 1.1.1.1, for rpf, use2.1.1.1 as the next hop to get there, which is out e3/1.




Notes: Notes:


Confirm The SolutionConfirm The Solution

ip22-R72a#sh ip rpf 1.1.1.1RPF information for ? (1.1.1.1)RPF interface: Ethernet3/1RPF neighbor: ? (2.1.1.1)

RPF route/mask: 1.1.1.1/32RPF type: static mrouteRPF recursion count: 0Doing distance-preferred lookups across tables




Notes: Notes:


Confirm The SolutionConfirm The Solution



(1.1.1.1, 224.1.1.1), 00:00:48/00:02:59, flags: CTAIncoming interface: Ethernet3/1, RPF nbr 2.1.1.1, MrouteOutgoing interface list:Ethernet3/2, Forward/Sparse-Dense, 00:00:48/00:00:00

ip22-R72a#debug ip mpacket*Jan 14 10:18:29.951: IP: s=1.1.1.1 (Ethernet3/1) d=224.1.1.1 (Ethernet3/2) len 60, mforward*Jan 14 10:18:29.999: IP: s=1.1.1.1 (Ethernet3/1) d=224.1.1.1 (Ethernet3/2) len 60, mforward*Jan 14 10:18:30.051: IP: s=1.1.1.1 (Ethernet3/1) d=224.1.1.1 (Ethernet3/2) len 60, mforward

The real confirmation of the solution is that host A now gets packets.




Notes: Notes:


Time To LiveTime To Live

Sender

S

ReceiverR

e0/0 e0/1 e1/1 e1/2

RouterA RouterB

1.1.1.1 1.1.1.2 2.1.1.1 2.1.1.2 3.1.1.1 3.1.1.2

Problem: RouterA is not forwarding packets from source(S) to RouterB'sdirectly connected receiver(R)

sending to:224.1.1.1

Troubleshooting:

Let's first look at 'sh ip mroute' on ROUTERA to see the multicast routing state:




Notes: Notes:


Look at RouterA MROUTELook at RouterA MROUTE

ROUTERA#sh ip mrouteIP Multicast Routing TableFlags: D - Dense, S - Sparse, C - Connected, L - Local, P - PrunedR - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPT

M - MSDP created entry, X - Proxy Join Timer RunningA - Advertised via MSDPTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode

(*, 224.0.1.40), 00:00:01/00:00:00, RP 0.0.0.0, flags: DJCLIncoming interface: Null, RPF nbr 0.0.0.0Outgoing interface list:Ethernet0/1, Forward/Sparse-Dense, 00:00:01/00:00:00


The Source 1.1.1.1 is not being recognised

We can ignore the 224.0.1.40 since all routers will join this Auto-RP Discoverygroup. But we don't have a source listed for 224.1.1.1. In addition to "*, 224.1.1.1"we should be seeing "1.1.1.1, 224.1.1.1". We don't recognize the source as beingvalid for some reason.

Let's see if it's an RPF issue:




Notes: Notes:


Check the RPFCheck the RPF

ROUTERA#sh ip rpf 1.1.1.1RPF information for ? (1.1.1.1)

RPF interface: Ethernet0/0RPF neighbor: ? (0.0.0.0) - directly connectedRPF route/mask: 1.1.1.0/24RPF type: unicast (connected)RPF recursion count: 0Doing distance-preferred lookups across tables

The RPF looks right as it is correctly pointing to e0/0

The rpf check is correctly pointing out e0/0 to get to the source's ip address.

Let's see if pim is configured on the interfaces:




Notes: Notes:


Check the Interfaces and the TrafficCheck the Interfaces and the Traffic

ROUTERA#sh ip pim int

Address Interface Version/Mode Nbr Query DR

Count Intvl1.1.1.2 Ethernet0/0 v2/Sparse-Dense 0 30 1.1.1.22.1.1.1 Ethernet0/1 v2/Sparse-Dense 2 30 2.1.1.2

ROUTERA#sh ip mroute activeActive IP Multicast Sources - sending >= 4 kbps

The router does not see any traffic as no sources listed. You could check this with:

ROUTERA#debug ip mpacket

Perhaps the Receiver is not sending any igmp reports (joins) for group 224.1.1.1:

Check this with Debug or igmp group.




Notes: Notes:


IGMP GroupIGMP Group

ROUTERB#sh ip igmp groupIGMP Connected Group MembershipGroup Address Interface Uptime Expires Last Reporter224.0.1.40 Ethernet1/1 00:34:34 never 2.1.1.2224.1.1.1 Ethernet1/2 00:30:02 00:02:45 3.1.1.2

3.1.1.2 has joined the group so it can only be aTTL issue if traffic is actually being sent

224.1.1.1 has been joined on e1/2 so that's fine. Perhaps ROUTERB is not sendingPIM JOINS to ROUTERA informing ROUTERA that it needs to forward thetraffic:

Actually, ROUTERB is not going to send pim joins to ROUTERA. Dense Mode isa flood and prune protocol, so there are no joins, but there are grafts. But sinceROUTERB hasn't received any flood from RouterA, it doesn't know where to send

a graft.Perhaps it's a TTL (Time to Live) issue:




Notes: Notes:


Show IP TrafficShow IP Traffic

ROUTERA#sh ip trafficIP statistics:Rcvd: 248756 total, 1185 local destination

0 format errors, 0 checksum errors, 63744 bad hop count0 unknown protocol, 0 not a gateway0 security failures, 0 bad options, 0 with options

Indication of the problem is the 63744 bad hop count

Repeating the show will indicate that this is increasing

To fix this the source must increase the TTL of the multicast traffic

63744 bad hop counts, and each time I type this command, the bad hop countsincrease. This is the TTL incrementing. We have found the problem. To solve theissue we need to increase the TTL on our Sender application.




Notes: Notes:


The Fixed TTL - RouterAThe Fixed TTL - RouterA

ROUTERA#sh ip mrouteIP Multicast Routing TableFlags: D - Dense, S - Sparse, C - Connected, L - Local, P - PrunedR - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPTM - MSDP created entry, X - Proxy Join Timer RunningA - Advertised via MSDPTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode



(1.1.1.1, 224.1.1.1), 00:19:24/00:02:59, flags: TAIncoming interface: Ethernet0/0, RPF nbr 0.0.0.0

Outgoing interface list:Ethernet0/1, Forward/Sparse-Dense, 00:19:24/00:00:00




Notes: Notes:


The Fixed TTL - RouterAThe Fixed TTL - RouterAROUTERB#sh ip mrouteIP Multicast Routing TableFlags: D - Dense, S - Sparse, C - Connected, L - Local, P - PrunedR - RP-bit set, F - Register flag, T - SPT-bit set, J - Join SPTM - MSDP created entry, X - Proxy Join Timer Running

A - Advertised via MSDPTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode



(1.1.1.1, 224.1.1.1), 00:17:46/00:02:59, flags: CTAIncoming interface: Ethernet1/1, RPF nbr 2.1.1.1Outgoing interface list:Ethernet1/2, Forward/Sparse-Dense, 00:17:46/00:00:00




Notes: Notes:


TTL ThreasholdTTL Threashold

SenderReceiver

Ae0/0 e0/1

75a

1.1.1.1 1.1.1.2 2.1.1.1 2.1.1.2

Problem:

The sender is sending to 224.1.1.1 but receiver is not gettingthe multicast packets from the source

Troubleshooting:

There may be several routers between the source and the 75a router, but let's firsttake a look at the 75a since it's directly connected to the receiver:






Notes: Notes:


Turn On Debug For The ServiceTurn On Debug For The Service

ip22-75a#conf tEnter configuration commands, one per line. End with CNTL/Z.ip22-75a(config)#access-list 101 permit udp host 1.1.1.1 host 224.1.1.1ip22-75a(config)#end

ip22-75a#debug ip mpacket 101IP multicast packets debugging is on for access list 101ip22-75a#*Jan 17 09:04:08.714: IP: s=1.1.1.1 (Ethernet0/0) d=224.1.1.1 len 60, threshold denied*Jan 17 09:04:08.762: IP: s=1.1.1.1 (Ethernet0/0) d=224.1.1.1 len 60, threshold denied*Jan 17 09:04:08.814: IP: s=1.1.1.1 (Ethernet0/0) d=224.1.1.1 len 60, threshold denied

Limit traffic to just the service required

Traffic is limited by the TTL threshold

The router is not forwarding the packets because of a threshold being denied. Webetter look in the config and find out what this is all about.




Notes: Notes:


Examine the Interface ConfigurationExamine the Interface Configuration

interface Ethernet0/1ip address 2.1.1.1 255.255.255.0

ip pim sparse-dense-modeip multicast ttl-threshold 15

This interface will only forward traffic with a TTL greater than 15

Often people think the threshold prevent traffic with a greater value being sent

The reverse is true.

Reduce the threshold or delete the line to fix the problem .

The customer configured a ttl threshold of 15, thinking that anything greater than attl of 15 will not be sent. Actually, just the opposite is configured. The applicationis being sent with a ttl of 15. By the time it gets to the 75a router, the multicastpackets have a ttl less than 15. We better look at Multicast-Commands, and seewhat is goingon:

[no] ip multicast ttl-threshold <ttl-value>Configures a packet TTL threshold for traffic going out the interface.Any multicast packets with a TTL less than the threshold are notforwarded out the interface. The default value is 0 which means allmulticast packets are forwarded out interface.

So any packets with a ttl LESS than the threshold are not forwarded. This commandis usually used to provide a border to keep internal multicast traffic from driftingout of the intranet.

Resolution: Either remove this command (and use 'ip pim border' instead) or lowerthe ttl threshold so that the traffic can pass.




Notes: Notes:




IGMP




Chapter Summary




Notes: Notes:

Managing Devices with SNMPManaging Devices with SNMPManaging Devices with SNMP

Chapter 8Chapter 8




Notes: Notes:



In this chapter, we will

• Examine the way in which Network Management stations communicate

with managed devices• Identify the structure of SNMP for management

• Detail the component parts of SNMP management




Notes: Notes:


Managing Devices with SNMPManaging Devices with SNMP

Network Management Concepts

Management Information Base (MIB)

SNMP

Chapter Summary




Notes: Notes:


Modern NetworksModern Networks

Server

Router

Router

server

Router

• Modern networks are complex and diverse

Networks tend to evolve over time. This means that there can be many differentgenerations of technology present at the same time. A management system that cancope with only the latest technologies is therefore not the best solution in practice.

Management systems must be capable of coping with multi-vendor and multi-technology infrastructures. Also service Level Management demands that allcomponents and servers be manageable in order to be able to deliver the required

level of quality and performance.






Notes: Notes:


Location of Network ManagementLocation of Network Management

• Network Management is a Layer 7 function — Devices that do not have a full protocol stack must be enhanced

Physical

Data Link

Network

Transport

Session

Presentation

ApplicationM

Physical

Data Link

Network

Transport

Session

Presentation

Application M

Physical

Data Link

Network

Transport

Session

Presentation

Application M

We must ask ourselves how can layer 7 obtain information about lower layers, layer1 say. It could ask layer 6 to ask layer 5 to ask layer 4 to ask layer 3 to ask layer 2to ask layer 1 “How are you layer 1”. Layer 1 could then reply to layer 2 thatreplies to layer 3 that replies to layer 4 that replies to layer 5 that replies to layer 6that replies to layer 7 “I am fine!”. But if layer 7 is to monitor all layers for faultsand try to correct them, could it trust the reply. A fault or failure in a middle layercould prevent layer 1 replies, or worse, change them.

For layer 7 to reliably monitor and control it must obtain its information, as far as itcan, from sources independent of the layers between. It therefore mustcommunicate, at least for part of the time, with lower layer management entitiesusing services independent of the main communications pathways.

Also how can a device manage intermediate systems such as routers?




Notes: Notes:


Location of Network ManagementLocation of Network Management

• Network Management is a Layer 7 function — Devices that do not have a full protocol stack must be enhanced

Physical

Data Link

Network

Transport

Session

Presentation

ApplicationM

Physical

Data Link

Network

Transport

Session

Presentation

Application M

Physical

Data Link

Network

Transport

Session

Presentation

Application M

Additional services must be added to the router to provide the layers that aremissing in order for the layer 7 management functions at least to communicate. Inthe case of SNMP this implies the addition of UDP and SNMP itself. A normalrouter will require UDP in any case for it to communicate with other applicationservices including BOOTP and some routing protocols needed to undertake thenormal router functions. However the instrumentation, usually special software,that provides the management services with management information and allowschanges to be made in control tables within the router must be added.These are the elements found in the Agent services.




Notes: Notes:


Evolution of SNMPEvolution of SNMP

HEMS SGMP

CMOT

SNMP SNMPv1Recommendedstatus

More than30vendorsdemonstrateSNMPproductsatInterop

SMIMIB-1“standardprotocols”

MIB-2

1986 1987 1989 1990 1991

RMONMIB SNMPv2

1992 1993

SNMPv3

1998

Fundamentalflaws foundand SNMPv2largelyabandoned

• Official internet standard is still SNMPv1 — Security limitations caused development of v2 and v3 — SNMP v3 has upen ended security model but complex to use — Management needed when network fails to function

The evolution of SNMP started in the middle 1980s when the Internet was beginning to becomeestablished and routing table errors could cause parts of the network to become unreachable. High-level Entity Management System (HEMS) was a monolithic process run typically within the UNIXsystems that formed the main notes of the network, what are today called routers. While this provedto be useful it was not easily portable across platforms so Simple Gateway Monitoring Process(SGMP) was built which defined the protocol interacts between systems independently of theunderlying platform. This was however still large and relatively complex, too complex toimplement within a simple agent in a low level device.

At about the same time, the end of the 1980s, OSI was under extensive standardization development.The OSI model had been agreed in 1982 and the form of the major layer 3 to 5 protocols were fixed.However the management protocols were still incomplete and extensive standardization workremained. The Internet community therefore decided that as one day OSI would replace TCP/IP, itwould be silly to develop a different management approach only later to need to replace it. Better toassist in the OSI standardization and adopt compatible techniques, at least for management dataaccess.

By 1989 it was clear that it would be many years before the OSI Common Management InformationProtocol (CMIP) would be complete, and that the direction being taken would yield a large complexprotocol. It was therefore decided to build an interim simple protocol able to access the same datastructures. This was SNMP, or what is now known as SNMPv1.




Notes: Notes:


Managementapplication

UDP/IP

Network

DataAgent

UDP/IP

SNMP

SNMP StructureSNMP Structure

• Management application is user provided

• An Agent runs on every managed platform

• Management communications is provided by SNMP

A management application is typically a Windows Icon Menu Pointer GraphicalUser Interface (WIMP-GUI) Application. It sits on top of UDP and IP sending andreceiving SNMP exchanges with many agents across the network.

Every device that is to be managed must have an agent to communicate with themanagement application in order to implement the management service. Theexchange of data depends upon both manager and agent understanding the details

and meaning of the Management variable in use. In practice these sit within theagent device, usually as values held in random access memory. The managementapplication on the other hand must hold details of which variables are understoodby which devices in the network so must hold more details on backing storage.




Notes: Notes:


SNMP within Internet Protocol SuiteSNMP within Internet Protocol Suite

• SNMP runs over UDP

SNMP

User Datagram Protocol (UDP)

ARP

IEEE 802 LANX.25PacketradioEtc.

Transmission Control Protocol (TCP)

Internet Protocol (IP)

SMTP

SimpleMail

TransferProtocol

TELNETvirtual

terminal

FTP

FileTransferProtocol

ICMP

OSILayer

5,6 ,7

4

3

2,1

IPS Layer

Application

End-to-end

Internetwork

Physical

network

SNMP is one of more than a hundred protocols within the Internet Protocol Suite(IPS). Often this is called TCP/IP. IP is the layer 3 protocol that undertakesrouting and the delivery of data to the correct destination. IP does not guaranteereliable delivery however, it is a best efforts or datagram protocol. In most user-base applications reliable transmission is provided above IP by using TransmissionControl protocol (TCP) which sequences data to keep it in order and retransmitsdata that is corrupted or lost.

SNMP uses User Datagram Protocol which is a much simpler protocol than TCP. Itcan be configured to detect transmission errors and discard transfers that arecorrupted by using a checksum similar to TCP. But unlike TCP it does notsequence data nor undertake retransmissions. In the event of an error in any part ofthe layers 1, 2 or 3 data may be lost. UDP is often termed unreliable.




Notes: Notes:


Why over UDP?Why over UDP?

• UDP is unreliable — don’t we want reliable management?

• . . . Yes of course but

• When do you need network management most?

• . . . and will the network deliver data reliably then?

• If we place SNMP above TCP it cannot see the flaws in the network — We need to see the networkas it is in order to manage it — TCP would correct errors or crash! — The network would appear perfect or broken beyond repair

• If we are not managing the network then we can use TCP — Managing end system components can be done over TCP

At first it seems strange that we should select UDP rather than TCP to carry SNMP managementdata. After all surely we want reliable management. However when there is a fault in the networkthere is little chance that all the data will arrive correctly and a protocol like TCP might neversucceed in delivering a correct version of every piece of data. It might then just abandon thetransmission as “broken”. Under failing conditions the management application would find itdifficult to discover exactly what the failure is if no data arrives. Some faults will be completelyhidden by TCP. How for example can you detect that a network is delivering data packets out oforder if TCP reorders them?

