Multimedia Communication Network (EECS 491)

Voice over IP (VoIP)

Report

Daniele QuerciaMay, 2001

Abstract

Once you are aware of the benefits and applications of Voice over IP, it is toogood to resist. Perhaps that is why vendors are flooding the market with VOIPproducts and services. The following paper analyzes the various issues in theevolving VOIP technology and the challenges in the development of VOIPproducts

Table of Contents

1. Introduction2. Applications of VoIP3. IP Network Support for Voice

3.1 Delivering voice over IP3.2 IP support for voice

4. Benefits of VoIP5. Problems

5.1 Quality of Service5.2 Quality of Voice5.3 Standards and Interoperability

6. Software Support for VoIP6.1 Audio CODECs6.2 Voice-to-packet software conversion6.3 Tunable factors in VoIP equipment

6.3.1 Jitter buffer settings6.3.2 Packet size6.3.3 Silence suppression

7. Implementing VoIP in Systems8. Market products

8.1 IP telephones8.2 PC based software Phones

9. Future10. References

1. Introduction

Since data traffic is growing muchfaster than telephone traffic, therehas been considerable interest intransporting voice over datanetworks (as opposed to the moretraditional data over voicenetworks).Support for voice communicationsusing the Internet Protocol (IP),which is usually just called “Voiceover IP” or VoIP, has becomeespecially attractive given the low-cost, flat-rate pricing of the publicInternet. In particular, Internettelephony is one of the fastest-growing areas in communicationtoday. Many industry analystsestimate that the overall VoIPmarket will become a multi-billiondollar business within three years.While many corporations have longbeen using voice over Frame Relayto save money by utilizing excessFrame Relay capacity, thedominance of IP has shifted mostattention from VoFR (Voice overFrame Relay) to VoIP. The feasibilityof carrying voice and call signalingmessages over the Internet hasalready been demonstrated butdelivering high-quality commercialproducts, establishing publicservices, and convincing users tobuy into the vision are justbeginning.VoIP can be defined as the abilityto make telephone calls (i.e., to doeverything we can do today with thePublic Switched Telephone Network,PSTN) and to send facsimiles overIP-based data networks with asuitable quality of service (QoS) anda much superior cost/benefit. So,Internet telephony must beprovision of phone service over theInternet. But in sharp constrast with

the conventional telephony, itcarries voice traffic as data packetsover a packet-switched datanetwork instead of as a synchronousstream of binary data over a circuit-switched, time-division multiplexed(TDM) voice network. There aresome substantial benefits to thescheme, which is why companiesand individuals are finding itincreasingly attractive. In fact, forInternet service providers, thepossibility of introducing usage-based pricing and increasing theirtraffic volumes is very attractive.This network convergence alsoopens the door to novelapplications. Interactive shopping(web pages incorporating a “click totalk” button) are just one example,while streaming audio, electronicwhite-boarding and CD-qualityconference calls in stereo are otherexciting applications.Successfully delivering voice overpacket networks presents atremendous opportunity; however,implementing the products is not asstraightforward a task as it may firstappear.

2. Applications of VoIP

The PSTN simply cannot bereplaced, or even dramaticallychanged, in the short term. Theimmediate goal for VoIP serviceproviders is to reproduce existingtelephone capabilities at asignificantly lower “total cost ofoperation” and to offer a technicallycompetitive alternative to the PSTN.The first measure of success forVoIP will be cost savings for longdistance calls as long as there areno additional constraints imposed onthe end user.VoIP could be applied to almost anyvoice communications requirement,ranging from a simple interofficeintercom to complex multipointteleconferencing/shared screenenvironments. The quality of voicereproduction to be provided couldalso be tailored according to theapplication. Customer calls mayneed to be of higher quality thaninternal corporate calls, for example.

Some examples of VoIP applicationsthat are likely to be useful would be:a) PSTN gateways:Interconnection of the Internet tothe PSTN can be accomplished usinga gateway. A PC-based telephone,for example, would have access tothe public network by calling agateway at a point close to thedestination (thereby minimizing longdistance charges).b) Internet-aware telephones:Ordinary telephones (wired orwireless) can be enhanced to serveas an Internet access device as wellas providing normal telephony.

c) Inter-office trunkingover the corporateintranet:Replacement of tie trunksbetween company-ownedPBXs using an Intranet linkwould provide economies ofscale.d) Remote access from abranch (or home) office:A small office could gainaccess to corporate voice,data, and facsimile servicesusing the company’sIntranet.e) Voice calls from amobile PC via the

Internet:Calls to the office can be achievedusing a multimedia PC that isconnected via the Internet.f) Internet call center access:Access to call center facilities via theInternet is emerging as a valuableadjunct to electronic commerceapplications.

