3 Hello, Hello,…..The telephone works by converting acoustic energy into electrical energyIt turns the sound waves of the speaker’s voice into a varying electric current which is sent along a wire and is then turned back into sound wavesAlexander Graham Bell is usually credited with the invention of the telephone which was patented in 1876At least ten men before him had the idea of the telephone and two of them produced a practical telephonePhilip Reis made a telephone in 1863 but did not take out a patentElisha Gray also invented a telephone but was beaten to the patent office by Bell by a few hours
4 Telephony Equipment Telephone sets (analog or digital) Telephone switchesKey systemsPBXsCentral OfficeTandemEtc.Tandem Switch is a telephone central office switch that links telco end offices together and does not connect to the customer directly. Also called a "Class 4 switch" or "TDM switch," a tandem switch is a computer that is specialized for TDM-based, circuit-switched telephone calls. Tandem switches are typically from Lucent and Nortel Networks (see ESS and DMS).
5 Telephone SetA telephone typically consists of the following components:Handset containing a Microphone and SpeakerSwitch hook, which is a lever that is depressed when the handset is resting in its cradleTwo-wire to four-wire converter to provide conversion between the four-wire handset and the two-wire local loopDuplex coil to block the sound of your own voice from leacking backKeypad (either rotary or touch-tone)Ringer
6 Microphone and Speaker Microphone turns the sound energy from voice into electrical energyThe microphone contains a flexible piece of plastic called a diaphragm with an iron coil attached to it and a nearby magnetWhen you speak the sound energy in your voice makes the diaphragm vibrate, moving the coil nearer to or further from the magnetThis generates an electric current in the coil that corresponds to the sound of your voiceIf you talk loud, a big current is generated; if you talk softly, the current is smallerThe loudspeaker in a phone works in the opposite wayIt takes an incoming electrical current and uses magnetism to convert the electrical energy back into sound energy you can hearThe telephone works by converting acoustic energy into electrical energy. It turns the sound waves of the speaker’s voice into a varying electric current which is sent along a wire and is then turned back into sound waves. The telephone consists of a transmitter and a receiver. The transmitter contains a metal box filled with carbon granules. The sound waves from the speakers voice strike the side of the metal box, known as the diaphragm, causing the diaphragm to press against the carbon granules. This pressure varies as the sound waves vary and an electric current passing through the carbon granules is varied in accordance with the changes in the pressure on the carbon granules. The variable current is an electrical copy of the sound waves and is sent along a wire to the receiver. The receiver contains an electro-magnet and a thin disc capable of vibrating. The variable electric current enters the electro-magnet which varies in strength according to the variations in the current it receives. The variations in the strength of the electro-magnet cause the thin disc to vibrate producing sound waves which are a reproduction of the sound waves produced when the speaker talked into the transmitter. This enables the person listening on the receiver to hear what the person speaking into the transmitter has said.
7 Basic Operation A typical telephone does the following: Requests service from the networkPerforms dialing functionsPerforms a notification function (it rings)Provides answer and disconnect supervisionConverts outgoing speech to electrical signals, and vice versaAutomatically adjusts to the supplied power
8 The N2 Consideration For N users to be fully connected directly Requires N(N – 1)/2 connectionsRequires too much space for cablesInefficient & costly since connections not always onN =N(N – 1)/2 = 4,999,950,000
9 Circuit Switching Centralized switching power was needed Patchcord panel switch invented in 1877Operators connect users on demandEstablish circuit to allow electrical current to flow from inlet to outletOnly N connections required to central office (CO)Major switching revolution to follow…N1N – 123
10 Basic Call Progress: On-Hook TelephoneSwitchLocalLoopLocalLoop-48 DC VoltageDC Open CircuitNo Current Flow5
16 Analog Telephony - Signaling The purpose of signaling in a voice network is to establish a connectionFlashing light and ringing devices to alert the called party of incoming callCalled party information to operator to establish callsControl information that implies change of status in the network, along the signaled path (includes call records allocated for the call (timeslots, senders and receivers, memory, processes, etc.)Signals can be impulses, voice band tones, binary signals or messages transported in a packet networkThree forms of signaling:Access SignalingStation Loop SignalingAddress SignalingDial toneBusyRingbackCongestionReorderReceiver off-hook howlerNo such numberSpecial recordings11