Indeed with TCP sitting below SNMP all networks seem either perfect or so broken that nocommunication is possible. The only time that TCP would offer better service would be if themanagement communication with the device being managed was over a different network to thenetwork being controlled. This is known as “out-of-band” management. In reality this is often thecase with telecommunications. It does however pose another question: how do you manage themanagement network? After all if management is important to the mission then clearly themanagement network must be highly reliable and management of this will be vital. This “Meta-management” as it is known must itself then run over yet another network or must run over somekind of datagram protocol.




Notes: Notes:


Network-managementapplication

Managementworkstation

Network-management

database Agent

SNMPprotocol

Managementinformationbase (MIB)

Management FrameworkManagement Framework

• Data originally defined in “two standards” — Structure of Management Information (SMI) (RFC 1155) — Management information base (MIB) (RFC 1156 replaced by 1213) — Extensive extension for different technologies, 100+s standard MIBs

• Data access originally defined in SNMP protocol — SNMP defines simple protocol for transferring data (RFC 1157)

• These have been modified/extended by subsequent RFCs

The framework for management data was originally defined within two standards:-

The structure of Management Information (RFC 1155) and the ManagementInformation Base (RFC 1156 which was replaced by RFC 1213). While laterstandards have added considerably to the protocol and the SMI the documents aremore difficult to read and understand. When learning about SNMP it is generallybest to start from SMIv1 and then extend this knowledge to the superset formed by

SMIv2. We shall do this here. Also the development of SNMPv3 has providedsubstantially greater security features but the documents describing the protocol aremuch easier to read and understand when the concepts of SNMPv1 have beenmastered.

We shall first concentrate on understanding SMIv1 and the MIB defined by RFC1213 which is the minimum subset supported by all real devices. Later we willexamine some of the extensions in SNMPv3 and SMIv2.




Notes: Notes:





SNMP

Chapter Summary




Notes: Notes:


Paul’s machineis working fine!

PeterPaul

Mary

GeorgeRingo

Paul

What didRingocall thatparameter?

SNMP MIBs use a managed NamespaceSNMP MIBs use a managed Namespace

• Need unambiguous names

The MIB forms a managed namespace. We need this namespace to be infinitelyexpandable and allow different parts of its coverage to be controlled by manydifferent authorities.

While it is not a simple problem to solve, the MIB is no different to the allocationof names for domains or email addresses. In much the same way we use a treestructure so that names can be made unambiguous.




Notes: Notes:


ISO - CCITT Managed NamespaceISO - CCITT Managed Namespace

1

0 1 2 3

0 2

0 1 2 3 4 5 6

1

1 2 3 4

1

ccitt iso joint-ccitt-iso

org (identified organizations)

dod

internet

mgmt

mib-2

1.3.6.1 internet

1.3.6.1.1 directory

1.3.6.1.2 mgmt

1.3.6.1.3 experimental

1.3.6.1.4 private

1.3.6.1.5 security

1.3.6.1.6 SNMPv2

1.3.6.1.7 mail

1.3.6.1.8 features

5 8

private

Added after RFC 1213

standards

The tree starts from an unnamed root node and is controlled by two standardsorganizations - ISO and the ITU-T (formerly known as the CCITT). Nodes withinthe tree are given both names and numbers and can be referred to using either. Byconvention the names are written using lower case and are case sensitive. The earlyMIBs such as RFC 1213 tend to include some cryptically coded names. “mgmt”for management and “org” for “identified organizations”. More recently addedMIB extensions have taken a different direction. Multi-word names are used butare encoded starting in lower case and then concatenating words together withoutspaces but capitalizing the first letter of each new word.At the highest level thereare nodes controlled by CCITT, by ISO and jointly controlled by both. ISO havedelegated responsibility for part of its number space to other organizations andthese are grouped under one node, node 3 “org”. All internet variables thereforestart 1.3 or iso.org. The 6th organization is the US Department of Defense (DOD)so all nodes are under iso.org.dod (1.3.6). The DOD have delegated everything tothe Internet community so all variables start iso.org.dod.internet (1.3.6.1).




Notes: Notes:


Private ExtensionsPrivate Extensions

Internet

Directory Management Experimental Private

System

Interfaces

AT IP ICMP

TCP

UDP

EGP

Transmission

SNMP

PPP RS-232

Enterprises

IBM DEC Cisco Motorola

MIB

1

1 2 3 4

18 20

1

1

2

34

56

7

11

10

8

1

1612 36 9

Motorola

11911




Notes: Notes:


Branches Under Internet (1.3.6.1) in RFC 1213Branches Under Internet (1.3.6.1) in RFC 1213

• Directory (1) — Reserved for use with the OSI directory (X.500) and White Pages — Not used by SNMP

• Management (2) — Objects defined in IAB-approved documents — Currently has only one entry: the SNMP MIB-2

• Experimental (3) — Objects used in Internet experiments — “Successes” migrate to mgmt(2) — Use of this branch is deprecated

• Private (4) — Objects defined unilaterally — Currently has only one entry: enterprise(1)

— Allows manufacturers to support capabilities not in mgmt(2)

Under the internet node (1.3.6.1) RFC 1213 placed four branches. Directory node1, which is reserved for OSI X.500 management. So far this has not been widelyused. Management node 2, known as “mgmt” in RFC1213 which hold MIB-2 andall its standard extensions. Experimental node 3, which can be used duringdevelopment in order to test the functioning of new features and variables withoutimpacting operational services. Private node 4 which has a single sub node“enterprise” under which each organization that wishes to register can obtain itsown node. There are now more than 12,000 registered enterprises each of whichcould have their own MIB variables.




Notes: Notes:


Later ExtensionsLater Extensions

• Security (5) — Development of objects still continues for confidentiality and integrity

services and mechanisims — Includes: Kerberose, MD5-DES-CBS, IPSEC and other integrity

mechanisms• SNMPv2 (6)

– Mechanisms developed to manage the “party MIB” used for SNMPv2security

– No longer used• Mail (7)

— RFC 1495 mail management• Features (8)

— Media-feature-tags RFC 2506 — Includes feasures such as pixels, DPI, color, image-coding etc

Considerable development work is currently being undertaken to develop MIBs thatcan be used to manage security features. Currently few of these have reachedcompleted RFCs but are being placed under node 5.

Node 6 defines SNMPv2 features. However SNMPv2 has been obsoleted by workon SNMPv3. Parts of this group can still be used to control proxy management andcommunity naming.

Node 7 is for Mail Management and RFC 1495 defines variables here.Node 8 is used for media management and details of RFCs that define variable herecan be found at http://www.iana.org/assignments/media-feature-tags .




Notes: Notes:


Management Information BaseManagement Information Base

• Data is held in a database called the MIB

• Database is split into a number of management groups (areas)

• MIB-1 holds data for eight managed areas - RFC 1156 — Total of 114 object definitions

• MIB-2 definition RFC 1213 — Added two new categories — Improved support for multi-protocol devices — About 50 percent larger (171 object definitions)

Now let us turn our attention to the standard subset of variable all devices that useIP should support, RFC 1213 MIB-2. All of the variable in MIB-1 appear in MIB-2with some additions and improvements added.




Notes: Notes:


MIB-2 Variable GroupsMIB-2 Variable Groups

• Group Description Number of MIB variables — system The managed node 7 — interfaces Network attachments 23 — at IP address translation 3 — ip Internet Protocol 38 — icmp Internet Control Message Protocol 26 — tcp Transmission Control Protocol 19 — udp User Datagram Protocol 7 — egp Exterior Gateway Protocol 18 — transmission Physical interface - — snmp Simple Network Management Protocol 30 — Total: 171

Details of these groups are given in Appendix B.The system group is used to define the names and contacts for the managed device. It also containssysUpTime which identifies the time since the device last rebooted in 0.01 sec units.The interfaces group includes variables that refer to layer 1 and layer 2 functions of deviceinterfaces.The at group holds the ARP table in MIB-1but this was moved to the ip group to a table called theipNetToMediaTable in MIB-2.The ip group includes all the details about ip addresses, routing and statistics information.The icmp group holds input and output counters for icmp messages which carry ping (echo),redirects, host unreachable and time exceeded messages among others.The tcp group holds counters for tcp transfers as well as a table giving details of current tcpconnections open.The udp group holds counters of input and output udp transfers.The egp group holds details of exterior gateways status; this gives information about the directconnection to the Internet.The transmission group does not contain variables in RFC 1213 but is used to hold extensions inother RFCs that detail technology specific layer 1 information.




Notes: Notes:


MIB-1 and MIB-2 GroupsMIB-1 and MIB-2 Groups

1.3.6.1.2.

system

interfaces

at

ip

icmp

tcp

ospf bridge . . . etc. . .

2

1 3

4

5

6

14 17 etc.

udp

7

transmission

10

rmon

16

egp

8

snmp

11

1 MIB-2

MIB-1 RFC 1156

MIB-2 RFC-1213

Notice that MIB-2 adds the transmission and snmp groups as nodes 10 and 11.Node 9 was reserved for CMOT but has never been implemented in any RFC.Extensions for routing protocols, bridges, host services and the like normally extendbeyond node 11.




Notes: Notes:


Structure of Management Information (SMI)Structure of Management Information (SMI)

• Defined by RFC 1155 (Structure of Management Information) — Expanded by RFC 1212 (Concise MIB Definitions) for MIB-2 — Further expanded by RFC 1451 (SMI for SNMPv2) — RFC 2578 SMIv2

• Defines all properties of the MIB object — Syntax for the object type — Access allowed for the object (read only, read-write, write only, or not

accessible) — Status of the object (mandatory, optional, deprecated, or obsolete) — Textual description, cross-references, and default value are optional — Indexes used for lookup if a table-oriented object

• MIB data defined using ASN.1

Originally it was defined in RFC1155 and later expanded in RFC 1451 for SMIv2.The latest version of SMIv2 is documented in RFC 2578. There are fundamentaldifferences between SMIv1 and SMIv2. Many products still have their MIBsdocumented in SMIv1 and it would be possible to document most MIBs usingSMIv1 if required. SMIv2 adds some macro definitions as well as mechanisms forbetter documentation of notifications (Traps) and other services. Most recentlydeveloped MIBs use SMIv2 but not all management products yet support it.








Notes: Notes:


Defining DataDefining Data

• Application-defined data types for SNMPv1 — IPAddress : four octets in “dotted” order — Counter : Can be incremented but not decremented, ranges from

0 to 232 – 1 and wrapping around to 0 — Gauge : Can be increased or decreased, ranges from 0 to 232 – 1 and does

not wrap — TimeTicks : Non-negative integer representing the time in hundredths of a

second since some “epoch”• SMIv2 renames Counter and Gauge renamed Counter32 and Gauge32

— SNMPv1 Universal INTEGER restricted to 32 bits, SMIv2 adds — NsapAddress : OSI NSAP address encoding — Counter64 : For counters that could wrap in less than one hour with only 32

bits — UInteger32 : For integers ranging from 0 to 4294967295

In addition to the UNIVERSAL classes a number of application defined data typesare used. These data types are specific to IP and SNMP.

An IP address is in practice a 32 bit field but it is always written as 4 decimalnumbers separated by dots. SMI introduces a data type that is always presented indotted decimal form with leading spaces removed.

A counter can be used for counting things. It is different to an integer in that its

size is fixed (32 bits) and when it reaches the maximum value held in 32 bits itwraps back to zero. The advantage of this is that provided a manager looks at avariable that is held as a counter often enough it can deduce when the wrapping hasoccurred and compensate by adding 2 32 to the value read. Also there is no need toreset counters. By reading their current value and storing this, the increase in acounter can be deduced by retrieving a later value and subtracting the stored initialnumber.




Notes: Notes:


Example VariablesExample Variables

MIB variables Group Meaning type

sysUpTime system Time since last reboot scalarifMtu interfaces MTU size scalarifAdminStatus interfaces up/down scalaripDefaultTTL ip Default TTL value used scalaricmpInRedirects icmp ICMP redirects seen scalartcpMaxConn tcp Max. connection allowed scalar

udpOutDatagrams udp Datagrams sentscalar

ipNetToMediaTable ip ARP table tableipRouteTable ip IP routing table table




Notes: Notes:





SNMP

Chapter Summary




Notes: Notes:


SNMP Protocol StructureSNMP Protocol Structure

• Based on UDP over IP — Stateless data transfer — Simple to operate and implement — No dependence on virtual circuits — Makes NMS location independent

– Can be anywhere on an internetwork• Major drawbacks

— Security less easy to provide than with TCP – Overcome in SNMPv3 or by adding SSL

— Network management reliant on transport – Which may be failing

— No acknowledgements of data receipt

Because SNMP runs above UDP it cannot assume previous interactions will arrivein sequence so each transfer needs to be stateless. Reading individual objects isstraight forward as a request will identify exactly the object to be read. However toread a table that is of unknown length and with unknown content requires steppingthrough the table row by row, remembering each row read and reading the next.

Stateless systems are relatively simple to build but require operations to be small

and self contained. Also maintaining security is not easy to achieve.




Notes: Notes:


SNMP the ProtocolSNMP the Protocol

• SNMP Architecture includes the protocol, SMI and MIBs

• SNMP the protocol is the set of rules for reading and changing data

SNMPmanager

SNMPmanagedsystem

SNMPmanager

UDP

IP

Link Layer

g e t

g e

t_ n e x

t

s e t

g e

t_ r e s p o n s e

t r a p

SNMPagent

UDP

IP

Link Layer

Managedresources

SNMP objectmanipulation

messages(PDUs)

SNMPdevice

g e

t g e

t_ n e x

t

s e t

g e t_ r e s p o n s e

t r a p

SNMP managedobjects

Managementapplication

SNMP

SNMPv1 has a relatively simple structure in that a management application cansend any one of three messages to an agent and receive a get-response in reply.Also an agent can sent a uni-directional trap to the manager which isunacknowledged.




Notes: Notes:


SNMP MessagesSNMP Messages

• SNMP uses a simple fetch–store protocol — get command to fetch a value — set command to store a value — Operations accomplished as side effects of these commands

• There are no commands such as reboot

• It is often called a remote debugging paradigm

SNMP is often said to have a “remote debugging paradigm”. By this is meant thatthe manager can fetch and store data values and observe the responses in much thesame manner that a programmer debugs a program but does this potentially at leastremotely. SNMP has no action messages such as that found in OSI CMIP. Thereason for this is that we wish SNMP to be very general and function across a widerange of machines. Implementing an action requires service functions that just maynot be available on some low level devices. If we wish to achieve an “action”effect then we must implement within the agent a function that undertakes theaction when a variable in the MIB is changed in some manner. There are someMIBs that have such features, particularly within private MIBs.




Notes: Notes:


Managementsystem Agent

Get/ getnext / set

Tr ap/ r esponse

SNMP ActionsSNMP Actions

• SNMP defines four actions that can be performed on data — Get Used to retrieve management data — Get-next Used to retrieve lists and tables of management data — Set Used to manipulate management information — Trap Used by agent to report extraordinary events

• SNMP makes things happen in agent by setting an “action” variable

ANMPv1 has 4 actions. The first three are invoked by the manager and the fourthby the agent. A “get” will retrieve one or more instances of individual variables.A list of object identifiers is included in the get request and the get responseincludes the object identifiers together with their retrieved values. The number ofidentifiers included must be selected so that the response can fit within the limits ofthe maximum packet size supported in both agent and manager.

When building agent devices it is often necessary to implement the code so that theresponse fits within 576 bytes - the smallest maximum transmission unit sizesupported within IP. This avoids the need for the response to be fragmented withthe potential for fragments to get lost in a failing network.




Notes: Notes:


SNMP MechanismSNMP Mechanism

• Data is encoded (serialized) using ASN.1 — ASN.1 Basic Encoding Rules (BER) — Gets around variations in machine architecture

• Simple (trivial) authentication mechanism in use — Uses “community” name to define access rights

– Very limited and easily bypassed — Replaced by Secure SNMP

– Not a problem with SNMPv2 and SNMPv3• Reliant on polling agents at regular intervals

— Inefficient• Most agents trap on limited set of major events

— Manager follows up trap with poll

The data fields and the SNMP protocol data units themselves are encoded using the ASN.1 basicencoding rules which are machine and language independent.

SNMPv1 has very limited security. Each transfer includes a community name which the agentchecks for validity and can be used to control the level of access. However because this value iscarried in clear text it can relatively easily be discovered using a protocol analyzer and so is notconsidered secure. SNMPv3 overcomes this problem by using optionally both cryptographicauthentication and encryption.

Normally an agent process within a managed device would inform the management application iflinks failed or some other critical event occurred using a trap transfer. However since theunderlying network is datagram (UDP/IP) such a trap is not sure to arrive. To overcome this andensure that the management application can become aware of a managed device failing, themanagement application cannot be passive and just wait for a trap transfer to report a failure. Itmust from time to time send a get-request for an object that it is sure the device supports so that itcan verify that the device is still operational. This is known as polling.




Notes: Notes:


SecuritySecurity

• Authentication — Only authorized managers can adjust critical parameters — Masqueraders cannot probe for sensitive information

• Privacy — Prevent unauthorized snooping by eavesdroppers on a LAN

• SNMPv1 support limited to trivial authentication

• SNMPv1 uses “community name” as only authentication — Sent in the clear with each SNMP PDU — This is sometime referred to a kidding yourself security — Defaults to “public”

As the community name used in the security is easily visible with a protocolanalyzer it is generally recognized that SNMPv1 is not in itself secure enough foruse on an open LAN where stations may readily read other users data. This isclearly not secure and so it is generally referred to as “kidding yourself security” asif you think it is secure you are indeed kidding yourself.