17 Access Signaling Determines when a line is off hook or on hook Loop Start (almost all telephones)Seizure is detected when current flows through local loop, due to off-hookGround Start (PBXs)Seizure is detected when one wire is groundedSeizure can be initiated in both directionsE&M, FSX, FXO – all analog type signaling12
18 Loop (Local or Station) Loop StartStationPBX or Central OfficeLoop (Local or Station)+Switch–DC Current+Switch–Supervision =Two states—ON HOOK or OFF HOOKCall Information—Busy, Ringing (ring back), Fast Busy (network availability), othersACRinging+Switch–14
19 Station Loop Signaling When the switch receives the off-hook signal, it responds with an audible signal indicating that it is ready for a call - the dial toneThis two-way exchange between the PBX and the telephone is known as station loop signaling
20 Address SignalingIn response to the audible prompt of the dial tone, the caller can request connection to another telephone by transmitting the address (telephone number) of the requested telephone (sometimes referred to as the called party identification number) to the switchThis is known as address signalingTelephones generally use two basic types of address signaling:PulseToneISDN - Digital Transmission“Out-of-Band”Dial PulseDTMFAnalog Transmission“In-Band” Signaling0–9, *, # (12 Digits)
21 Pulse Dialing (rotary dialing) Rotary dial phones represent the digit being dialed by momentarily stopping the current flow when the user turns the circular dialFor example, the circuit is broken three times, which creates three pulses in the current, to dial the digit “3”In general, the repetition rate of the pulses must be between 8 and 11 pulses per second (pps)An interval between two signals of ms is used to separate signals
23 A little about VoiceSound is a disturbance of mechanical energy that propagates as a waveSound waves, like other waves, are caracterized by:Frequency: Represents the number of periods in a second and is measured in hertz (Hz) or cycles per second Human hearing frequency range: 20Hz to 20kHz (audio)Amplitude: The measure of displacement of the air pressure wave from its meanWavelength: The distance between repeating units of the propagation wave
24 A little about VoiceA voice frequency is one of the frequencies, within part of the audio range, that is used for the transmission of human speechUsable (inteligente) voice frequency ranges from approximately 300 Hz to 3400 Hz and it is called voice bandVoice chanel has a range of 0 – 4kHz. Area between 3.4 and 4 kHz is used for system controlTypical telephone line has frequency response of 300 Hz to 3400 Hz. The signal starts to attenuate in the frequencies below 300 Hz because of the AC coupling of audio signals (audio signal goes trough capacitors and transformers). The absolute, upper theoretical limit (with perfect filters) is therefore 4kHz - but this isn't reality, 3.4 kHz maximum frequency is. The difference is, thus, left there as a guard band (control).
25 Voice Stream Information A real-time voice signal must be digitized and transmitted as it is producedSpeech signal level varies continuously in time (quasi-periodic)Speech is slow-varying signalFrom sample to sample there is a significant amount of correlation!Th i s c ou rse w i ll b e ve r y i n t e re s t i ng I th in kNormal speech consists of talkspurts, which typical last a few hundred milliseconds and silence periods, which occur within a spoke word and between words.
26 Digitization of Voice Analog Signal Why voice digitization?Ensures better qualityProvides higher capacitySupports longer transmission distanceThink as of discrete electrical impulsesDifferent voice digitization techniques existIn this lecture we will talk about Pulse Code ModulationPAM = pulse amplitude modulationPDM = pulse duration modulationPPM = Pulse position modulationPCM = Pulse code modulationADPCM = Adaptive differential PCM
27 Digitization of Voice Analog Signal Analog InputDigital OutputSamplerQuantizerDigital EncoderOriginal signalSample valueApproximationMethodology
28 Pulse Code ModulationPCM is invented by Alec H. Reeves (English inventor) in 1938Interesting fact: PCM bandwidth grater then that required by the original analogue signalTwo principles of digitization: Sampling and QuantizationSample analog signal in time and amplitudeFind closest approximationSampling: Divide the horizontal axis (time) into discrete piecesDiscretizes the continuous timeQuantization: Divide the vertical axis (signal strength - voltage) into pieces. For example, 8-bit quantization divides the vertical axis into 256 levels. 16 bit gives you levels. Lower the quantization, lower the quality of the soundReduces the infinite range of the sampled amplitudes to a finite set of possibilitiesLinear vs. Non-Linear quantization (more to come)It allows perfect signal reconstruction, which compensate for the attenuation providedthat the channel noise level is insufficient to the damage the bit stream
29 Voice Sampling Harry Nyquist (1889-1976) Born in Sweden, moved to USA in 1907, AT&T>>>Bell LabsDr. Nyquist and Dr. Claude Shannon are responsible for virtually all the theoretical advances in modern telecommunications4 kHz assumes conversational speech, not singing or CD-quality audioSampling twice per cycle allows us to reconstruct the signal – which is what we want to do to rebuild the voice signal at the receiving endLimit of human hearing: ~22KHz?CD sample rate: 44,100 samples/sec
30 Sampling Theorem t x(t) t x(nT) t x(nT) t x(t) Perfect reconstruction if sampling rate 1/T > 2 * maximum signal frequency(a)tx(t)tx(nT)Sampler(b)tx(nT)tx(t)Interpolationfiltern – index of the sampleT – sampling periodWhat happens to all those higher frequencies you can’t sample?They add noise to the sampled data at lower frequencies ALIASING NOISE
31 Sampling Lets consider a sign wave to be sampled (e.g. voice signal) If you sample at 1 time per cycle, you can think it is a constantSampling 1.5 times each cycle appears as a low frequency sine signalNyquist Theorem: For Lossless digitization, the sampling rate should be at least twice the maximum frequency responses
32 QuantizationQuantization can be Uniform, Logarithmic, Adoptive, Differential or VectorThe distance between the finite set of amplitude levels is called quantizer step size and is usually represented by delta ΔEach discrete amplitude level xi is represented by a codeword c(n) for transmission purposeThis codeword indicates to the de-quantizer, which is usually at the receiver, which discrete amplitude is to be usedThe main aim of a specific quantizer is to match the input signal characteristics both in terms of its dynamic range and probability density function000001010011100101110111x1x2x3x4x5x6x7x8x9y1y2y3y4y5y6y7y8InputOutputMid-riser quantizer
33 QuantizationHow many bits do we need to accurately represent the range we’re sampling?For the 0 – 4kHz range, 8 bits is enoughIf eight bits are allowed for the PCM sample, this gives a total of 256 possible valuesPCM assigns these 256 possible values as 127 positive and 127 negative encoding levelsThese values translate into binary codes which become the corresponding PCM values
34 Quantization Error Each sample is a digitized estimate if the analog value is 5.3, the closest we can come is 5all we’ll get on the other end is 5the 0.3 is lost foreverSomething is lost in the sampling: Quantization Noise or Quantization Erroroutput y(nT)input x(nT)Quantization error (noise):e(n) = x(nT) – y(nT) -Δ/2 ≤ e(n) ≤ Δ/2Statistically quantization noise is stationary and is unrelated to the input signal. It is also considered uniformly distributed.
35 Uniform QuantizationAll of the quantizer intervals are of the same widthIt can be defined by two parametersNumber of quantizer levels; Nql = 2BQuantizer step size; Δ = Yi – Yi-1Total mean square error for uniform quantizer:Signal to Noise Ratio (SNR) formula is useful to determine number of bits needed in quantizer for the certain signal to quantization noise ratio and in estimating the performance of u uniform quantizer for a given bit rateSQNR = 6.02B dBThe SQNR increases approximately 6dB for each bit
36 Quantization (liner vs. logarithmic) Logarithmic QuantizationLinear Quantization (uniform)The 127 quantization levels are spread evenly over the voice signal’s dynamic rangeThis gives loud voice signals the same degree of resolution (same step size) as soft voice signalsEncoding an analog signal in this manner, while conceptually simplistic, does not give optimized fidelity in the reconstruction of human voiceMost of the energy in human voice is concentrated in the lower end of voice’s dynamic range (no shouting – just from the boss)Quantization levels distributed according to a logarithmic, instead of linear, function gives finer resolution, or smaller quantization steps, at lower signal amplitudes8-bit PCM in North America uses a logarithmic function called μ-law and in Europe A-lawCompression techniques can reduce the amount of data needed torepresent a digital signal without sacrificing too much quality. A simpletype of compression employs nonlinear amplitude steps. The basicdigital encoding standard used for telephone voice, G.711, usesamplitude quantization defined by one of two logarithmic functions, theA-law and the µ-law. The logarithmic steps allow an eight-bitquantization to encode a dynamic range similar to that for a twelve-bitlinear quantization.16 bit linear would be rather expensive for telephony.8 bit linear poor quality.Solution: use 8 bits with an “logarithmic” encoding.-Known as companding (compressing/expanding)-Goal is that quantization noise is a fixed proportion ofthe signal, irrespective of whether the signal is quiet orloud.