By convention if you do not mind other users retrieving SNMP data from your

device then allow access with the default community name “public”. Often this isthe default community name used by vendors when SNMP is implemented initiallywithin one of their products.

Because this is well known it is critical that this value is changed as soon aspossible when security is important.




Notes: Notes:


SNMPv2 and SNMPv3 SecuritySNMPv2 and SNMPv3 Security

• SNMPv2 added optional encryption and MD5 authentication — Keys could be different between each manager/agent pair — Keys could also include clocks from the two devices — In effect the key changed with each clock tick — So secure that under fault conditions it was possible to loose contact

• SNMPv3 adds variable security model — Can be configured with users own authentication and encryption if required — Standard models defined too — Mechanism for aligning times and after reboot to overcome problems with

SNMPv2 — New branch of the MIB used to manage securitysnmpUsmMIB

SNMPv2 added MD5 authentication and DES encryption. It also allowed the valueof the system clock in both agent and manager to be included within the calculationof MD5 encoded values. The impact of this is in effect to change the effective keyeach time the clock ticks (100 times a second). This was found to be impractical inmost real cases as the delay across the network was variable and often its variationwell exceeded the resolution of the system clocks (0.01 sec)

SNMPv3 allows a user defined security model to be used if needed so it can bemade as secure as required.




Notes: Notes:


Error ReportingError Reporting

• SNMPv1 is “all or nothing” — Each PDU is treated as an atomic operation — Any syntax errors cause the operation to be aborted

• SNMPv2 and SNMPv3 are more forgiving — Other mechanisms for access to tables — The impact of errors is limited to the smallest scope possible

If a management device makes a request of the agent using a message containing anerror, the whole transaction is ignored. This can become a source of frustrationwith SNMPv1 so in SNMPv3 get-requests for objects that are correct even whensome part of the message was in error will still be actioned. Error reports indicatethe reason for the error and a pointer points to the incorrect syntax in the messagereply




Notes: Notes:


SNMPv1 FunctionsSNMPv1 Functions

• Get — Get-request — Get-response

• Get-next — Get-next-request — Get-response

• Set — Set-request — Get-response

• Trap — Trap

(a) Get values (b) Get-next values

(c) Set values (d) Send trap

G e t R e q u e s t P D U

G e t R e s p o n s e P D U

G e t N e x t R e q u e s t P D U


S e t R e q u e s t P D U


T r a p P D U

Manager Agent Manager Agent

Manager Agent Manager Agent

Get-request, getNext-Request and set-request all start from the managementapplication and each have the same response — a get-response. In SNMPv3 this isretitled a Response.

A trap passes from the agent to the manager and has no response.




Notes: Notes:


GetGet

• Used to retrieve management information

• Gets a specific instance of a specific variable

• Can specify more than one item — As many as will fit in one SNMP packet

• All variables must be correct or the command is ignored — Can be used to retrieve only items whose complete OID is known

The get is used to read individual scalar values. In general it is not possible to readtables with a get.

References to objects must point to the individual leaf entries that contain actualvalues and not to nodes that represent tables or sequences of values. Tables areidentified as multiple instances of the same variables identified from each otherusing the value of the index entries associated with the row in the table. Each row

must have a unique index value and this forms part of the identification of thevariables in the row. In order to reach a table with a get it is necessary to read eachelement one at a time and it is necessary to know the value of all indexes associatedwith the row. The index values are generally themselves columns of the table andso to read a table it is necessary to know already the values of the index entries.But these too are indexed with the same value so it is in practice unlikely that amanagement application would already know all the required index values withreading the table itself.




Notes: Notes:


Get-responseGet-response

• Returns the value requested by an SNMP Get

• Consists of pairs of variable instance and value

• Also used to respond to Get-Next and Set requests

The get-response includes within its reply the exact objuct identifier previouslyrequested together with its value .




Notes: Notes:


GetNext-requestGetNext-request

• Function and interpretation are identical to Get with one exception — Returns the oid of the next variable instance in the MIB tree and its value,

rather than the one specified in the request• Allows exploring MIBs to determine their contents

— An MIB “walk”• Allows reading tables of unknown length

GetNext is used to read tables. Instead of returning the value of the object whoseidentifier is included in the request, getNext returns the immediately followingelement within the MIB together with its object identifier. The returned objectidentifier can then be used in the following getNext-request in order to return thenext following element again. In this way the manager can step through theelements within a table or even the whole of the MIB one element at a time withoutneeding to know the details of the contents until it drops off the end of the requiredtable or indeed the whole MIB. This is known as “walking the MIB”.




Notes: Notes:


SetSet

• Used to set an MIB variable to a specific value

• Community name provides the only protection against invalid use

• Each SNMP packet is “all or nothing” to prevent ambiguous results

• Managed device can be instructed to take an action by setting the value ofthe appropriate MIB variable

A set takes the same format as a get but each object is immediately followed by avalue to be used by the agent to set the required variable. The returned get-response verifies what value has actually been set.




Notes: Notes:


SNMPv1 TrapSNMPv1 Trap

• Sent by an SNMP agent to a manager

• Provides asynchronous notification of an event

• No response is expected

• Seven generic trap types defined:

• Trap Value— 0 coldStart— 1 warmStart— 2 linkDown— 3 linkUp— 4 authenticationFailure— 5 egpNeighborLoss— 6 enterpriseSpecific

The SNMPv1 trap passes from the agent to the manager and can be one of 7 types:-0 coldStart sent when device is powered up from cold

1 warmStart sent when a device is reset or restarted

2 linkDown sent when a link fails

3 linkUp sent when a link returns from a failed state

4 authenticationFailure sent when a message with incorrectcommunity name received

5 bgpNeighborLoss sent when contact with the Internet

border gateway is lost

6 enterpriseSpecific Some other meaning devices by the vendor.

The field following the trap type gives a vendor

specific number defining the trap meaning.




Notes: Notes:


SNMPv2 and v3 Additional FunctionsSNMPv2 and v3 Additional Functions

• GetBulkRequest — Efficiently retrieve large blocks of information

• InformRequest — Manager can reliably send a trap to another manager

• SNMPv2 and SNMPv3 have significantly improved security — Authentication and privacy

• Improved error handling and reporting

SNMPv2 and SNMPv3 add two additional messages. These improve the efficiencyretrieving tables and provide a transfer that acts like an acknowledged form of Trap.




Notes: Notes:


GetBulk-requestGetBulk-request

• Similar to get_next but adds two new fields — Nonrepeaters and max-repetitions

• First <non-repeaters> variables in list treated as get-next• Remaining variables in list will respond as if get-next had been repeated

<max-repetitions> times — Or until an MIB “leaf” other than another instance of the variable requested

is returned

Within a getBulk-request two parameters are given in addition to a list of objectidentifiers. The first gives the number of the object identifiers that are to beretrieved once only. These will normally be scalar variable rather than tables. Theremaining entries are assumed to be table objects and are repeatedly retrieved as ifrepeated getNext-requests had been used with the returned object identifiers usedfor the next request. The second parameter limits the maximum number of timesthe repetitions will be returned. The returns will stop either when the maximumrepetitions is reach or the end of the table is reached which ever comes first.




Notes: Notes:


InformRequestInformRequest

• Sent by one SNMP manager to another SNMP manager

• Similar in function to SNMPv1 trap , except it is acknowledged

• The variable bindings field always contains at least two elements— sysUpTime (defined in MIB-2)— SNMPv2EventID (defined in the manager-to-manager MIB) — May include additional items if specified by the requesting manager

The InformRequest transfer acts like a reliable trap in that the receiving manageracknowledges receipt. Normally this is used between different managers to allow aremote manager to forward alarms or alerts in a reliable manner to a globalmanagement station at a distant site. It is normally used in conjunction with themanager-to-manager MIB that includes an event table. This lists events that haveoccurred and includes details of what they are with pointers to more information ifrequired in a log table. The transfer needs then only to include the index entry tothe event table and the time. The receiving manager can retrieve more details aboutthe alarm if needed by using get, getNext or getBulk on the event table using theindex given.




Notes: Notes:


Single and Multi-valued ObjectsSingle and Multi-valued Objects

• Objects have multiple instances — one for each row

• Each row has an index made up from different values

— Normally different columns in the same table• For simple scalar objects the instance is taken as zero

ifIndex

Integer

Integer

Instance 0

Instance 1Instance 2

DisplayString

DisplayString

12

Type Value Type ValueEthernet1Ethernet2

ifType

Integer

Integer

66

Type ValueifMtu

Integer

Integer

15001500

Type Value

ifDescr ...

...

...

...

Scalar object exampleObject name Instance Type Value

sysContact 0 DisplayString Groucho Marx

Table object example (single index)

ifTable {1.3.6.1.2.1.2)

In order to reduce the amount of code that would be needed within a agent, theformat of object identifiers used for individual scalar object and for table entries arelargely the same. To identify a table object the object identifier of the columnobject has all the values of the indexes appended to it in order. Scalar objects areassumed to have only a single index with a single value “.0” which is appended tothe end of the identifier for the object to produce the “instance identifier”.




Notes: Notes:


Scalar ExampleScalar Example

• Object name Object identifier Instance identifier

— tcpRtoAlgorithm 1.3.6.1.2.1.6.1 1.3.6.1.2.1.6.1.0— tcpRtoMin 1.3.6.1.2.1.6.2 1.3.6.1.2.1.6.2.0— tcpRtoMax 1.3.6.1.2.1.6.3 1.3.6.1.2.1.6.3.0— tcpMaxConn 1.3.6.1.2.1.6.4 1.3.6.1.2.1.6.4.0— tcpActiveOpens 1.3.6.1.2.1.6.5 1.3.6.1.2.1.6.5.0— tcpPassiveOpens 1.3.6.1.2.1.6.6 1.3.6.1.2.1.6.6.0— tcpAttemptFails 1.3.6.1.2.1.6.7 1.3.6.1.2.1.6.7.0— tcpEstabResets 1.3.6.1.2.1.6.8 1.3.6.1.2.1.6.8.0— tcpCurrEstab 1.3.6.1.2.1.6.9 1.3.6.1.2.1.6.9.0

Here are some examples of instance identifiers that could be used to retrieve scalarobjects in a get-request. Notice that the instance identifier is just the objectidentifier with “.0” appended.

Out of interest these are all of the scalar values in the TCP group of MIB-2(RFC1213).






Notes: Notes:


Overview of MIB TableOverview of MIB Table

ip

1

1 2 3 4

22

4

ipNe tToMedi aTabl e

ipNe tToMedi aEnt ry

ipNe tToMedi a I f Index ipNe tToMedi aPhysAddress ipNe tToMedi aNe tAddre ss i pNe tToMed i aType

3

3

3

3

00 00 c0 c5 ed 91

00 00 c0 10 ce 70

1 4 4 . 1 9 . 7 4 . 5

1 4 4 . 1 9 . 7 4 . 6

One of the most widely referenced tables is the ARP table which can be found asthe ipNetToMediaTable (node 22) in the ip group. All tables tend to be constructedin much the same manner with a “table” definition followed by an “entry”definition under which is a node for each column of the table.




Notes: Notes:


Access to This TableAccess to This Table

• A get-next request can be made to each of the column entries

• The first row of the table will be returned together with its identifier and

index values• Get-next can be repeated using the returned object identifiers

— The reply will be the next row of the table• Tables must be read row by row with SNMPv1

• SNMPv2 or v3 allows a Get-bulk — Each row will yield a repeated reply saving half the transfers

A row of a table can be read in a single transfer by using a getNext-request thatrefers to each of the column nodes within the object fields in the snmp message.The response will then include the fully qualified object identifiers for the next(first) entries including their index fields followed by their values.






Notes: Notes:





SNMP

Chapter Summary




Notes: Notes:



In this chapter we have

• Examined the way in which Network Management stations communicate

with managed devices• Identified the structure of SNMP for management

• Detailed the component parts of SNMP management




Notes: Notes:

Next Generation Network

Technology

Next Generation NetworkNext Generation Network

TechnologyTechnology

Chapter 9Chapter 9




Notes: Notes:




• Identify the key technologies that will form the foundation of 21CN

• Compare Access options

• Expose the advantages of MPLS switched core

• Describe how voice will be carried over the IP infrastructure

• Describe how QoS can be delivered for multimedia services

• Examine new applications that will lead customer demand for 21CN service




Notes: Notes:


Next Generation Network TechnologyNext Generation Network Technology

Next Generation Architecture

xDSL Technologies

Deploying IEEE 802.1q VLANs

Core Technologies

QoS

Voice Services

New Applications: IPTV

Chapter Summary




Notes: Notes:


21CN High Level Architecture21CN High Level Architecture

CustomerEnvironment

ComplexCost SensitiveMulti-FunctionMulti-DeviceMulti-Service

Real Time andStreamed

ConsumerSME

Enterprise

MSAN

SimplifiedAggregation andConcentration

xDSLEthernet

WiFiWiMax

Metro

Complex, Costly,Large, Fast

Multi-functionWire-speed processing

Multi-access Aggregation

Application and ServiceInsertion Point

IMS service brokerrequests

Core

Optical Bulktransport

BigFast

Cost-effective

Millions ∼∼∼∼ 5500 ∼∼∼∼ 86 ∼∼∼∼ 20

Locations in UK:

The Architecture can be divided into 4 parts. The customer environment is todaymulti-functional, complex and consumer oriented. It is very cost sensitive and mustdeiver real-time streamed services as well as messaging and data services. The21CN network will have an interface to the customer through the MSAN which willbe simplified and reliable. Reliability will be ensured using aggregation andprotection switching. At the edge the service will be simple in order to controlvolume costs but at the Metro level routing and customer services will be injected.The core is fast and optical.




Notes: Notes:


Very Large Next Generation Carrier NetworksVery Large Next Generation Carrier Networks

CoreSwitches

Metro Nodes

MSANs

In very large country wide installations a multi level hierarchy of devices will beneeded. Each MSAN might support a few thousand homes in a town or perhaps afew hundred in a rural community. Indeed some technologies that run at very highspeed over copper pairs may restrict access distances to 100s of metres and so wemight see MSANs on street corners providing VDSL access at 20Mbit/s.

With so many distribution devices, perhaps more than 1000 and perhaps as many as30,000 in the UK, these might be concentrated down over two levels.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:


2015 Percentage of Households Access Speed2015 Percentage of Households Access Speed

• Access by wired line or wireless

2 Mbit/s or less5 km limit

6 Mbit/s3.7 km limit

20 Mbit/s1 km

100 Mbit/sAnd above

0.3 km

20%

40%

60%

80%

1995 2000 2005 2010 2015 2020

ADSL

VDSL

Fiber

If the demand for increasing speed continues as it has in recent times we can expectto maintain the services with current technologies until 2010 when Wireless andeventually Fiber will take over.




Notes: Notes:


xDSLxDSL

• x DSL (digital subscriber line) refers to a family of techniques that useexisting copper loops for high-bit-rate signals

•

There are several types of DSL modems — ADSL (asymmetric DSL) — HDSL (high-data-rate DSL) — SDSL (single-line DSL) — VDSL (very high-data-rate DSL)

• DSL is transmission-level convergence — The voice and data/video signals happen to use the same physical pipe — They have no other relationship

Digital subscriber line, or loop as the US call it, is the dominant area ofdevelopment for the next generation. Price per line, flexibility of downloadingencoding firmware, low power requirements and high packing densities will alldrive the direction of the technology.




Notes: Notes:


xDSL ComparisonxDSL Comparison

Internet access,telecommuting,video-on-demand

9,000–18,000 feet (2.7–5.5 km) of1-pair UTP

1.5–12 Mbit/s down,16–640 kbit/s up

26 Mbit/s down

ADSL

ADSL2+

ATM, HDTV900–5,000 feet (0.3–1.5km) of1-pair UTP

13–52 Mbit/s down,1.5–2.3 Mbit/s up

100Mbit/s +

VDSL

VDSL2

T1/E1 replacement10,000 feet (3 km) of 1-pair UTP

1.5–2.0 Mbit/s,symmetric

SDSL

T1/E1 replacement15,000 feet (4.5 km) of2- or 3-pair UTP

1.5–2.0 Mbit/s,symmetric

HDSL

ApplicationDistanceBit ratesType

HDTV = high-definition televisionSee www.adsl.com, www.dslforum.organd ITU-T Recommendations G.991 –G.997

At the moment ADSL running over ATM dominates. Currently delivering up toabout 8 Mbit/s, this is expected to evolve using ADSL.2 to perhaps 16 Mbit/s.

Eventually however it is expected that VDSL running over Ethernet will becomemore important as already higher speeds seem achievable and Ethernet presentationat the Metro Node make this more attractive within the architecture.To reach the upper speed limits of either technology the loop length must be

restricted to much less than 5 km, to perhaps 1km or even less. This will meandeploying MSANs in street furniture or even underground if loop services are toremain on copper.




Notes: Notes:


ADSL2+ subscriber rangeADSL2+ subscriber range

There has been considerable development of these technologies over the last 5years. The advances of ADSL to ADSL2 and then ADSL2+ has increased thedownlink speed to above 16 Mbit/s and on very short loops even further. Howeverunder 40% of the loops in the UK are less than 1km and 40% over 2 km.