37 μ-law and A-lawIn both schemas (mu-law, a-law), the signal to quantization noise performance can be very close to that of uniform quantizerQuntizer levels are closely spaced for small amplitudes which progressively increase as the input signal range increasesBoth give similar quality to 12-bit linear encoding
38 Bit Rate of Digitized Signal Assuming all of the discrete amplitude values in the quantizer are represented by the same number of bits B and the sampling frequency is fs, the channel transmission bit rate is given byTc= Bfs bits/secondBandwidth in Hertz: how fast the signal changesHigher bandwidth → more frequent samplesMinimum sampling frequency = 2 * bandwidthGiven the fixed sampling frequency the only way to reduce the channel bit rate Tc is by reducing the length of the codeword c(n)This creates bigger difference between the analogue and discrete amplitudes which reduces the quality of reconstructed signalVarious types of scalar qantizers are used in order to reduce bit rate while maintaining a good speech qualityRepresentation accuracy: range of approximation error- Higher accuracy:Smaller spacing between approximation valuesMore bits per sample
40 CODEC – Coder/DecoderAfter being sampled and quantized signal is coded for transmissionCODEC is a program/system that encodes and decodes digital data signalAudio codec refers to the device encoding an analog audio signal to a digital audio signal, or decoding an analog audio signal from a digital audio signalThere are many variations of Audio/Voice codes as we will see in the upcoming lecture
41 Digital Signal Transmission – Multiplexing Multiplexing involves the sharing of a transmission channel by several connections or information flowsChannel = 1 wire, 1 optical fiber, or 1 frequency bandSignificant economies of scale can be achieved by combining many signals into oneFewer wires/pole; a fiber replaces thousands of cablesBCAMUXShared Channel
42 Time-Division Multiplexing - Telephone digital transmission High-speed digital channel divided into time slotstA1A23T0T6T…B1B2C1C2Each signal transmits 1 unit every 3T secondsCombined signal transmits 1 unit every T secondsB1C1A2C2B2A1t0T 1T 2T 3T 4T 5T 6T…
43 T-Carrier System Digital telephone system uses TDM PCM voice channel is basic unit for TDM1 channel = 8 bits/sample x 8000 samples/sec = 64 kbpsT-1 carrier carries Digital Signal 1 (DS-1) that combines 24 voice channels into a digital stream2241b. . .2322FrameMUXFraming bitBit Rate = 8000 frames/sec. x (1 + 8 x 24) bits/frame = Mbps
44 Voice TransmissionPCM yields to the net composite bandwidth for transmission of 64,000 bits per second or 64kbs (8000 samples/second x 8 bits/sample). This is the standard transmission rate for one channel of telephone digital communicationsFor a voice call, this level is generally considered to be "toll quality“, or the quality of voice that one would anticipate receiving from a commercial telephone carrierVoice TDM circuit-based transport :T1/E1DS3/E3SONET (OC-3, OC-12, etc.)Voice packet-based transport :ATMFrame RelayIPMain Topic
45 Before we continue - Terminology PSTN: Public Switch Telephone NetworkPOTS: Plain Old Telephone ServiceLEC: Local Exchange CarrierCLEC: Competitive Local Exchange CarrierILEC: Incumbent Local Exchange CarrierLATA: Local Access and Transport AreaIXC: Inter-Exchange CarrierPBX: Private Branch eXchangeBandwidth: Line capacity (in KHz or Kbits/s)Voice Circuit: The bandwidth used by a voice communicationIN Services (or Class Services): 800 Number, LNP (Local Number Portability), Call Forward, etc.
46 Legacy Telephony Architectures The Public Switched Telephony Network (PSTN) is a global circuit-switched network that was designed primarily for voice trafficEnterprise Telephony is a business telephone system that provides basic business features (call-hold, three-way calling, etc.In common:Circuit switching based on TDMCommon Infrastructure Model (call control, bearer channel, local loop (phones connect directly into a switch)Services they offerDifference:# of local loops – pstn switch >100,000, pbx switch > 5,000The way they treat signaling- pbx (proprietary) but it uses CAS and PRI as interface to pstn, pstn uses SS7, ISDN and in-band signaling linksChannel Associated Signaling (CAS), one of two types of telecom signaling technologies, is the transmission of signaling information within the information band, or in-band signaling. This means that voice signals travel on the same circuits as line status, address, and alerting signals. As there are twenty-four DS0 channels on a full T1 line, CAS interleaves signaling packets within voice packets. Various types of CAS signaling are available in the T1 world. The most common forms of CAS signaling are loopstart, groundstart, and E&M signaling. The biggest disadvantage of CAS signaling is that the network uses bits from information IP packets, such as voice packets, to perform signaling functions.
47 PSTN Overview - History Graham Bell connected 2 rudimentary phones (Carbon Membrane, Battery and a Magnet ) with an electrical cord in this was a direct connection between 2 phones with no dialingDesign evolved from one-way to a bi-directional voice transmissionFirst improvement: connect every phone to an operator to switch callsSecond improvement: Dialing and Mechanical Switches (later on: Electronic Switches)
48 PSTN Overview – Architecture A pair of copper wires (Local Loop) runs between a subscriber home and a local Central Office (CO or Class 5 switch)Exchange – second name for a voice switchCOs connect to their local Tandem Switch (or Class 4 Switch)Local Tandem Switches connect to higher Layer Tandem SwitchesSwitches connect through TrunksSame portions of the PSTN use as many as five levels of hierarchyTraffic categoriesUpstream or downstreamIncoming, outgoing, internal, terminating, originating, transitService Nodes – connected to the edge of the telecom networkVoice mail systemsVoice response systemsAnnouncement devicesService Control Points (SCPs)Space for competitionService nodes can be used to control call setup
49 PSTN Hierarchical Layout Trunk is the link between switchesUsually copper cables between local switchesOptical Fiber between higher level switchesTrunks carry digital voiceLocal Loop is the link between a subscriber and the Central OfficeVoice stream is usually analogThe capacity (or bandwidth) of the line is limited to 4KHz (64KBits/s)ISDN (digital voice, out of band signaling) did not really catch upClass 4 SwitchClass 4 SwitchTrunkTrunkTrunkClass 4 SwitchAreaTrunkTrunkCO/Class 5 SwitchCO/Class 5 SwitchLocal LoopLocal Loop
50 Typical PSTN Hierarchy International AreaClass 1Class 2Tertiary AreaSecondary AreaClass 3Primary AreaClass 4Local AreaClass 5End-to-end connection can have max of 12 circuits – addressing is hierarchicalSource: Computer Networks, Andrew S. Tanenbaum
51 PSTN Service Model Permanent circuit is set up on demand Transfer capacity can be used as best or as poorlyCustomer pays based on used network resourcesUsage-based billingTime-based billingProviders attach some other fees (sometimes explainable/sometimes not)
52 Signaling ConceptsSignaling is the generation, transmission, and reception of information needed to direct and control the setup and disconnect of a callTwo groups of signaling methodsUser-to-network signaling – end user/PSTN signaling – Pulse, Dual Tone Multi Frequency, ISDN (BRI and PRI)Network-to-network signaling (trunk signaling) – intercommunication between PSTN switchesSignaling: on hook, off hook, digits collectionIn-band SignalingOut of Band Signaling
53 PSTN Signaling Network-to-network/Signaling between trunks: Old Switches: in-band is not flexible, hard to deploy IN servicesModern Switch use out of band Signaling. SS7 is the most popularSS7:Very flexibleSaves bandwidth for voiceDelegates IN Services to special nodes in the network (SCPs)
54 Call RoutingNumber analysis links the information received from signaling to call routing (dialed digits, incoming circuit info)Analysis returns a set of routing alternatives or an instruction to perform number translation (800, 866, etc. numbers)Primary Route AlternativeSecond Route AlternativeRoute 1Route 2Final Route AlternativeRoute 3Trunk Group*Seizure algorithms search and reserve free circuits or trunks!**If the end of the tree is reached, the call is blocked
55 PSTN Players Service Providers: Local Services: LECs (Local Exchange Carriers). Baby Bells IN services: Caller Id, Call Blocking, Call Waiting Call Back, Last calls, Call ForwardingLong Distance: IXCs (Inter-Exchange Carriers). Sprint, ATT, MCI, etc. IN Services: 800/888/866 Numbers, Calling CardsEquipment Vendors: Ericsson, Nortel, Alcatel, Lucent
56 PSTN Legacy - Carrier Grade When was the last time you couldn’t get dial tone in your home?Occasional “backhoe fade”, but it almost always worksCompare that to your cable service, or network access at workOver 100 years of engineering goes into the phone network, and things like 911 have become a critical serviceData networking is much less matureComputing has barely been around since ~1950Serious data networking? ~1960s
57 PSTN Legacy - Carrier Grade (2) HI RELIABILITY99.999% - “5 Nines”Becoming a requirement on data networking vendors who want to sell equipment to the telco world SCALABILITYSupport 10s/100s of millions of end-usersSupport 100s of thousands of simultaneous callsNote the dependence on statisticswe expect that not everyone will want to talk on the phone at the same timewe also expect that the calls will be relatively shortData modems started to violate these assumptions
58 PSTN Legacy - Carrier Grade (3) EFFECTIVENESSTo quote James Earl Jones, “The network’s no good unless the call goes through”You dial and you get through in 2-3 secondsConversation is perceptible (MOS scores – later)Lots of standardsHigh interoperabilitySupport structure in place