Notes: Notes:


VDSL2VDSL2

• Deteriorates quickly from a theoretical maximum of 250 Mbit/s

• 100 Mbit/s at 0.5 km and 50 Mbit/s at 1 km

• Degrades at a much slower rate from there, and still outperforms VDSL.

• Starting from 1,6 km its performance is equal to ADSL2+.

VDSL2 (Very-High-Bit-Rate Digital Subscriber Line 2, ITU-T G.993.2 Standard) is an accesstechnology that exploits the existing infrastructure of copper wires that were originally deployed forPOTS services. It can be deployed from central offices, from fibre-fed cabinets located near thecustomer premises, or within buildings.ITU-T G.993.2 VDSL2 is the newest and most advanced standard of DSL broadband wirelinecommunications. Designed to support the wide deployment of Triple Play services such as voice,video, data, high definition television (HDTV) and interactive gaming, VDSL2 enables operatorsand carriers to gradually, flexibly, and cost efficiently upgrade existing xDSL-infrastructure.ITU-T G.993.2 (VDSL2) is an enhancement to G.993.1 VDSL that permits the transmission ofasymmetric and symmetric (Full-Duplex) aggregate data rates up to 200 Mbit/s on twisted pairsusing a bandwidth up to 30 MHz.VDSL2 deteriorates quickly from a theoretical maximum of 250 Mbit/s at 'source' to 100 Mbit/s at0.5 km and 50 Mbit/s at 1 km, but degrades at a much slower rate from there, and still outperformsVDSL. Starting from 1,6 km its performance is equal to ADSL2+.ADSL-like long reach (LR) performance: ADSL-like long reach performance is one of the keyadvantages of VDSL2. LR-VDSL2 enabled systems are capable of supporting speeds of around 1-4Mbit/s (downstream) over distances of 4 to 5 km, gradually increasing the bit rate up to symmetric100Mbit/s as loop-length shortens. This means that VDSL2-based systems, unlike VDSL1 systems,are not limited to short loops or MTU/MDUs only, but can also be used for medium rangeapplications.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:


VLAN

Network-managementstation

VLAN ConfigurationVLAN Configuration

• The network manager configures the VLAN membership — Better than re-cabling — But, still requires manual effort — Someday, may be automated (artificial intelligence)

• Provides isolation between different carrier services

VLAN operation enables a manager to group ports together reflecting the manner inwhich they normally interact. Separation into groups improves security.Communication with devices in different groups should be rare but if required fromtime to time can be provided by a router, either separately connected or a a functionwithin one of the switches.




Notes: Notes:


VLAN Trunking TagVLAN Trunking Tag

DestinationAddress

SourceAddress

Ether-Type/ Length Payload CRC

4-byte Tag Header contains:· 2-byte Tag Protocol Identifier (TPID) with a fixed value of 0x8100.

Indicates that frame carries the 802.1Q/802.1p Tag information.· 2-byte Tag Control Information (TCI) with the following elements:

TPID Priority CFI VID

Normal 802.3Frame

TAG Inserted for802.1P/Q

3-bit user_priority : 3-bit binary number capable of representing priority levels, 0 through 7.This field is used primarily by the 802.1p standard.

1-bit Canonical Format Indicator (CFI) : With a CFI value of 0, it indicates canonical format;A value of 1 indicates non-canonical format used in Token Ring and FDDI

12-bit VLAN Identifier (VID) : VLAN 0 and 4095 are reserved, other values represent VLANs.This field is used primarily by the 802.1Q standard.Standard allows less than the maximum 4094 VLANs to be supported.

TCI

For details see IEEE Std 802.1Q, 2003 Edition

IEEE 802.1Q adds a 4 byte shim header to the Ethernet frame. The first two bytesare in effect a new Ether-type field that identifies the presence of the 802.1Qinformation. The second two bytes hold the 12 bit VLAN identifier plus 4 furtherbits used by 802.1P for priority.




Notes: Notes:


Additional Features on Switches:802.1PAdditional Features on Switches:802.1P

• IEEE 802.1P Priority for class of service

• Allows user configuration of service class

— Station assigns a priority value to each frame — Switch assigns to one of a number of queues dependent upon value — Example

user_priority Traffic Type

1 Background

2 Spare

0 Best Effort

3 Excellent Effort

4 Controlled Load

5 Video

6 Voice

7 Network Control

User Priority to Traffic Class mapping

The first three additional bits hold a priority field which give 8 different prioritylevels.




Notes: Notes:


Enabling TechnologiesEnabling Technologies


xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:


Delivering QoS in the Core NetworkDelivering QoS in the Core Network

• The Internet has evolved into a ubiquitous network — Not designed to give QoS to voice or other real-time users

• New applications demand increased core bandwidth• Traditional routing at Layer 3 at high speed is not feasible in software

— Need switching in hardware at Layer 2 or Layer 3• Multi-Protocol Label Switching (MPLS) implemented in Label Switching

Routers (LSR)

LSR LSR LSR

LSR LSR LSR

We are now over the next 3 slides going to talk about MPLS. The aim is not to gointo much detail but to deliver just a flavor of what MPLS can do in the core of alarge network

Those parts of the core which are to carry quality of service traffic would beenhanced to support MPLS. In practice this usually implies that the hardware insome manner assists the switching of packets based upon labels added at the ingess(entrance) router and removed at the egress (exit) router.




Notes: Notes:


PPP Shim MPLS Header Layer 3

LSR LSR LSR

LSR LSR LSR

55 17 17 15 15

33

Labeladded

Labelremoved

LER LER

Multi-Protocol Label SwitchingMulti-Protocol Label Switching

• MPLS adds a label as a shim header above the link header and belowNetwork Layer — Includes a 20-bit label and TTL — It can use identifiers instead in ATM and frame relay

• Label Edge Routers (LER) add and remove labels

• LSRs use Label Distribution Protocol to agree labels for destination

The label is normally 32 bits long and includes a 20 bits in the label and 8 bits inthe TTL. The routers use LDP to obtain the labels to use for the destination andload these into switching tables used by the hardware of the LSR. The Layer 3software is then not involved in delivery of the packets other than at entry and exitso reducing the load on the core.

The actual value used I the label will be mapped and changed as packets pass frominput to output interface at the LSRs. This again will be hardware assisted.




Notes: Notes:


47.1

47.247.3

IP 47.1.1.1

D es t O ut4 7 .1 14 7 .2 24 7 .3 3

1

23

D e s t O u t4 7 . 1 14 7 . 2 24 7 . 3 3

12

1

2

3

IP 47.1.1.1

IP 47.1.1.1IP 47.1.1.1

D e s t O u t4 7 . 1 14 7 . 2 24 7 . 3 3

Normal Routing by IP Forwarding Hop-By-HopNormal Routing by IP Forwarding Hop-By-Hop

Routing Table

Once the Label Switched Path databases are complete traffic is forwarded hop byhop based upon these tables.




Notes: Notes:


IntfIn

LabelIn

Dest IntfOut

3 0.40 47.1 1

IntfIn

LabelIn

Dest IntfOut

LabelOut

3 0.50 47.1 1 0.40

47.1

47.247.3

12

3

1

2

1

23

3IntfIn

Dest IntfOut

LabelOut

3 47.1 1 0.50 Mapping: 0.40

Request: 47.1

M a p p i n g : 0. 5 0

R e q u e s t : 4 7. 1

MPLS Label DistributionMPLS Label Distribution

Labels are distributed along IP routes.






Notes: Notes:


IntfIn

LabelIn

Dest IntfOut

LabelOut

3 0.50 47.1 1 0.40

47.1

47.247.3

12

31

2

23

3

IntfIn

Dest IntfOut

LabelOut

3 47.1.1 2 1.333 47.1 1 0.50

IP 47.1.1.1

IP 47.1.1.1

Explicitly Routed LSP ER-LSPExplicitly Routed LSP ER-LSP

1

• We can have multiple paths based on QOS or address lengthIntfIn

LabelIn

Dest IntfOut

3 0.40 47.1 1

When complete the database of LSPs is in place the service can be viewed as atunnel from Ingress to Egress.

With Explicitly Routed LSPs the ingress router can slect which tunnel to use basedupon Diffserv QOS or even IP address.




Notes: Notes:


Label Exp. S TTL

Label: Label Value, 20 bits (0-16 reserved)Exp.: Experimental, 3 bits (was Class of Service)S: Bot tom of Stack, 1 bit (1 = last entry in labe l s tack)TTL: Time to Live, 8 bits

Layer 2 Header

(eg. PPP, 802.3)

•••Network Layer Header

and Packet (eg. IP)

4 Octets

MPLS ‘Shim’ Headers (1-n)

1n

MPLS on PPP links and LANs uses ‘Shim’ Header InsertedBetween Layer 2 and Layer 3 Headers

MPLS on PPP links and LANs uses ‘Shim’ Header InsertedBetween Layer 2 and Layer 3 Headers

Label StackEntry Format

MPLS Encapsulation - PPP & LAN Data LinksMPLS Encapsulation - PPP & LAN Data Links

Shim labels will be used for encapsulation over LANs. Some carriers are nowconsidering building long haul services over Gigabit Ethernet and these too woulduse shim labels.




Notes: Notes:


MPLS TunnelingMPLS Tunneling

• MPLS maps labels from input interface to output interface — LSRs can use hardware assistance to switch labeled traffic at link speed — Possible to reach speeds up to 10 Gbit/s

• Selection of label and thus the path chosen can be related to QoS — Selected by protocol, TOS, RSVP flow, or other recognizable characteristics

• MPLS can control the entire path — Tunnels created end-to-end and one MPLS header is wrapped within

another — One header identifies the end destination network — Second outer link label can identify intermediate network point — This enables end-to-end quality issues to be coordinated for the traffic

without the need to know complete path• MPLS is not a requirement for VoIP

— Enables QoS to be delivered more easily in high bandwidth core networks — Thought to be feasible at higher speeds than ATM

Traffic destined for the same destination but of a different class of service mayfollow different paths. It is possible to control and coordinate the end to end QOSby allocating a destination label for the egress network at the LSR and adding alabel referencing the class of service required. This would then be tunneled withina second label at the ingress router that address the egress LSR itself rather than the

exit network. In this way the egress router can handle the exit traffic of differingQOS arriving from the same source.




Notes: Notes:


Pseudo WiresPseudo Wires

• IETF are developing telecommunications protocols for MPLS

• Multiple labels may be carried by a packet

— Outer labels defining the service or application — Inner label identifies the path to the destination• Telecommunication services would look like wires and so called Pseudo-

Wire Emulation — Uses protocol called PWE3

IP #L1#LpIP #LpIP #L2#Lp

IP #L3#Lp IP #Lp

IP #L1#LvIP #LvIP #L2#Lv

IP #L3#Lv IP #Lv

In telecommunications services circuit switching has been in use since theinvention of the telephone exchange in 1892. In a circuit, bits pass as a stream withsynchronous timing in use with a network controlled constant rate clock. It isdesirable that in the future we would build networks that carry both packet andcircuit services but share a common core. This has been done for some time usingcircuit based cores deployed using ATM, but these tend to be expensive andinefficient. Current IETF thinking suggest Gigabit Ethernet technology will be thepreferred option using a packet based core.MPLS can be used to deliver the required QOS and by using multiple labels. Onelabel can define the edge service required (say phone or video) while another outerlabel can be used inside the network to deliver the QOS.




Notes: Notes:


Using MPLS in the 21CN CoreUsing MPLS in the 21CN Core

Initial Deployment• Protocols: IPv4, OSPF, MPLS, LDP (DU, LL ret)• Virtualisation: IP VPNs (RFC4364 [RFC2547bis ]),PWE3 (RFC3916)• Resilience: Tx and FRR (RFC 4090)• GR (RFCs 3623, 3478, 3473)• QoS: 7 levels (6 user, 1 control)• Connectionless• Unicast

Possible Future Deployment• Many additional VPNs (000s)• Traffic engineering via RSVP-TE (RFC3209)• Capacity and failure response optimisation viaa bandwidth manager – connection orientedflows• Multicast• IPv6

Across the 21CN core MPLS will deliver labelled circuits that can carry trafficwith guaranteed QoS between Metro-Nodes at the edges. Using Pseudo-WireEmulation, TDM and non-native IP services can be provided. This will includeEthernet, T1 and other PSTN truning services.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:


FRTS = Frame Relay Traffic ShapingRSVP = Resource Reservation ProtocolWFQ = weighted fair queuing

Quality Delivery OptionsQuality Delivery Options

• There are two approaches to delivering voice quality — High-bandwidth provisioning

– Easy and cheap with switched LANs where Gbit/s speeds are possible – Not so easy or cheap to apply over WAN services

— Use QoS tools to give voice traffic the conditions for good quality• Tools for maintaining voice quality:

— QoS signaling – RSVP, IP precedence, DiffServ

— Prioritization tools: queuing techniques; e.g., WFQ — Slow-link efficiency tools; e.g., MTU reduction — Call admission control — WRED

We have already seen that the key elements that impact quality are delay andpacket loss. Delay is made worse by jitter, which can be overcome only byincreasing the delay. Packet loss generally occurs when queues overflow andqueues are caused by varying load. There are basically two approaches we cantake:

1. Make the network capacity so large delay and load are insignificant. If we sizeevery trunk and every router so that it always runs below 50% capacity then delaywill be low. But many people say they cannot afford to do this so we must findways of ensuring that those parts of the data that need good QOS can get it even atthe expense of other data.

2. Take QoS measures!

We will work through the QoS measures possible (shown in the sub-bullets of thesecond bullet above). I personally keep returning to this slide to check off eachtechnique as we cover it.




Notes: Notes:


High-Bandwidth ProvisioningHigh-Bandwidth Provisioning

• Quality of service is largely limited by queues within the network

• This is particularly a problem where sources of data burst to high rates

— Steady rates can be sized and provisioned readily — When the maximum capacity is very high, cost of aggregated maximumbecomes too great

• Aggregating sources into a very high-bandwidth backbone is a solution — Peaks of demand tend to average out — The greater the number of sources, the more stable the average

• A measure of bursty nature of a traffic source is sporadicity (S) — S = (Maximum throughput) / (Average throughput)

– e. g. 1 stream of G.711 at 64 kbit/s voice at 40% activity: – S = 64000 / ( 64000 x .4) = 2.5

— For aggregated channels maximum value is taken using confidence level – Normally use 99% confidence level

If we remove the jitter which is caused by variations in delay then the overall delaywill come down significantly. One measure of jitter is sporadicity. There is no

jitter in circuit switched networks generally because the max throughput andaverage throughput of each channel is the same and so sporadic is 1. On VoIPnetworks, particularly where there is silence suppression, this is no longer true. Butif we aggregate many calls into a single high bandwidth network the averagesporadicity will reduce improving jitter. This really means that we are better offwith a few shared high bandwidth trunks than many individual low bandwidth ones.

If we size our VoIP network with no over subscription (S=1), that is on a networkusing 64 kbit/s G.711 codes we allow say 100 kbit/s per user then we will neverhave a problem as there is so much bandwidth the limit cannot be reached. On a100-BASE-T LAN we could size at say 25 Mbit/s and so could support say 250channels at this level of sizing.If we want to go beyond this then we can either go to switched 100-BASE-T with100 Mbit/s each and never reach the limit or start taking the activity level intoaccount. At .4 utilization our 25 Mbit/s of throughput would deliver say 2.5 times250 channels.

The Quality spreadsheet will calculate for you confidence intervals.




Notes: Notes:


Sporadicity of Aggregated StreamsSporadicity of Aggregated Streams

10 100 1000 10000

S p o r a

d i c i t y

1

2

3

4

Number of streams

• The greater the number of streams,the more stable the loading will be,and thus the delay will be lower and stable

• Typically work with a 99% confidence level

Big is beautiful!

The key message sporadicity delivers is that low bit rate links such as 56 kbit/smodem access are bad news because they have only one channel and so highspradicity. If I try to speak at 64 kbit/s then with IP headers I will need more than64 kbit/s and a lot more than the 40 kbit/s my modem delivers. I am bound to getclipping and bad quality.

If I take say 100 channels each at 64 kbit/s then it is unlikely that everyone will talk

at the same time.High bandwidth provisioning requires say 100x100kbit/s = 10 Mbit/s

Using circuit switching we need 6.4 Mbit/s (64000x100)

Using the Quality Spreadsheet the 99% confidence level is 51 speakers, that is tosay 99% of the time there will be less than 51 speakers. This will need 4.16 Mbit/sto carry the traffic, including all the overheads for IP headers.




Notes: Notes:


Controlling QoS With RSVPControlling QoS With RSVP

• Resource Reservation Protocol (RSVP RFC 2205) — Used by hosts to request QoS — Used by routers to provide QoS — Dynamic; responds to changing requirements

• RSVP is a transport-level signaling protocol — Reserves resources along the path of the call

• Routers must be capable of RSVP operation — Traffic control is required to implement RSVP

– Implies a method to control the use of RSVP – Resources must be available – Authorization to use those resources – Need end-to-end capability




Notes: Notes:


Intserv QoS ControlIntserv QoS Control

• Two QoS types — Controlled load (RFC 2211)

– Performance should be similar to uncongested — Controlled quality (RFC 2212)

– Limits on delay – Guaranteed bandwidth – Based on MTU, data rate, buffers, etc.