59 Enterprise TelephonyPrivate companies have different needs than residential users: more services, short dialingWith VoIP this gap has been closingOptions:Small Business Lines (SBL): higher motley fees, limited capabilitiesCentrex Lines: Local PSTN Provider manages Telephony Exchange (TE). Costly and not flexible but offers more features (transfer calls, calls on hold)VPNs: Private network where telephone company manages a private dialing planAcquire own Switch or PBX. Flexibility to add, move, numbers. Key Systems are small PBXs
60 Enterprise Telephony – PBX Business circuit switched telephony system with business featuresCall HoldThree way callingCall transferForwardingPBX usually provides a programming interface: CTI (Computer Telephony Integration) to support additional applications: Call Centers, Conferencing, etc.
61 Enterprise Telephony - PBX Example Key SystemPSTNDS3PBXRemote OfficeMain Office
62 Direct Inward Dial (DID) In DID service the telephone company provides one or more trunk lines to the customer for connection to the customer's PBX and allocates a range of telephone numbers to this line (or group of lines) and forwards all calls to such numbers via the trunkAs calls are presented to the PBX, the dialed destination number (DNIS) is transmitted, usually partially (e.g., last four digits), so that the PBX can route the call directly to the desired telephone extension within the organization without the need for an operator or attendantThe service allows direct inward call routing to each extension while maintaining only a limited number of subscriber lines to satisfy the average concurrent usage of the customerDID service is usually combined with direct outward dialing (DOD) allowing PBX extensions direct outbound calling capability with identification of their DID number
63 Pros and Cons of Voice Telephony Reliable and Excellent qualitySecure and robust enough for DoDUpgradeable to VoIPCons:Steep learning curveExpansive to manage and upgradeCumbersome for phone Add/Move/Change
65 Telecommunications Regulations Federal Communications Commission (FCC) regulations cover telephony, cable, broadcast TV, wireless etc.“Common Carrier”: provider offers conduit for a fee and does not control the contentCustomer controls content/destination of transmission & assumes criminal/civil responsibility for contentLocal monopolies formed by AT&T’s acquisition of independent telephone companies in early 20th centuryRegulation forced because they were deemed natural monopolies (only one player possible in market due to enormous sunk cost)FCC regulates interstate calls and state commissions regulate intra-state and local callsBells independents interconnected & expanded
66 Telephony Industry - Beginnings In 1876 Graham Bell formed Bell Telephone which licensed local telephone exchanges in major US citiesAT&T was formed in 1885 to connect the local Bell companiesIn 1912 AT&T agreed to become regulated monopoly. In exchange they had to connect competing local companies and let Federal Communication Commission (FCC) approve their prices and policies
67 Deregulation of Telephony 1960s-80s: gradual de-regulation of AT&T due to technological advancesTerminal equipment could be owned by customers (CPE) => explosion in PBXs, fax machines, handsetsModified final judgement (MFJ): breakup of AT&T into ILECs (incumbent local exchange carrier) and IXC (inter-exchange carrier) partLong-distance opened to competition, only the local part regulated…Equal access for IXCs to the ILEC network1+ long-distance number introduced then…800-number portability: switching IXCs => retain 800 numberOn January 1st, 1984, a court forced AT&T to give up its 22 local Bell companies, establishing seven Regional Bell Operating Companies (RBOC)1995: removed price controls on AT&T
68 US Telephone Network Structure (after 1984) Eg: AT&T, Sprint, MCILocal access and transport area - LATAEg: Bell Atlantic, Nynex, US West, Southwestern Bell, BellSouth, Pacific Telesys, Ameritech
69 Telecom Act of 1996Required ILECs to open their markets through unbundling of network elements (UNE-P), facilities ownership of CLECsToday UNE-P is one of the most profitable for AT&T and other long-distance players in the local market: due to apparently below-cost regulated pricesILECs could compete in long-distance after demonstrating opening of marketsOnly now some ILECs are aggressively entering long distance marketsCLECs failed due to a variety of reasonILECs still retain over 90% of local marketWireless substitution has caused ILECs to develop wireless business unitsVoIP driven cable telephony + wireless telephony => more demand elasticity for local servicesCLEC – Competitive Local Exchange Carrier
70 VoIP EraIn February 2004, the FCC ruled that electric power companies could use their wiring for Internet service, including voice over IP (VoIP). They also ruled that companies providing computer-to-computer VoIP service should not be subject to the same regulations as telephone companies. (This ruling did not apply to companies that operate gateways between the Internet and the telephone network)The RBOCs remain dominant, although they are under threat from wireless operators, cable companies, VoIP providers, etc.