• Specified in reservation message — From receiver to source of signal

• Interpreted by routers along route — May be redefined along route

• Confirmed by initial sender (source of signal)




Notes: Notes:


RSVP and IntservRSVP and Intserv

• Integrated services (Intserv) — Defines some QoS parameters, and — How to use them with RSVP

• Sender (transmit end) in new session registers with RSVP — Describes the traffic to be sent — Describes sender QoS capabilities — Sent in path message to receiver

• Path message processed and possibly modified by nodes along route — Describe capability to conform to the traffic defined by the source

• Receiver interprets result and forms a reservation request — Based on the request of the source as modified by routers on the route — Sent back to the sender via the same route — There may be many such receivers

– Reservations are merged as they are combined in the reverse direction

The integrated services model, or intserv, negotiates a particular QoS at the time itis requested. Before exchanging traffic, the sender and receiver request a particularQoS level from the network. Upon acceptance, the intermediate network devicesassociate the resulting traffic flow with a specific level of jitter, latency, andcapacity. Resource Reservation Protocol (RSVP) is an example of such a model.

Here’s Cisco’s definition of RSVP:

The Resource Reservation Protocol (RSVP) is a network-control protocol thatenables Internet applications to obtain special qualities of service (QoSs) for theirdata flows. RSVP is not a routing protocol; instead, it works in conjunction withrouting protocols and installs the equivalent of dynamic access lists along the routesthat routing protocols calculate. RSVP occupies the place of a transport protocol inthe OSI model seven-layer protocol stack.




Notes: Notes:


Use of RSVP Path and Reservation MessagesUse of RSVP Path and Reservation Messages

• Path message defines the QoS request of the sender — Nodes along the route retain the path information

– Confirm or deny the ability to provide the required resources

• Receiver verifies the accumulated confirmations — Creates a reservation message; sent back to sender along the same path — Confirms to the nodes that the reservation should be initiated

— May be collected, combined with many reservation messages from severalreceivers

Path message

Reservationmessage

A VoIP node can signal that it will connect via RSVP and then from the source aPath message is sent containing the QOS information through each router betweensource and destination. If it gets through satisfactorily the respondent sends back areservation message which follows back along the same route and each routerallocates the capacity. As routes change from time to time the exchange is repeated

every now and again and the old paths timeout and are removed freeing up therouter resources.

Netmeeting sends RSVP packets so if attendees are real interested in this, showthem with a Netmeeting call and Ethereal.




Notes: Notes:


Problems With RSVPProblems With RSVP

• RSVP reserves resources based on required bandwidth, IP address, andport (socket) — However, in H.323 call signaling, this is not determined until

– After ringing starts, and – After capabilities exchange, and – Logical channels are selected

— Call may be answered before RSVP can be confirmed – May not get required resources!

• RSVP and queuing priorities cooperation is lacking — For example, WFQ

RSVP uses a pair of IP addresses and a pair of port numbers to form theidentification of the flow. This is a layer 4 address. Routers are intended to belayer 3 devices and adding this functionality greatly increases the processingneeded for every packet not just voice packets. At the heart of the internet this is abig problem and RSVP would not be feasible. This can probably only be

guaranteed on a purpose-built Intranet.




Notes: Notes:


RSVP: LimitationsRSVP: Limitations

• Best suited to intranets instead of Internet — Scalability is poor — More suited to broadcasts; i.e., video

• Internet barriers — Contrary to best-efforts delivery for all packets — Requires billing for better service in public networks — Interconnected autonomous systems may not cooperate, and — Route control may not always be possible

• Intranet barriers — Conflicting priorities between services

– Which services get priority, resource reservation — May simply require more resources (bandwidth)

• Eventually RSVP may be used to set up QoS MPLS paths but initially othersimpler mechanisms will be used




Notes: Notes:


IP PrecedenceIP Precedence

• IP precedence is included in the IP packet header in the TOS field

• With IP precedence, voice can be given priority over data

• Format of precedence in TOS byte:

Precedence

RFC 791 Precedence000 Normal001 Priority010 Immediate011 Flash100 Flash Overide101 Critical

110 Internet Management111 Network Management

The second byte of the IP header was originally called the Type of Service byte inRFC 791. This enabled precidence to be given to one service over another whenrouters routed in “seconds per packet” during the 1970s. As routing speeds incresedduring the 1980s and 1990 this fell out of use. The IP TOS field can be used tohold the precedence. This can now be extended from 3 bits to 6 bits and re-titled theDifferential Services Code Point or DiffServ for short.

The differentiated services model, or diffserv, takes a different approach. A few,coarse classes of traffic handling-similar to gold, silver, or bronze levels of frequentflier cards-are established by the network administrator. When the sender needs aparticular kind of handling, it marks the individual packets accordingly.




Notes: Notes:


DiffServDiffServ

• Work in progress — Higher priority as a result of VoIP requirements for QoS

• Objectives: — Control throughput, delay, jitter, and/or loss — Permits priority access to the data network — Deals with the special requirements of some applications — Attempts to satisfy expectations of users paying for better service — Permits differentiated pricing of Internet services

• Based on use of TOS field in packet header (1 byte) — Mostly ignored except on proprietary systems

– Implementation is vendor specific

The Internet Engineering Task Force (IETF) is currently working on a QoS modelcalled "Differentiated Services" or more commonly DiffServ. DiffServ redefinesthe IP Type of Service (ToS) byte into the DiffServ Byte ("DS Byte"). This is usedto signal the required QoS level for a packet. It is also used to identify packets asbelonging to one class or another. DiffServ defines Per-Hop Behaviors (PHBs)which will foster common QoS behaviors in the network. The aim is to provide thebasis for standards-based QoS in a VPN from end-to-end

Note: Some BGP implementations intentionally reset the TOS bits to zero. In allcases, carrying TOS intact is optional.




Notes: Notes:


DiffServ and TOS FieldDiffServ and TOS Field

• RFC 2474 proposes changes to the meaning of the TOS field — First 6 bits are entitled thedifferentiated services codepoint — Last 2 bits are unused

• Eight codepoints allocated for backward compatibility with RFC 791 — xxxxx0 codes are for standard actions — YYY000 calledclass selector codepoints

– Devices using these must offer different queues – Must give priority to YYY=110 and 111 used for routing traffic over 000

• DiffServ nodes allocate nonclass selector codepoints to different services — Have local meaning and need conversion at boundary of the domain

Differentiated services codepoint

By allocated a different value of DSCP to each service, particularly over thecustomer xDSL loop, it will be possible to give precedence to voice services overInternet Web-surfing and downloading.




Notes: Notes:


Weighted Fair Queuing (WFQ)Weighted Fair Queuing (WFQ)

• Each quality-of-service technique establishes a series of router queues

• The router must decide which packet to output next

• We could give high-priority traffic absolute priority — But then low-priority traffic may never get through — Delays to TCP data traffic cause retransmission

– This makes things even worse• Instead, we normally choose to use weighted fair queuing

We could use Priority Queuing (PQ) but that would give priority to voice say at the expense of datatotally. If there was lots of voice we would pass no data at all under some circumstances.

We could give voice the highest priority, but limit the size of the queue to ‘n’ packets. Traffic loadsin excess of that would fall into the default queue. We could also limit high priority queues topackets of a given size, so voice packets (small MTU) and other small stuff (Hello, ICMP, ARP) allgo to the highest queue(s) and other stuff falls later. Finally, data can be selected for queues usingAccess Control Lists. This is all covered in detail in course 481.

WFQ ensures that even with lots of voice some data passes too. We can weight either flows(conversations) or classes of data to ensure that appropriate proportions of capacity are used. If thedata traffic is TCP data, it naturally adjusts its retransmission timers to match the delay through thenetwork and limits the amount of data sent with its window flow control. This means that the datacircuits will eventually back off to match the capacity allocated. The voice on the other hand willprobably continue to be sent at the speed of the talker but will be discarded where congestion occurs.

Note that for LANs, the Cisco default is first in first out. For links at or below 2Mbit/s the default isWFQ.




Notes: Notes:


Forms of WFQForms of WFQ

• There are two forms of WFQ — Flow based — Class based




Notes: Notes:


Forms of WFQ: Flow BasedForms of WFQ: Flow Based

• With flow-based WFQ, packets are classified by flow — Packets for the same full association with the same TOS field belong to the

same flow• Each flow corresponds to a separate output queue

— WFQ allocates an equal share of the bandwidth to each active queue• Flow-based WFQ is also called Fair Queuing (FQ) because all flows are

equally weighted

Flow 1

Flow 2

Flow 3

Imagine a link over which we wish to carry three different services. One, the firstshown here, is a file transfer and no matter how much bandwidth is available thiscould consume it all if allowed to. The second is a very low bit rate application, aping perhaps, sending 1 packet per second. The third is a voice transfer sendingperhaps 50 packets per second. The link over which we might carry these threeservices will have some limit of transfer capacity. On a domestic DSL service theupstream rate may be 256 kbit/s. The voice service would constitute about 72 kbit/swhen the user spoke and so should easily be carried without problem BUT with afile transfer running sending as many large packets as possible it is entirly possiblethat this would normally lock out both other applications.

With flow base WFQ each flow is guaranteed an equal share of the link capacityand so the file transfer can consume as much as available after the other twoservices have bee guaranteed their share.




Notes: Notes:


Forms of WFQ: Class BasedForms of WFQ: Class Based

• Class based — Packets assigned to different queues based on their QoS group or the IP

precedence in the TOS field — QoS group is an internal classification of packets used by the router

– Configured using a class map — TOS-based WFQ classifies packets based only on the 2 low-order IP

precedence bits• Weights can be allocated to each class

— For example, a weight of 50 allocates at least 50 percent of the outgoingbandwidth during busy times

Class base WFQ gives even greater control.




Notes: Notes:


Forms of WFQ: Class BasedForms of WFQ: Class Based

Class 1

Class 2

Class Default

Weight = 50

Weight = 30

The capacity of the output trunk is divided in proportion to the weights which total100. In this case the default calls will always get 100 - 50 - 30 = 20% of thecapacity. Any unused capacity form the first two classes will be diverted to thedefault class.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:


Standards for VoIPStandards for VoIP

• There are two sources of standards for VoIP technology — International Telecommunication Union–Technical Standardization Section

(ITU-T) — Internet Engineering Task Force (IETF)

• ITU-T architecture for multimedia conferencing over packet networks — ITU-T H-series standards define transmission of non-telephone signals — H.323 Most of the currently available products use this — Interfaces easily with world’s telephone networks — Signaling is transferred from ISDN — H.248 for controlling Media Gateways

• IETF SIP: the Session Initiation Protocol — Much simpler to understand and use for simple voice calls — Should be simpler to implement for developers, but…

– Interfacing to ISDN is more difficult — Media Gateways and conferencing provided by MEGACO

– Same as H.248




Notes: Notes:


H.323 RecommendationsH.323 Recommendations

• H.323 Multimedia communications services over Packet Based Networks — H.323 Annexes — H.225.0 (Call Signaling and RAS) — H.245 (Media control) — H.235 (security) — H.341 (SNMP) — H.450 (Supplementary Services) — H.246 (Interworking Gateways) — H.248 Gateway Control protocol (Megaco)

H.323 is an overview recommendation and forms the basis for a suite of protocolsdefined in a family of recommendation.




Notes: Notes:


What Counts as Multimedia?What Counts as Multimedia?

• Voice and audio

• Video

• Conferences

• Mixed conversation

BC B

Voice is 3.1kHz bandwidth while Audio may be greater – even CD quality stereowith a 20kHz bandwidth.




Notes: Notes:


Voice and AudioVoice and Audio

• Audio signals — Encoded using standard codecs — CODer/DECoder converts to

digital — G.711 at 64 kbit/s as default — Others: G.722, G.728, G.729,

MPEG1 audio, and G.723 – Examine these later in the

course• Formatted in digital form

— Described in H.225 and RTP

BB

All voice is audio but not all audio is voice. MPEG-3 and MPEG-4 both provideforms of encoding audio which carriers music and speech very well. By contractG.723 is optimized for speech but carries music rather badly.




Notes: Notes:


VideoVideo

• Video signals — H.261 QCIF as minimum — QCIF: Quarter Common Intermediate Format — Other formats possible: H.263

• May include audio as well

B

Video in the context of H.323 standardization normally implies H.261 or H.263.Better video representation and much more efficient encoding is now availablefrom MPEG-4 and this can also be carried in the same way.




Notes: Notes:


ConferencesConferences

• Point-to-point conference: simple conversation between two terminals — The majority of IP telephony applications

• Multipoint conference — Three or more people

– Some mixing of the signals may be required – Normally only one person speaks at a time

Hi, I amJohn

Robinhere . .

My nameis Jane

Hello: Iam Jan

I am incharge!

A one to one voice call is still considered a conference but just point-to-point. Yourcurrent generation G.711 encoded GSTN voice call takes 64kbit/s in BOTHdirections at the same time or a total of 128kbit/s of network capacity. Most of thetime you either talk or you listen so at least half the capacity is lost. If you listen tomy voice you will notice that even while I am speaking I am not producing soundsall the time. There are gaps . . . between . . . words . . . pauses . . . . . . . . . . . . . . . . .. . . for effect! Throughout a conversation the GSTN is still carrying 64kbit/ ofsilence. But VoIP using H.323 can remove the silence (if implemented correctly).This enables us to carry at least twice the capacity and perhaps more even withoutany change in the way we encode the voice itself.Also at first it would seem that themore people that joined the conference the more capacity would be required.However in practice, in voice conferences there is only one person speaking at atime so the total capacity increase is not as great as with a GSTN conference. Alsowith the right multicast techniques we may find that the resultant mixed signal needbe carried only once over most of its passage back to all receivers.




Notes: Notes:


How Is a Normal Phone Call Connected?How Is a Normal Phone Call Connected?

• Calls start from phones attached — Connected by lines or loops to Central Office (CO) or Local Exchange (LX) — Private branch exchanges (PBX) are owned by user organizations

– Small PBXs are called key systems — Transit Exchanges (TX) join LX to distant TX or CO

Key system

PBX

Fax

Loops or lines

Trunks

CO

TX TX

LX

LX




Notes: Notes:


Encoding the Content of CallsEncoding the Content of Calls

• Voices and multimedia contents of calls carry real time information

• In traditional voice networks timing is maintained

• Packet networks do not guarantee timing

• Media channels are transported with RTP

• Real-time Transport Protocol (RTP) includes sequence and timestamps — Defined in RFC 1889 — Silence can be removed, reducing the bits needed for a call — Receiver can reorder out-of-sequence packets — Smoothes out delay variations called jitter

– Achieved by delaying samples to the speed of the slowest• Feedback is provided to the sender using RTCP

— Receiver sends back quality information on jitter and packet loss

— Also carries identity information




Notes: Notes:


Removing Silence With RTPRemoving Silence With RTP

1 0 0

1 2 0

1 4 0

1 6 0

Time

Silence threshold

Sequence = 1Timestamp=100




With RTP we can reduce the bandwidth by suppressing the transport of silence.When I talk to my wife I will normally use a standards “land-line” which carriedher voice to me as 64 kbit/s of transmission and carries my silence back to her atthe same time in another 64 kbit/s. The current phone network spends half itscapacity carrying silence. High quality silence perhaps, but silence none the less.

With RTP we can carry perfect silence with no capacity at all!




Notes: Notes:


Real-Time Transport Protocol (RTP)Real-Time Transport Protocol (RTP)

• TCP/IP protocol suite includes protocols for real-time applications — Real-Time Transport Protocol (RTP) — Real-Time Control Protocol (RTCP)

• RTP provides — Timestamping, sequence number

– For playback timing and synchronization — Setting up real-time applications

– Audio and video• RTCP provides

— Reporting on achieved results — Delay, packet loss statistics — Receiver report on jitter delay

• Defined in RFC 1889




Notes: Notes:


Real-Time Applications on Packet NetworksReal-Time Applications on Packet Networks

• To be intelligible, our speech must be played out with the same timingrelationship between words as the original — Received packets may not all arrive with exactly the same delay

– This is called jitter• Real-time Transport Protocol marks the voice samples with a timestamp

— That timestamp is used to play out the packet in sequence – With the correct relative time relationship

You’re right This is an IP telephony course

SentReceived




Notes: Notes:


Session Initialization Protocol (SIP)Session Initialization Protocol (SIP)

• Application Layer protocol (RFC 2543) — Create, modify, terminate sessions — Two or more participants

• Multimedia protocol—similar to H.323 — Not just voice — Default “well-known-port” is 5060

• Supports five main functions — User location — Terminal capabilities — User availability — Call setup — Call handling

I have taken this SIP discussion from RFC2543bis Nov 24 2000

We will look at it in overview and see how the protocol is put together. So far Ihave not been able to find any SIP products that work well or even close to as wellas the H.323 classroom demonstrations. This section is theory, smoke andmirrors!!!!!




Notes: Notes:


SIP AddressingSIP Addressing

• User location — SIP address is a URL of the form sip:user@address — Must be a specific host IP and port

• Caller will access the SIP server process in the called device — SIP server is the destination of the initial setup message (invite)

– Server can provide correct destination or relay the request — To locate the SIP server, the calling terminal can use DNS — SIP URL domain name must have SRV, MX, CNAME or Type A DNS

records — These are checked to locate a sip.udp or sip.tcp record

• Directing the call at a server permits the endpoint to move — Endpoint address may change; e.g., DHCP assigned — Increases the stability of the DNS cache

SIP default “well known”port is 5060. This is used to pass signaling messagesbetween servers.




Notes: Notes:


Simple SIP Call SetupSimple SIP Call Setup

• SIP signaling based on requests and responses, called transactions in SIP — Text based (rather than encoded binary messages used in H.323)

– Based on HTTP/1.1 (RFC 2068)• Step 1 is to open a signaling channel with an Invite message

— Sent to the SIP address URL (which may include a port number) — Can use UDP or TCP to well-known port 5060 at user IP address — Invite message contains enough information to immediately establish a

media channel to the caller – Includes addressing and codec capabilities

INVITE: address and codec

Media channel200 OKaddress & codec

Media channel

ACKACK = acknowledgment

This is a simple point to point call with signals passing from client port to 5060 onthe destination.




Notes: Notes:


SIP EntitiesSIP Entities

SIP User Agentpc.work.com SIP User Agentbed.home.net

Location Server

SIP Redirect Server

SIP ProxySIP Proxy(outbound)

SIP RegistrarSIP Registrar

home.netBestneutral.com




Notes: Notes:


SIP ProxySIP Proxy

• SIP allows an organization the opportunity to use a SIP Proxy — This handles the inward and outward transfer of SIP calls — Can undertake address conversion, security and integrity operations

Alice’s SIP phone Bob’s SIP phoneatlanta.com

Proxybiloxi.com

Proxy

INVITEINVITE

INVITE100 Trying100 Trying

180 Ringing180 Ringing

180 Ringing 200 OK200 OK

200 OK

ACK

Media Session

BYE200 OK

An organization might not wish to allow calls to be made and received into and outof its organization in an uncontrolled manner. Also calls made to an individual thatdid not answer could be lost rather than being directed to an attendant or to thelocation a user had moved to.

A SIP proxy allows an organization to overcome these problems. It will relay on

the request for calls in the form of the INVITE but may modify its contents andmap addresses if required. On the inward side it could bar access or redirect thecaller as needed.

A SIP Proxy can also require authentication using user names, passwords orauthentication headers to prevent unauthorized use of the service of gateways orother network resources. It also enables an organization to use simple short formdomain addressing internally and to convert this to the fully qualified Internetdomain at the exit of the local domain.




Notes: Notes:


Megaco Protocol (RFC 3015)Megaco Protocol (RFC 3015)

• Megaco Protocol Version 1.0 in RFC 3015 — Replaces 0.8 in RFC2885

– Previously known as Media Gateway Control Protocol (MGCP)• Purpose of Megaco:

— Control of distributed gateways between IP networks and GSTN• Implements the signaling layers of H.323

— Specifically for voice — Purpose is similar to gatekeeper controlling access to a gateway

– Now common protocol with H.248• Provides for both a media gateway and a signaling gateway

— Interface to the PSTN signaling network – Common Channel Signaling System #7

• Master/slave protocol allowing intelligence to be held within service

MGCP = Media Gateway Control Protocol

MEGACO and H.248 have been developed jointly by a single IETF/ITU combinedgroup. The aim is to provide a means of signaling between media gateways andtheir controllers that can interface to signaling systems in the phone network. Ingeneral this will be SS7 or Q.931.




Notes: Notes:


Purpose of MegacoPurpose of Megaco

• Megaco provides a protocol for control of media gateways — Media gateways could be IP phones, IP PBXs

– Could also be attachments to circuit switched networks• Enables service features to be built as part of network

— Enables simple end systems to be used with limited intelligence — Enables telephone services to be built over IP networks

• Provides mechanisms for building attachments to public SS7 networks






Notes: Notes:


New H.248 DivisionNew H.248 Division

H.248 (Mainbody andAnnexes A to E)

H.248.1 Gateway control protocol Version 2

H.248, Annex F H.248.2 Facsimile, text conversation and call discriminationpackages

H.248, Annex G H.248.3 User interface elements and action packages

H.248, Annex H H.248.4 Transport over SCTP

H.248, Annex I H.248.5 Transport over ATM

H.248, Annex J H.248.6 Dynamic tone definition package

H.248, Annex K H.248.7 Generic announcement package

H.248, Annex L H.248.8 Error codes and service change reason description

H.248, Annex M.1 H.248.9 Advanced media server packages

H.248, AnnexM.2

H.248.10 Media gateway resource congestionhandling package

H.248, Annex M.3 H.248.11 Media gateway overload control package

H.248, Annex M.4 H.248.12 H.248 packages for H.323 and H.324 interworking

H.248, Annex M.5 H.248.13 Quality alert ceasing package

H.248, Annex M.6 H.248.14 Inactivity timer packageH.248, Annex N H.248.15 SDP H.248 package




Notes: Notes:


EnterprisePSTN

Access

WirelessAccess

TrunkingGateway

SS7Network

IP/ATM

GatewayController

GatewayController

MediaGateway

NEWDOMAIN

SignalingGateway

SS7SIGTRAN/TALI/Q.2111

Q-BICC/SIP-T

MEGACO/H.248

SS7SS7

ASP

MEGACO/H.248

RTP/RTCP

Next Generation Network: Soft Switch ModelNext Generation Network: Soft Switch Model

Next generation switches will be soft and will depend upon a family of protocols.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary






Notes: Notes:


IPTV Applications and there NeedsIPTV Applications and there Needs

• What is IPTV?

• Web TV

— Video viewed from the Internet live or stored on a server — Access made through a web browser interface — Needs web access and core capacity at speed compatible with service

• Video played on Demand over IP network on a PC or viewer — Each user is typically a stream of packets sent by the server — User interface generally allows pause/rewind/forward like video recorder — Can be used for premium movie services — Needs web access and core capacity at speed compatible with service

• Multicast broadcast TV played over the Internet or private IP network — Many viewers watching the same single stream at the same time — May or may not be able to store and replay locally

— Core capacity for all channels being viewed — Access capacity for number of channels viewed in parallel

There is a big debate in the industry about exactly what is and what is not IPTV.Web TV is video and TV programs accessible though web browser interfaces.Normaly this is not multicast and may not even be live TV.

Some call VoD IPTV while other people suggest that this is not television but aninternet version of a video player.

To deliver live TV, particularly quality HDTV over IP a very well engineered and

managed network is necessary. If the network is shared with Internet access qualityof service protocols need to be run on the routers and switches to give preferencesto the TV signals over other Internet services.




Notes: Notes:


Encoding and Trans-codingEncoding and Trans-coding

• An important part of the Headend function is encoding TV signals

• Feeds may arrive in one satellite modulation format and be re-coded to

another for more efficient onward transmission• NTSC feeds may be converted to PAL

• Encoding of analogue to MPEG-2 or even MPEG-4 may be required

• The selection of the vendor for headend equipment is often based uponthe quality of such codecs and trans-coding

The Integrated Receiver Transcoder (IRT) receives a modulated QPSK carrier andtranscodes it into a more bandwidth efficient 64 QAM format. The unit accepts LBand input from a satellite down-link converter and produces a signal appropriateto transmission in a 6 MHz television RF channel.

The IRT decrypts and performs Forward Error Correction (FEC) on the incomingsatellite data stream. It then clears information streams not intended for local cable

transmission and inserts new information streams for this purpose. It prepares thesignal for transmission across the terrestrial cable system by re-encryptingprograms under local headend control. IRTs are linked via an Ethernet connectionin a local headend LAN.

The IRT provides local generation and insertion of program specific data, includingtier level, purchaseability, price and rating codes. The unit can also be controlledvia a management system. IRTs may be optionally daisy chainedtogether via themultidrop port and controlled remotely over the satellite link where no Ethernetconnectivity exists. The IRT also provides an expansion interface port so thatexternal devices can be cascaded to allow for processing beyond the capacity of asingle IRT.




Notes: Notes:


Control SystemsControl Systems

• Headend equipment must be controlled

• Older systems use illuminated buttons

• Latest units based on Windows PCs — Easy-to-use graphical user interfaces to configure equipment — Control conditional access and MPEG encoding rates — Broadcast equipment and receivers — Easy ‘drag and drop’ grouping feature for your receiver base — Graphical user interface to schedule receiver control and conditional access

parameters on an — Immediate, one time, daily or weekly basis — On-line help — Password protection on user interfaces — Full redundancy and back-up options — Remote access of head-end control station

European companies currently lead the world in TV control systems. TANDBERGhas a complete range of management system for both small and large MPEG-2broadcast head-ends for configuration, system monitoring and redundancy. Ideallysuited to controlling and monitoring satellite, cable and terrestrial super head-ends,especially where n+m multiplexing is required to save costs. Powerful re-multiplexing capabilities make it perfect for digital turnaround applications. Costeffective device only control is available for the simpler regional head-end.These have recently been installed in the largest cable systems in the world andcontinue to dominate the control of state of the art headend control.

The latest generation systems introduced in 2005 have the capability to work usingall IP services. While the channel and studio side has been IP enabled on manysystems for a year or so, now even distribution can be based on IP. The first All IPsystem deploying MP4 encoding for HDTV was installed in Europe during 2004.This is likely to spread throughout the whole industry over the next few years.




Notes: Notes:


Program DistributionProgram Distribution

• Switches switch Ethernet Frames and/or IP packets in hardware — By using hardware they are faster than routing in software

• Routers route IP packets based upon IP addresses — Able to direct traffic towards the exact destination

IP DistributionNetwork

Primary Streamer

Backup Streamer

Layer 2Switches

MulticastRouters

Control station uses SNMP to switch streamersif necessary

Streamers stream from same IP addressRouters use VRRP/HSRP to guarantee routerreachability

Were a network is to be used for distributing commercial television servicesincluding advertising relibility of service is critical. TV industry standards requireservices that can automatically recover from any failure within 10 seconds or less.Where paid-for advertising is in use penalties may result from an interruption ofservice of only 3 seconds. Early designs of set-top services map TV channels on tomulticast groups at Layer 3 with fixed source addresses. This requires that alternatesources are available within the distribution network which will take on the same IPsource address in the event of a primary failure. Network service detectors can beused to verify that channels are received correctly down-stream and the servicemanagement system deployed to use SNMP for switching network componentswhen failures occur.




Notes: Notes:


ResilienceResilience

• When designing TV distribution reliability is key

• Industry standard requires switch-over on any failure within 10 seconds

• Penalties can result if advertisements are interrupted by even 3 seconds

• Services may use downstream detectors to verify channel reception

IP DistributionNetwork

Primary Streamer

Backup Streamer

Layer 2Switches

MulticastRouters

Control station uses SNMP to switch streamersif necessary

Streamers stream from same IP addressRouters use VRRP/HSRP to guarantee routerreachability

Were a network is to be used for distributing commercial television servicesincluding advertising relibility of service is critical. TV industry standards requireservices that can automatically recover from any failure within 10 seconds or less.Where paid-for advertising is in use penalties may result from an interruption ofservice of only 3 seconds. Early designs of set-top services map TV channels on tomulticast groups at Layer 3 with fixed source addresses. This requires that alternatesources are available within the distribution network which will take on the same IPsource address in the event of a primary failure. Network service detectors can beused to verify that channels are received correctly down-stream and the servicemanagement system deployed to use SNMP for switching network componentswhen failures occur.




Notes: Notes:


Satellite AccessSatellite Access

• Commercial channelsdistributed by satellite

• Decoders deliverservice that may betranscoded to matchdistribution standards

• Operators may usetelecommunicationscarriers service onagency basis

• Time-slip TV producedby storage on serverfarm near downlink

Core Internet

IPTV Head End

VoDServerDecoders and

Transcoders

Streamers

Access

Managementand

Controlsystem

Because so many TV stations exist in the USA and originate content there, mostcommercial cable and IPTV networks need to take the programming from thesesources. The primary means of distribution is still satellite although as highbandwidth broadband telecommunications networks with fiber optic cores becomeavailable across the world this is expected eventually to change. For now a key partof an IPTV head end is satellite feeds.




Notes: Notes:


Core NetworkCore Network

• Core network is highbandwidth

• Must deliver highreliability over longrange

• Multiple QoS

• May interface to theInternet — Must use BGP

routing at theinterface to theInternet

— Often MPLS forspeed inside

Core Internet

IPTV Head End

VoDServerDecoders and

Transcoders

Streamers

Access

Managementand

Controlsystem

Any IPTV operator that wishes to deliver consistent quality service to theircustomer needs to engineer the core network that carries traffic to within a fewkilometres of each subscriber with care. Indeed the closer to the customer we canreach with high speed fiber core switches, the better will be the services.




Notes: Notes:


Video On DemandVideo On Demand

• Take-up of video on demand services is key to design of network

• The greater the take-up the shorter the distance needed to the user

• Possible Designs

High Take-up SystemLow Take-up System

Access

Core

The best way to design the delivery of VoD depends upon take-up. Where strongbroadcast/multicast services are available then take-up will be low and a lowcapacity single system serving many access racks can be an acceptable answer. Thekey disadvantage with this kind of access is that all services may need to passacross the core and so cause major loading problems. Where high usage usexpected a VoD server located in the distribution network serving those usersclosely coupled in the same rack offers a distributed solution that can be scaled to alarger size. The problem with the distributed solution is then delivering to eachVoD server the library of content for access. Physical media transfer (a man in avan) is still the lowest cost solution for moving content and is well suited to moves.Time-slip TV on the other hand might be better delivered in two stages. Localdistributed servers can be configured to access and store locally programs whenfirst requested. By dividing programming into content of 1 hour or less – typical TVprogramming – files for transfer of even HDTV quality rarely exceed 2 Gbytes.Over a core supporting 1 Gbit/s this will take less than 15 seconds.Take-up of VoD services will be reduced where the viewer has access to localstorage of programs. The use of Tivo systems with hard drives in the set-top boxesallows the intelligent recording of broadcast content and then reduces the demandfor time-slip viewing perticularly where this is charged at a premium rate.




Notes: Notes:


Internet Access ServicesInternet Access Services

• Interconnection to Internet designed for required bandwidth and reliability

• Typically dual homes connection

• QoS services might be available for external services — Must be available for internal voice and video

Core Internet

To deliver Internet access the user needs either a PC connection to the access or aset-top service delivering access. Most new IPTV systems provide set-top accessthrough the TV and with increasing deployment of wide screen HDTV thisbecomes more usable – perhaps as good as a PC.

The core network must be connected to the Internet and users expect normalInternet services such as Email, Web page storage, Domain name storage and news

services.




Notes: Notes:


Triple Play NetworkTriple Play Network

• Voice support must be provisioned with required capacity needs — Call controller call rates must be sized and supported — Resilience of call server might be an issue and must be considered — QoS over the access will probably be required

– Perhaps with DiffServ and/or VLAN for voice — Interconnection to external SS7 gateway services must be considered — Gateway to International ISDN services may be required

Core InternetISDN

Call Servers

GW

ResidentialGateway

Adding VoIP support results is little increase in bit rate demand but major increasesin revenue. However to deliver good quality voice it may be necessary to ensureQoS over at least the access. Making a phone call while downloading big files overthe Internet access will be a problem without it. This requires QoS support in thecustomer loom termination (typically a domestic DSL router) and matching servicein the access concentrator.




Notes: Notes:




xDSL Technologies


Core Technologies

QoS

Voice Services


Chapter Summary




Notes: Notes:




• Identify the key technologies that will form the foundation of 21CN

• Compare Access options

• Consider VLAN implementation using IEEE 802.1q

• Expose the advantages of MPLS switched core

• Describe how voice will be carried over the IP infrastructure

• Describe how QoS can be delivered for multimedia services

• Examine new applications that will lead customer demand for 21CN service




Notes: Notes:

The Customer Home NetworkThe Customer Home NetworkThe Customer Home Network

Chapter 10Chapter 10




Notes: Notes:




• Identify the functions and construction of IPTV set top boxes

• Appreciate how Next Generation home interfaces will function

• Describe home interfacing to Triple-Play networks




Notes: Notes:


Structure of home media interface

Next Generation Set-top Box

Home Interface to Triple Play Networks

Protected Broadcast Architecture

Encryption and Authentication Systems

Watermarking

Digital Rights Management

Chapter Summary

Next Generation and Future TechnologyNext Generation and Future Technology




Notes: Notes:


Multimedia Home Platform ContextMultimedia Home Platform Context

At its simplest level, the MHP is set in the following context. The software of theMHP has access to flows of streams and data, and may write some data to storage.The platform may be able to route streams and data outwards to a sink or store.




Notes: Notes:


Multimedia Home PlatformMultimedia Home Platform

The platform will receive inputs from Viewer input devices and outputcommunications through a screen or other outputs like loudspeakers to present tothe viewer. The platform may have access to communications with remote entities.

The diagram shows a possible set of external interfaces between an MHP and theoutside world. This is one example only but it serves to illustrate a series ofprinciples.

The resources of the MHP, accessible by an application, may be contained in aseries of different but connected physical

entities.

The local cluster may connect a number of MHP terminals and resources.

A cluster may also include resources which are not part of the MHP infrastructureand are not available to the application.

The local cluster is understood to be consistent with the DVB IHDN specification.The detailed description of the MHP in the local cluster is not in the .first version ofthe specification.




Notes: Notes:


Basic MHP ArchitectureBasic MHP Architecture

The Architecture describes how the MHP software elements are organized.The MHP model considers 3 layersResourcesThe hardware entities in the platform include a number of functions. They are represented by hardware orsoftware resources. There is no assumption about how they are grouped. The model considers that there can bemore than one hardware entity in the total Platform.From an abstract point of view it makes no difference if the logical resources are mapped into one or severalhardware entities. What is important is that resources are provided to the MHP transparently. An applicationshould be able to access all locally connected resources as if they were elements of a single entity.System softwareApplications will not directly address resources. The system software brings an abstract view of such resources.This middle layer isolates the application from the hardware, enabling portability of the application.The implementations of the Resources and System software are not specified in this document.Application Manager

The system software includes an application management function, which is responsible for managing thelifecycle of all applications, including Interoperable ones.ApplicationApplications implement interactive services as software running in one or more hardware entities. The interfacefor MHP applications is a top view from application to the system software.The API lies between the Applications and the System Software seen from the perspective of an application.