71 Rest of the WorldOther countries have also introduced private ownership and competition in the telephone industryHowever telephone company was usually a government owned and operated monopolyOnly two percent of world telecommunication revenue is generated by companies that are fully owned by the governmentIn spite of privatization and the introduction of competition, the initial incumbent telephone companies remain very powerful, accounting for 85% of telephone company revenue
73 What does VoIP mean? Voice using IP but not necessarily over the InternetWe’ll see that VoIP requires a great deal of control over the networkespecially for queuingOnce you go out over the global Internet:variety of link speedsvariety of carriers and policiesbest-effort serviceFirst proprietary IP Telephony implementations in 1994, VoIP-related standards available 1996
74 What does VoIP mean? (2) Today, that control is largely lost Efforts to provide control are developing – we’ll look at some of them (RSVP, MPLS)The Result: widely variable voice quality over the InternetNo reason for users to tolerate this – we have a phone network that works pretty well
75 So, Why VoIP? “If it not’s broken, don’t fix it” some truth to this statementtelephony is not “broken”so what is “broken”?Why carry voice? First that comes to mind…Internet has provided NEW SERVICESIntegration of voice and data creates new servicesThere are potential $$$ in carrying voice
76 Why VoIP? (2) Why use IP for voice? Circuit switching works very well But now we have at least two networksThe voice network – equipment that costs A LOTThe data network – the Internet – equipment that might not cost as muchIf I could carry both types of service over the same network, wouldn’t that be a good thing?The phone company already had this idea: B-ISDN – Broadband Integrated Services Digital NetworkHow about operation and management savings?
77 Why VoIP? (3) Lower Equipment Cost Phone systems are generally proprietaryYou buy a PBX and the handsets (phone) that go with itIf you buy an server program, does everyone have to use the same client??? (Outlook can work with Thunderbird, Gmail, Pine, Elm, Eudora, Netscape, etc.)PCs running Linux can often be used unlike PSTNHard to develop third-party software for phone systemsThe WWW is the complete opposite of this paradigmTraditional telephony is like mainframesIP networksMore open standardsMore competition (for the most part)Moore’s Law tends to motivate data networking
78 Why VoIP? (4) Voice/data integration and advanced services User interfaces with HTML (point-and-click) are much easier to deal with than keypad sequences (e.g., #33#7)Unified MessagingEasier new service introductionLower bandwidth requirementsNew voice communications gear can take advantage of low-rate vocodersLegacy telephone network largely stuck with 64 kbps per callSome bandwidth reduction in some areasBandwidth management is better (think of silence periods)
79 Why VoIP? (5) Mobility Widespread availability of IP IP address not tied to a particular areaWidespread availability of IPIt’s EVERYWHEREBut so are telephones Other packet-based mechanisms: FR, ATMNot as presentCircuit-switch networking product development has stoppedAll R&D effort in telephony goes to VoIP telephony
80 Still to rememberIPhas no guaranteesengineered for data trafficdata integrity is very important; TCP does a great job at thislatency – and VARIABLE latency – was not a major concern in the design and engineering of TCP/IPasynchronous – can start, stop, tolerate variability in transmission speedsQuality is generally lower than PSTN but the gap is almost closedEmergency calling featuresPSTN better in handling 911 and power outages
81 VoIP Challenges Speech Quality Network Reliability and Scalability Not quite there yet! But it is a relative measureNetwork Reliability and ScalabilityDe-coupling between application and networkManaging access and prioritizing trafficWhose voice is more important than data?Business CaseVendor bewareSecurity