Notes: Notes:


Interfaces Between an MHP Application and the MHP SystemInterfaces Between an MHP Application and the MHP System

Applications use the API to access the actual resources of the receiver, including:databases, streamed media decoders,

static content decoders and communications. These resources are functional entitiesof the receiver and may be finally mapped onto the hardware of the receiver insome manner.




Notes: Notes:


Plug-insPlug-ins

A "plug-in" is a set of functionality that can be added to a generic platform in orderto provide interpretation of application and content formats not specified by thisspecification to be included in MHP terminals.

NOTE: Those organisations concerned with interoperation between the standardMHP platform and other platforms need to specify the plug-in properly for suchplatforms.

The choice of which plug-ins to use must be in the hands of the end-user in orderthat he can have a choice of sources of service. This option can be exercised in anumber of ways, including the purchase of equipment with "built-in" plug-infunctionality, the positive selection of a download, or the automatic selection of adownload where there is no memory resource limitation.

The plug-in may stay resident where the design of the platform implementationallows. The MHP including the plug-in

must behave, once the plug-in is loaded and operational, in the same way as aplatform supporting the format of the

delegated applications without the use of a plug-in.




Notes: Notes:


Block Diagram of ETSI Standard Interface BoxBlock Diagram of ETSI Standard Interface Box




Notes: Notes:


Broadcast Channel Protocol StackBroadcast Channel Protocol Stack

Except in the case of MPEG-2 sections when an MHP application attempts toaccess conditional access scrambled data through one of these broadcast channelprotocols, the MHP terminal shall attempt to

initiate descrambling of this data without the application needing to explicitly askfor it. Attempts to access conditional access scrambled data at the level of MPEG-2sections shall not happen without the application explicitly asking for this.




Notes: Notes:


Interaction Channel Protocols StackInteraction Channel Protocols Stack

Th Interaction channel provides programme selection and control of servicesprovided. The set of DVB defined interaction channel protocols that are accessibleby MHP applications in some

or all profiles are based on world wide web Internet protocols.

The UNO-RPC consists of the Internet Inter-ORB Protocol (IIOP) as specified inCORBA/IIOP.

Applications are likely to be deployed using Java virtual machines in order tomaintain hardware independence




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


Next Generation Set Top BoxNext Generation Set Top Box

The architecture now mirrors PC structures. Notice the interface at the bottom forEthernet. This allows IPTV access to LAN interfaces. Also the TV out can be forHigh definition plasma screens or projection systems. The key aspect that is stillundecided is how much hard coded software will be built into the silicon. Will forexample MPEG decoding logic be hard coded or will this be held in flash ROM.Also on the left side is an interface to a hard disk drive. As the years go by thecapacity of hard disks goes up but the base price does not change. We might expect200 or even 500 Gbytes for about $50. As time goes by the capacity may increasebut the price is not likely to vary much. Will the market stand a set-top box that isas expensive as a low end PC?




Notes: Notes:


Next Generation Set-Top BoxNext Generation Set-Top Box

• Convergence of computing, communications and consumer electronics.

• Software codecs for Windows Media* Video 9

• MPEG-1, MPEG-2 and MPEG-4 compression formats

• Ultra Low Voltage Processor 800 MHz delivers scalable processing

• Ethernet controller providing integrated network connectivity — 10BASE-T and 100BASE-TX physical layer capabilities

• The next generation media centres will have high quality outputs — Surround sound using Dolby 5.1 audio — HDTV digital outputs

The next generation will provide high quality outputs for display and soundtechnology as well as common decoding for digital video standards from what eversource is used. Convergence of cable, over the air and Internet delivered TV willbe a key feature of the future. Internet delivered television will become a majorchallenge to governments and regulators who in some countries wish to restrictpublic access to some kinds of service.




Notes: Notes:


Software in Set-Top BoxSoftware in Set-Top Box

• Core middleware software building blocks — Advanced compression

encoding for digital videoenabling service providers todistribute video-on-demand

— Middleware for NetworkMedia Processing

— Digital Rights Managementsoftware

— Flash memory over-networkdownload

• Distribution over IP — Provides common transport

across all distributions

The middleware is a set of software tools that can be used by programme makersand delivery systems to provide user functionality with minimal network bandwidthimplications. The ability to upload such service software dynamically into flashmemory will enable the next generation of set top box to grow in functionality overtime.




Notes: Notes:


Internet Television PortalsInternet Television Portals

The fastest growing area is Internet distribution of TV. Most of the early systemsare news or shopping related. However once set top boxes are IP enabled there willnot be any material difference between one distribution and another. The Internetprovides a vary low cost mechanism and allows almost any user to become a TVdistributor at the price of little more than broadband access and a PC.




Notes: Notes:


Web TV From North AmericaWeb TV From North America

IPTV is television over IP, in effect over the Internet. Web TV is access via theWorld Wide Web. In practice these are much the same thing.




Notes: Notes:


Web TV From EuropeWeb TV From Europe




Notes: Notes:


Web TV From EuropeWeb TV From Europe






Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


TV Down the Phone LineTV Down the Phone Line

• Today we deliver digital TV using MPEG-2 over channels of about 5 Mbit/s

• With MPEG-4 we can deliver HDTV with 5.1 Dolby sound and 5 caption

channels in the same bandwidth• We can now deliver this over a phone line carrying 10 Mbit/s

• BT is to spend £10,000,000,000 over 5 years enabling this to every UKhome

• Services are already feasible in parts of Sweden

Internet TV has been talked about since the start of the web as we know it now. Early attempts to doit - the UK's Home Choice started in 1992 - were thwarted by the lack of a fast network.Now that broadband networks are bedding down, and it is becoming essential for millions, the bigtelcos are keen to start shooting video down the line.In the face of competition from cable companies offering net voice calls, they are keen to be the topIPTV dogs.Internet Protocol TV is seen as the future of television, and it sits neatly with its vision of theconnected entertainment experience.Telcos have been wanting to do video for a long time. The challenge has been the broadbandnetwork, and the state of technology up until not so long ago did not add up to a feasible solution.Compression technology was not efficient enough, the net was not good enough. A lot of stars havealigned in the last 18 months to make it a reality.Last year, he said, was all about deal making and partnering up; shaping the IPTV ecosystem.This year, those deals will start to play out and more services will come online.2006 is where it starts ramping up and expanding to other geographies - over time as broadbandbecomes more prevalent in South America, and other parts of Asia, it will expand.What telcos really want to do is to send the "triple-play" of video, voice, and data down one singleline, be it cable or DSL (Digital Subscriber Line).Some are talking about "quadruple play", too, with mobile services added into the mix.It is an emerging new breed of competition for satellite and cable broadcasters and operators.According to technology analysts, TDG Research, there will be 20 million subscribers to IPTVservices in under six years.






Notes: Notes:


Triple Play NetworksTriple Play Networks

• Triple play networks deliver Video, Voice and Data services over the samenetwork — By adding mobility this may also be called Quadruple Play

• Such networks could be wired or wireless — Most currently are wired based upon UTP — Future services could be offered over fiber at even higher speeds

• The attraction to the customer is low cost high quality varied services — Hundreds of channels of TV from anywhere in the world — Better TV definition and quality: Cinema quality in the living room — Information rich Internet access: — Near free interpersonal communications: Voice, text and possible video — Opportunities for new applications and businesses from anywhere

– Interactive gaming around the world – Enabling new business ideas for little initial investment

Delivering what the Internet promised in the 1990s




Notes: Notes:


Convergence Protocol StacksConvergence Protocol Stacks

• To deliver Triple Play or even Quadruple Play we need common flexibleprotocol stacks

•

The protocols must be open and proven to be interoperable• Data will run over IP and so the foundation of all new services is IP

TCP UDP

RTP

IP

HTTP

MPEG

Quadruple play adds mobility to triple play services. By providing the sameservices via mobile phone technologies the same services can be delivered into thehands of the you who are the most enthusiastic users of the new technologies.




Notes: Notes:


MPEG Compression ProtocolsMPEG Compression Protocols



• MPEG-3 abandoned but audio encoding

• MPEG-4 ISO/IEC JTC1/SC29/WG11 N4668

• MPEG-7 ISO/IEC JTC1/SC29/WG11N6828 — Adds descriptions language for multimedia

• MPEG-21 ISO/IEC JTC1/SC29/WG11/N5231 — Adds digital rights management

Moving Picture Experts Group (MPEG) a working group of ISO/IEC in charge ofthe development of standards for coded representation of digital audio and video.Established in 1988, the group has produced MPEG-1 , the standard on which suchproducts as Video CD and MP3 are based, MPEG-2 , the standard on which suchproducts as Digital Television set top boxes and DVD are based, MPEG-4 , thestandard for multimedia for the fixed and mobile web and MPEG-7 , the standardfor description and search of audio and visual content. Work on the new standardMPEG-21 "Multimedia Framework" has started in June 2000. So far a TechnicalReport and two standards have been produced and three more parts of the standardare at different stages of development. Several Calls for Proposals have alreadybeen issued




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


Conditional Access SystemsConditional Access Systems

A conditional access system comprises a combination of scrambling and encryptionto prevent unauthorised reception. Encryption is the process of protecting the secretkeys that are transmitted with a scrambled signal to enable the descrambler to work.The scrambler key, called the control word must, of course, be sent to the receiverin encrypted form as an entitlement control message (ECM). The CA subsystem inthe receiver will decrypt the control word only when authorised to do so; thatauthority is sent to the receiver in the form of an entitlement management message(EMM). This layered approach is fundamental to all proprietry CA systems in usetoday.




Notes: Notes:


Digital Encryption Standard: SimulcryptDigital Encryption Standard: Simulcrypt

Way back in 1988, an attempt was made by France Telecom and others to develop a standard encryption system for europe.The result was Eurocrypt. Unfortunately, in its early manifestations it was not particularly secure and multiplex operators wenttheir own way. Thus, in 1992 when the DVB started their consideration of CA systems, they recognised that the time hadpassed when a single standard could realistically be agreed and settled for the still difficult task of seeking a commonframework within which different systems could exist and compete.They therefore defined an interface structure, the Common Interface, which would allow the set top box to receive signalsfrom several service providers operating different CA systems. The common interface module contains the CA system, ratherthan the STB itself, if necessary allowing multiple modules to be plugged into a single STB. However, there were seriousobjections to the common interface from many CA suppliers on the grounds that the extra cost would be unacceptable so theDVB stopped short of mandating the Common Interface, instead recommending it, along with simulcrypt. The CommonInterface was endorsed by CENELEC in May 1996 and the DTG unanimously adopted its use for digital terrestrialtransmission in the UK at its meeting on 13th May 1996.

Since then the European Commission has required the use of a common interface mechanism for all integrated tv sets(excluding STBs which may employ embedded CA systems) and this is likely to be the eventual outcome - an embedded CAsystem in subsidised STBs and Common Interface slots in all other devices. It should be noted that the Common Interfaceconnector allows plug-in cards for other functions besides CA; for example it is proposed to provide audio description for thevisually impaired using a common interface card.Simulcrypt allows two CA systems to work side by side, transmitting separate entitlement messages to two separate types ofSTU, with different CA systems. It also gives the multiplex provider the opportunity to increase his viewer base bycooperating with other multiplex operators. Technical simulcrypt is the same thing but within a single multiplex, thus givingthe multiplex operator some leverage with the CA suppliers.

The simulcrypt system is shown diagramatically below. Note that it requires cooperation between CA suppliers - somethingwhich does not come naturally! If a viewer wishes to receive services from different providers who do not simulcrypt eachother's ECMs, the only option i s to acquire separate decryption for each CA system. The Common Interface enables amulticrypt environment, allowing an additional CA system to be added as a module. This is not quite the panacea it seems,since it still requires the CA vendor to develop the module, something he is unlikely to be keen on if his best customer doesn'tapprove. In practice, the possibility of multicrypt encourages the parties to conclude a simulcrypt agreement.




Notes: Notes:


Conditional Access IdentificationsConditional Access Identifications

• CA identifiers0x0001 to 0x00FF Standardized systems0x0100 to 0x01FF Canal Plus0x0200 to 0x02FF CCETT0x0300 to 0x03FF Deutsche Telecom 0x0400 to 0x04FF Eurodec0x0500 to 0x05FF France Telecom 0x0600 to 0x06FF Irdeto0x0700 to 0x07FF Jerrold/GI0x0800 to 0x08FF Matra Communication0x0900 to 0x09FF News Datacom 0x0A00 to 0x0AFF Nokia0x0B00 to 0x0BFF Norwegian Telekom 0x0C00 to 0x0CFF NTL0x0D00 to 0x0DFF Philips0x0E00 to 0x0EFF Scientific Atlanta0x0F00 to 0x0FFF Sony0x1000 to 0x10FF Tandberg Television0x1100 to 0x11FF Thomson0x1200 to 0x12FF TV/Com 0x1300 to 0x13FF HPT - Croatian Post and Telecommunications0x1400 to 0x14FF HRT - Croatian Radio and Television

0x1500 to 0x15FF IBM 0x1600 to 0x16FF Nera0x1700 to 0x17FF BetaTechnik




Notes: Notes:


Copy Protection SchemesCopy Protection Schemes

• The Copy Control Information (CCI) communications the conditions underwhich a consumer is authorized to make a copy. — CCI: CGMS + APS + Other

• Copy Generation Management Information (CGMS-A or D) — 0,0 “Copy-free” — 0,1 Undefined (to be used for “No-more-copies”) — 1,0 “Copy-once” — 1,1 “Never-copy”

• Analog Protection System (APS) Trigger Bits — 0,0 Off — 0,1 PSP on; inverted split color burst off — 1,0 PSP on; 2-line inverted split color burst on — 1,1 PSP on; 4-line inverted split color burst on

• Other

One of the most important documents on the future of digital television has recently been approvedby the DVB Project. The work of the Multimedia Home Platform committee, it sets out a migrationpath to an open standards future which will allow the market the freedom to develop wide ranginginnovative products.Up to now, digital television services, although based on DVB standards, have proprietary elementswithin them which make it difficult, for example, to add a satellite 'sidecar' to a terrestrial receiver,or vice versa. Obviously multiplex operators who started services before recent standards emerged,defend their positions and rightly claim that the standards making work, which includes strategiesfor migration and gives them time to deal with legacy boxes without jeopardising their commercialinvestment.The UK Terrestrials have no reason to be smug, however. Although the MHEG-5 API they haveadopted is likely to have a place in the new standard, the fact is that everyone will have to cope withregular upgrades, just as Windows 3.1 moved to Windows 95. The analogy is not complete however,becauseWindows 3.1 users could chose to continue just as they were - in a broadcast situation, usersof older spec machines expect them to continue to work when the broadcaster upgrades,even if theycan't avail themselves of all of the new features.The need for 'backward compatibility' is at the heart of the debate and the DVB Project, based at theEBU headquarters in Geneva, offers the right forum for industry experts to come up with the besttechnical for the commercial requirement. In the UK, the ITC are proposing to add support for theMHP standards as a requirement to licensees. None of this matters very much to the viewer who,today, just wants to watch television but for an industry beginning to come to grips with the issues ofconvergence, some quiet satisfaction is in order that they have managed to get the 'route map' inplace.




Notes: Notes:


Content Protection TechnologiesContent Protection Technologies

• Content protection technologies offer methods prevent unauthorizedaccess (for playback or recording) — May include text, graphics, pictures, audio and video

• Two important technologies — Encryption — Watermarking

• Encryption-based technologies transform copyrighted digital content intounintelligible or unviewable format.

• Watermark-based technologies embed data directly into copyrighteddigital content.

• Hybrid technologies: Combine features from encryption andwatermarking technologies.

As technology progresses so too do the commercial threats from the theft ofintellectual property. Content protection must deliver the ability to restrict accessand to trace breaches in access protection.




Notes: Notes:


Types of PiracyTypes of Piracy

• Commercial piracy : a commercial entity steals content, makes a master,and begins making and selling illegitimate copies — Commercial entities with a manufacturing facility will always be able to get

to a clear bit stream, or simply duplicate a prerecorded content• “Garage” piracy : an individual with smaller resources makes a few

hundred illegitimate copies, and sells or barters them — A “garage” pirate, skilled in engineering, will be able to take apart his

TV/VCR/STB, and probe a PC board for a clear bit stream• “Ant” piracy : an individual wants to make a few copies for his friends,

relatives, or even for his own use. — An “ant” pirate will have very limited resources

Piracy of programming becomes a problem when the intellectual property is of highvalue (or at least high price). It is probably true that any protection system builtcould be circumvented eventually with the expenditure of enough resources. Thecountermeasures must defeat the major threats long enough to allow commercialexploitation to deliver profitable distribution.

Intellectual property values reduce with time very quickly so protection for days or

weeks may be enough for some services. However major films can retain value formany years if protection can be maintained.




Notes: Notes:


Information Security ObjectivesInformation Security Objectives

• Confidentiality : protecting information from unauthorized disclosure — Primary tool: Encryption

• Data integrity : providing assurance that information has not been alteredin an unauthorized way — Primary tool: Hashing

• Authentication : — Message authentication : providing assurance of the identity of the sender

(gives no guarantees of timeliness or uniqueness). — Primary tool: Digital signatures — Entity authentication : providing assures of both the identity of — the sender and his active participation in the protocol. — Primary tool: Challenge-response protocols

• Non-repudiation : preventing a party from denying a previous action. — Primary tool: Trusted third party service




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


Encryption: Types of CipherEncryption: Types of Cipher

• Symmetric key cipher : enciphering and deciphering keys are the same orcan be easily determined from each other.

•

Stream cipher : breaks the message M into successive characters or bitsm1, m2,..., and enciphers each m i with the i th element k i of a key streamK=k 1k2... — Examples : RC4 and SEAL. Stream ciphers can either be symmetric-key or

public-key.• Block cipher : breaks the message M into successive blocks M1, M2,...,

and enciphers each Mi with the same key K. — Examples : DES, FEAL, IDEA, and RC5. Block ciphers can either be

symmetric-key or public-key.• Asymmetric (public) key cipher : enciphering and deciphering keys differ

in such a way that at least one key is computationally infeasible todetermine from the other.

— Examples : RSA, ElGamal, and Merkle-Hellman




Notes: Notes:


Symmetric EncryptionSymmetric Encryption

The key characteristic of symmetric systems is that the key is the same at both ends.For broadcast systems the same key would be held within every user device and soprotection of this is critical to intellectual property. By keeping the key the sameand the algorithm simple to implement speed of operation can be delivered.

Each user could be provided with a different key if each distribution was separatelyencrypted by distributed elements in the network. However this demands massive

processing in the network and probable not yet available in current networks.




Notes: Notes:


Asymmetric EncryptionAsymmetric Encryption

Asymmetric systems generally need longer keys with more complex algorithms butcan be dramatically more secure. Generally different keys must be used in eachdirection but each user can be provided with different keys and so differentidentification. Asymmetric public key systems, where one key and algorithm ispublic while the other secret, allows a very secure mechanism to be used butprocessing power required can be too large for real-time encoding.

Generally public key systems are used to deliver symmetric keys in a securemanner. This allows the secure distribution of session keys lasting a few weeks,days, hours or even just minutes. By regular and frequent key changes, a networkcan protect itself from compromising of a single symmetric key.




Notes: Notes:


Comparison of Encryption CharacteristicsComparison of Encryption Characteristics

Characteristic Symmetric Asymmetric

Speed Faster Slower

Secret information Shared Key Private Key

Key length Shorter Longer

Key Period Shorter Longer

Major Problem Key Distribution Public Key Authentication

Main Use Protection of Content Protection of Keys




Notes: Notes:


Digital SignaturesDigital Signatures

• Digital signature: data string which associates a message with someoriginating entity

•

Digital signature scheme: signature generation algorithm & associatedverification algorithm.

• Two general classes of digital signature schemes: — digital signature schemes with appendix — digital signature schemes with message recovery

• Digital signature scheme with appendix: DS scheme which requires themessage as input to the verification algorithm — Examples: DSA, ElGamal, and Schnorr.

• Digital signature scheme with message recovery: DS scheme which doesnot require a priori knowledge of the message forthe verification algorithm. — Examples: RSA, Rabin, Nyberg-Rueppel.

The Digital Video Broadcasting Project (DVB) is an industry-led consortium ofover 270 broadcasters, manufacturers, network operators, software developers,regulatory bodies and others in over 35 countries committed to designing globalstandards for the global delivery of digital television and data services. Servicesusing DVB standards are available on every continent with more than 110 millionDVB receivers deployed.

The networking of communications and access to content anytime, anywhere arebecoming the guiding principles by which the converging broadcast,telecommunications and IT industries are preparing for the future. It is in thiscontext that the DVB Project has considered how it can use its strengths to build afurther set of specifications and guidelines to support the transition to thisconnected world. This short PowerPoint presentation introduces DVB 3.0, the workprogramme that will take the DVB Project into the next phase of its existence. Seehttp://www.dvb.org/




Notes: Notes:


Public Key InfrastructurePublic Key Infrastructure

• Public-Key Infrastructure (PKI) — Combination of software/hardware products, encryption technologies, and services

that enables enterprises to protect their communications on the Internet or other

types of networks. — Integrates digital certificates, public-key cryptography, and certificate authorities intoa total, enterprise-wide network security architecture.

• A typical PKI encompasses : — Issuance of digital certificates to individual users and servers — End-user enrollment software; integration with corporate certificate directories — Tools for managing, renewing, and revoking certificates

• A typical PKI encompasses : — Issuance of digital certificates to individual users and servers — End-user enrollment software; integration with corporatecertificate directories — Tools for managing, renewing, and revoking certificates

• Certificate authorities are the digital world’s equivalent of passport offices




Notes: Notes:


Public Key CertificatesPublic Key Certificates

• Issued by a Trusted Third Party (TTP) — “data” part : issuer, owner, public key, validity period, etc. — “signature” part: digital signature over the data part.

• X.509 (ITU-T Recommendation & ISO/IEC Standard) — Version — Serial number — Signature algorithm identifier — Issuer name — Validity period — Subject name — Subject’s public key information — Issuer unique identifier (optional) — Subject unique identifier (optional) — Signature

Digital signatures can be produced by encrypting identity information using theprivate key of a crypto system so that any user can confirm the identity using thecorresponding public key to decrypt. These can be turned into digital certificates ofidentity by further signing these using the keys held by a certification authority.

These could be used to identify a customer, or at least the set top box of a customer,that is entitled to a service.




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


Data Hiding: Watermarkeing Data Hiding: Watermarkeing

• Data Hiding : process of embedding data(watermark) into multimedia suchas image, video, and audio — Invisibility: degree of distortion introduced by the watermark — Robustness: resistance against attacks and normal A/V processes.

– noise, filtering, resampling, scaling, rotation, cropping – lossy compression – A-D-A & D-A-D conversions

— Capacity: amount of data that can be represented by the embeddedwatermark

• Typical use of watermarks — Identification of the origin of content distributed copies of the content — Identification of the origin of tracing illegally distributed copies of the content — Disabling unauthorized access to the content




Notes: Notes:


Watermarking SchemeWatermarking Scheme

Watermarking takes the copyright image and modifies it to embed the watermarkso that the transmission source can be identified. A copyright owner can thenidentify the source of an illicit version that is discovered.




Notes: Notes:


Classification of WatermarkingClassification of Watermarking

• Domain Type —Pixel: Pixel values are modified to hold the watermark —Transform: Transform Coefficients are modified to hold watermark

– Discrete Cosine Transform (DCT) – Discrete Wavelet Transform (DWT) – Discrete Fourier Transform (DFT).

• Watermark Type —Pseudo Random Number (PSN) sequence (Mean zero Variance 1)

– Allows the detector to statistically detect the presence – Generated by feeding the generator with a secret seed

—Visual Watermark: The watermark is actually reconstructed and its visualquality is evaluated.

• Information Type —Non-blind: Both the original image and secret keys —Semi-blind: Watermark and Secret Keys

—Blind: Only Secret Keys




Notes: Notes:


Watermark Scaling FactorWatermark Scaling Factor

• The scaling factorcontrols the intensityof the watermark

The scaling factor used to add the watermark will have two impacts. The larger thevalue the easier it is to extract the watermark. The lower the value of the scalingfactor the smaller will be the impact of the watermark on the distributed image butthe harder it will be to recover the watermark.




Notes: Notes:


Discrete Wavelength Transform WatermarkingDiscrete Wavelength Transform Watermarking

Using different transforms of the watermark in different parts of an image the moredifficult it is for an enemy to circumvent the impact of the watermarking.






Notes: Notes:


AttacksAttacks

Typical attacks on defeating watermarks are processing images using software toolsthat change the image in technical ways that do not significantly vary the visualperformance.




Notes: Notes:


Extracted WatermarksExtracted Watermarks

Good watermarking techniques are now capable of defeating such attacks as theseexamples show.




Notes: Notes:


Extracted WatermarksExtracted Watermarks




Notes: Notes:


Pure SVD ExtractionsPure SVD Extractions




Notes: Notes:


Conditional Access (CA) SystemsConditional Access (CA) Systems

• A conditional access (CA) system allows access to services based oncertain conditions: — Payment — Identification — Authorization — Registration

• Service providers — Satellite broadcasters: DirecTV, Dish Network — Terrestrial broadcasters: ABC, CBS, NBC — Cable operators: Time-Warner, AT&T, Comcast

• Services — Subscription — Pay-Per-View — Video-on-Demand — CA vendors: NDS, Canal+, Nagravision

Conditional access systems are the key to successful paid commercial distributionsystems.




Notes: Notes:


CA System ArchitectureCA System Architecture




Notes: Notes:


Digital Rights ManagementDigital Rights Management

• A Digital Rights Management (DRM) system protects, distributes, modifiesand enforces the rights associated with digital content.

•

Primary responsibilities of a DRM system — Secure delivery of content — Prevention of unauthorized access — Enforcement of usage rules — Monitoring of the use of content

• Superdistribution: a relatively new concept for redistributing contentacross the Internet — Allows the customers to forward encrypted content to other — customers — The content forwarded to a potential buyer cannot be accessed — unless the new rights are obtained — May help widen the market penetration

• DRM system vendors: Intertrust, Microsoft, IBM




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:


Digital Rights Management (DRM) System ArchitectureDigital Rights Management (DRM) System Architecture




Notes: Notes:


Interoperability of DRM SystemsInteroperability of DRM Systems

• Today there are many standards that ensure the interoperability ofconsumer electronics devices. A consumer may buy a Sony TV set andconnect it to a RCA DVD player

• Interoperability is also essential for content protection systems likeContent Scramble System (CSS) for DVD player .

• Unfortunately, a client device supporting the DRM system X can onlydownload content protected by the same system

• Currently, there is no interoperability among the DRM systems for anumber of important reasons: — Every DRM system has secret keys/algorithms. DRM vendors are

concerned about sharing secret keys/algorithms — Metadata is data about data that describes the content, quality, condition,

and other characteristics of data. Although Rights expression languages(RELs) are emerging as essential components of DRM systems, they are

not standardized yet




Notes: Notes:


The Complete PackageThe Complete Package

• Conditional Access (CA) — Satellite, cable & terrestrial content — Content is protected in delivery — Consumer has access to content based on a condition — Privately defined system

• Digital Rights Management (DRM) — Primarily Internet content — Content is protected in delivery and storage — Consumer purchases usage rights — Privately defined system

• Copy Protection (CP) — Prevention of illegal copying — Interface protection & storage protection — CA + DRM + CP = Content protection




Notes: Notes:







Watermarking


Chapter Summary





Notes: Notes:



Now you have completed this chapter you will be able to

• Examine methods for content protection

• Appreciate how content can be protected using conditional access

• Compare Conditional Access with Digital Rights Management

• Describe how watermarking of content can be achieved




Notes: Notes:

Industry TrendsIndustry TrendsIndustry Trends

Chapter 11Chapter 11




Notes: Notes:




• Describe the short term future industry changes

• Appreciate the long term trend in the technologies




Notes: Notes:


Switch to HDTVSwitch to HDTV

0

20

40

60

80

100

120

2004 2006 2008 2011

World Cup12 million

BeijingOlympics36 million

Switchover 100 million

million

• Growth of HDTV base upon Japan and Korean experience

Recently published predictions of global HDTV sales shows that there are nowabout 12 million HDTV devices. The sales of these devices are driven by peerpressure and so it is expected that each market will grow in the same manner thatJapanese and Korean markets have grown where these products have been availablefor some time. Sales are often driven by major TV events. The first TV boom inmany countries was driven by the 1953 boom in sales to watch the coronation ofElizabeth II in the UK. The switch to colour TV came in the same way by a seriesof sporting events. Once a new technology becomes established and widelyaccepted the main international exchanges of TV migrate. This has alreadyhappened inside the network for moving programs. This was done in the 1990s viaanalogue recordings on tape then digital recordings on DAT tape. Now exchangesare made in MPEG-2 files and with stereo sound tracks. These files are nowgenerally moved via IP network connections. The next evolution is likely to be themigration to HDTV format using MPEG-4.




Notes: Notes:


IPTV and VODIPTV and VOD

• Commitment to next generation broadband access network is criticalenabler for sufficient quality of service — NTT target of 30m FTTH customers by 2010 in Japan

• Innovation and market development being held back by uncertainregulatory environments

• Demand could be tempered by dual screen environment rather thanconvergence

• Growing market for consumer electronics able to timeshift viewing mayaffect IPTV take up — Sales of DVD HDD recorders reached 5.5m in 2005 — Sony X Video Station to launch this year. A PVR with 8 tuners and 2

terabytes of hard disk memory

VoD services already exist in Japan, Korea and parts of the USA. There are smallpockets around Europe too. Early indications are that take-up is price sensitive asone might expect. Where Time-slip TV is offered this is attractive to viewers andresults in greater usage than premium rate moves. Even within subscribers theusage rarely reached 15% of subscribers at any time. Usage is more dependent uponwhat programming on free-to-air channels was. Where this was strong mostviewers would not invest the time to decide what movie to watch!




Notes: Notes:


Efficient DistributionEfficient Distribution

• Efficient distribution may require the duplication of some services — VoD services are best located near to subscriber — Broadcast channels of recorded video and moves may work best duplicated

To avoid transferring large amounts of data from one side of the network to anotherduplication of servers will be necessary in many networks. Bulk transfer of contentis probably best achieved by man-in-a-van transfer.




Notes: Notes:


Modem and DSL Access SpeedsModem and DSL Access Speeds

• Access speed increases about four times every four years

1980 1985 1990 1995 2000 2005 2010 20150.1

1.0

10.0

100.0

1000.0

10000.0

100000.0

0.31.2

2.4

14.428.8

56512

1000

10000

Access Speed inKbit/s

The current technology limit to high speed services is at the loop. We can alreadybuild core network infrastructures with virtually limitless capacity but the last mile,or at least the last 5 km is still the limitation economically. Part of the limitation isthe historic dependence upon copper loops. If we dug up the streets again andreplaced these with fiber the situation would change but there is not the economic,or in Britain, the political will to do this yet.If we look at technical development of copper based loop technology the speed hasbeen increasing year by year in a predictable way. The upper limit on this is thoughto be a bit less than 10 Mbit/s over 5 km, but perhaps 50 Mbit/s if we drop to 500m.




Notes: Notes:


Home IPTV ProfileHome IPTV Profile

• At the moment access speeds limit IPTV services

• We need 5 Mbit/s for each HDTV channel viewed in parallel

• 2 TV households are the norm

• Access needs to double average useage rate — 20 Mbit/s is target for dual HDTV service




Notes: Notes:


Channel SurfingChannel Surfing

• Channel surfing is a problem to be solved

• Switching channels with MPEG-4 takes several seconds – up to 6

• IGMP traffic could overload access routers with many surfers duringadvertising breaks

• Solution might be variable rate services

64 kbit/s

250 kbit/s

2 Mbit/s

5 Mbit/s

1 sec

3 sec

5 sec

10 sec




Notes: Notes:


Commercial SuccessCommercial Success

• Telephony carriers are losing money as competition drives down voicerevenue — Solution is seen as IPTV to increase revenues of broadband access

• Cable companies see expansion into voice as a way of increasing profits — IPTV delivery gives common network to deliver services

• Digital TV transition delivers better security of content — Content owners see DRM as a way of protecting who plays content and how

• Migration from Analogue to Digital increases channel space — More channels means fewer viewers per channel and lower advertising

revenue — Will we have hundreds of low quality channels nobody wants to watch?

• Is it feasible to deliver user created content?




Notes: Notes:


User Created Media: Shoutcast.comUser Created Media: Shoutcast.com

SHOUTcast is Nullsoft's Free Winamp-based distributed streaming audio system.Thousands of broadcasters around the world are waiting for you to tune in andlisten. Take a peek through the SHOUTcast directory (immediately listed below) tostart browsing the most popular stations. Be sure to select your connection speedand then what kind of music you're looking for over on the right hand side foroptimal listening pleasure. All you need is a player (we recommend Winamp ) andyou're set to go!

Wanna be a broadcaster? It's Free! Check the online docs to get started!




Notes: Notes:


Yospace.comYospace.com

Case Study: MCP Community Gallery Powers 3 UK's See Me TV

Mobile media company 3 was first to utilise the MCP Community Gallery technology to power theirsuccessful "reality TV" consumer service, See Me TV.

The service was launched in mid-October 2005, and as the UK's first ever mobile community videodownload gallery, it has been a phenomenal success.

3 UK's customers can submit video clips by MMS to shortcode 32323. The clips are moderated andaccepted clips are placed into an appropriate category within the handset based gallery. The originalsender is informed of the clip's acceptance by text message, which also invites them into See Me TVif they are a first-time contributor.

Any of 3 UK's 3.2 million customers can use their video mobile to browse, search and downloadclips from the gallery. The clips range in price from 10p to 50p. Customers who have submitted clipsinto the gallery can manage their clips from the service and view how many downloads they've had.Contributors are paid a small share of revenue generated from their submitted content, which is paidin cash on a monthly basis. The repayments are handled by Yospace's integration with PayPal.

See Me TV is hosted and managed by Yospace with integration into 3's messaging, portal and billingsystems.



Notes: Notes:




• Describe the short term future industry changes

• Appreciate the long term trend in the technologies

Documents

IPTV Course