82 Problems that need to be solved! Session managementUsers may move from terminal to terminal with different capabilities and change their willingness to communicateTo set-up a communication session between two or more users, a signaling protocol is needed: H.323, SIP, MGCP, etc.Media TransportGetting packetized voice over lossy and congested network in real-timeRTP – protocol for transmitting real-time data such as audio, video and gamesEnd-to-end delivery: underlying IP connects the whole world
83 The VoIP MarketVoIP technology has recently matured enough to deliver high-class services to residential and business users for the lower priceTelegeography report:VoIP subscriptions in U.S. will increase from 5.5 million at the start of 2006 to 23.7 million in 2010Revenues to jump from $2.6 billion to $8.1 billionTraditional LEC subscriptions will drop by 22% or $18.2 billion in lost revenue17,443,598 Skype concurrent users were online as of May 11, 2009Total Skype user accounts over 400 MilAdding about 30 million subscribers per quarter
84 Subscriber Growth Potential Same major VoIP players
85 US vs. Europe VoIPSeveral factors help account for Europe’s much more rapid VoIP service growthFreer access to incumbents’ local copper loopsMore aggressive competition and pricingActive participation of incumbent operators such as France Telecom, Deutsche Telekom and BT in the VoIP market
86 International TrendsMobile telephony and VoIP have been key drivers of growthSource: telegeography.com
88 Changing Expectations? THOUGHT 1: What has cellular telephony done to change our expectations regarding voice quality?Tradeoff: mobility vs. availabilityTHOUGHT 2: The rise of cloud computing - hosted voice systems eliminate the need to manage and maintain VoIP infrastructure, which makes IP-based telephony even more accessible
89 Introduction of VoIP – stretched road Takes cash flow from existing PSTN business modelsTime/usage based billing harder to maintainCellular networks expansion – investing in wireline is no longer attractiveUser perspective on QoS and security of VoIP
90 Future RoadmapData traffic grows over 30% a year and it is larger in volumeMobile data traffic outnumbered voice trafficBroadband networks are packet-based technologies (ATM, IP…)Transport core network is moving to the packet technologyEthernet, Metro Ethernet at 1GE, 10GE, 40GE, 100GEIP/MPLS over OTN/WDM
91 VoIP Solutions: Peer-to-Peer Uses PC software to make calls over public and private internetsIt is free but no quality of service guaranteesPlayers:Google TalkTM, Yahoo MessengerTMMicrosoft NetMeetingTMSkypeTMThe InternetPCDSL or Cable Modem RouterDSL or Cable Modem Router
92 VoIP Solutions: Enterprise – Toll Bypass Connecting enterprise PBXs with VoIP links to avoid paying for long distance chargesVendors:NortelNECAvayaToshibaEricssonCisco(PSTN)PBXPrivate Data Network or VPNPBXRouterRouter
93 VoIP Solutions: Service Provider – Local Access Using broadband access to provide local and long distance telephone serviceExample Services Providers:VonageATT CallVantageTMComcastVerizonBroadvoxCoxPCBroadbandModemCall AgentOrdinaryTelephoneBroadband Service ProviderPSTNISPVoice Gateway
94 VoIP Connection Strategies VoIP Customer BaseVoIP Customer BaseRouterRouterIPIPLANIP NetworkLANVoice and data(Voice & Data)IPIP PBXIP PBXIP PhoneIP PhoneTDMTDMVoice NetworkGatewayGatewayStrategy 1: VoIP Base to VoIP Base over IP NetworkStrategy 2: VoIP Base to VoIP Base over Voice NetworkNext two slides present 4 other basic scenariosHybrids of the six basic scenarios are possible
95 VoIP Connection Strategies (cont.) VoIP Customer BasePOTS Customer BaseRouterIPIPIPTDMLANIP NetworkVoice and data(Voice & Data)RouterGatewayPBXIP PBXPOTS PhoneVoice NetworkTDMTDMVoice NetworkIP PhoneGatewayStrategy 3: VoIP Base to POTS Base over IP NetworkStrategy 4: VoIP Base to POTS Base over Voice Network
96 VoIP Connection Strategies (cont.) POTS Customer BasePOTS Customer BasePBXPBXIPIPIPTDMPOTS PhoneIP NetworkPhoneVoice & data(Voice & Data)RouterRouterGatewayGatewayTDMTDMVoice NetworkStrategy 5: POTS Base to POTS Base over IP NetworkStrategy 6: POTS Base to POTS Base over PSTN