One of the immediate applicationsfor IP telephony is realtime facsimiletransmission. Facsimile servicesnormally use dialup PSTNconnections, at speeds up to 14.4Kbps, between pairs of compatiblefax machines. Transmission qualityis affected by network delays,machine compatibility, and analogsignal quality. To operate overpacket networks, a fax interface unitmust convert the data to packetform, handle the conversion ofsignaling and control protocols(theT.30 and T.4 standards), and ensurecomplete delivery of the scan datain the correct order. For thisapplication, packet loss and end-to-end delay are more critical than invoice applications.

3. IP Network Support forVoice

A key requirement for successfulVoIP deployment is the availabilityof an underlying IP-based networkthat is capable of supporting real-time telephone and facsimile.

3.1 Delivering voice over IPDelivering voice from one Internet-phone to another in a voice overInternet protocol (VoIP) system canbe thought of as a six-step process.In step 1, analog voice signals aresampled 8000 times a second andcoded into a 64 Kbps process.Step 2 involves processing thestream with a digital filteringalgorithm to remove any lineechoes. At this step, the bit streamis also analyzed for silent periodsusing a voice-activity detection(VAD) algorithm. When silence isdetected, bandwidth is saved bysuppresing it and not blindy codingit into a lengthly string of zeroes.The fact that a silent interval hasbeen suppressed may be explicitlycommunicated to the receiving end.The system will then fill the intervalwith “comfort noise” so listener willnot think that the line has gonedead.Step 3 is where the bit stream iscompressed and framed on the basisof several ITU standards.In step 4, the voice frame isconverted into IP packet, a processthat itself takes three steps. Thefirst of these is to create a real-timetransport protocol (RTP) packet byadding a 12-byte header to thecompressed voice frame. Then an 8-byte user datagram protocol (UDP)header with source and destinationsocket numbers is added. Finally a20-byte IP header containing the

source and the destinationgateway’s IP address is added.At step 5, the IP packet is sent ontothe Internet. Routers and switchesalong the way examine thedestination IP address, route it inthe correct direction, and deliver itto the destination IP address.Finally, in step 6, the destinationVoIP system reverses the processfor voice playback. The systemextracts the IP packet, UDP packet,and RTP packet, and then extractsthe compressed voice frame. Thevoice frame is decompressed andconverted to analog form for voiceplayback.

3.2 IP support for voiceMost of today’s data networkequipment - routers, LAN switches,ATM switches, network interfacecards, PBXs, etc. will need to beable to support voice traffic.Furthermore, VoIP-specificequipment will either have to beintegrated into these devices orwork compatibly with them. Threedifferent techniques are used(separately or in combination) toimprove network quality of service.• Providing a controllednetworking environment in whichcapacity can be preplanned andadequate performance can beassumed.• Using management tools toconfigure the network nodes,monitor performance, and managecapacity and flow on a dynamicbasis. Queuing mechanisms can alsobe manipulated to minimize delaysfor real-time data flows.• Adding control protocols andmechanisms that help avoid oralleviate the problems inherent in IPnetworks. Protocols such as RTP(real-time protocol) and RSVP

(Resources Reservation Protocol)are also being used to providegreater assurances of controlledQoS within the network.VoIP equipment should beconfigurable to capitalize on thesedifferent techniques but must alsobe sufficiently flexible to add newtechniques as they becomeavailable.

Real-time voice traffic can be carriedover IP networks in three differentways:• Voice trunks can replace theanalog or digital circuits that areserving as voice trunks or PSTN-access trunks. Voice packets aretransferred between predefined IPaddresses, thereby eliminating theneed for phone number to IPaddress conversions.

• PC-to-PC voice can be providedfor multimedia without connecting tothe PSTN.

• Telephony (any phone-to- anyother phone) communicationsappears like a normal telephone tothe caller but may actually consist ofvarious forms of voice over packetnetwork, all interconnected to thePSTN.

Gateway functionality is requiredwhen interconnecting to the PSTN or

when interfacing the standardtelephones to a data network. In thefuture, IP-enabled telephones willconnect directly. Future VoIPnetworks will include IP-based PBXs(iPBXs), which will emulate thefunctions of a traditional PBX. Thesewill allow both standard telephonesand multimedia PCs to connect toeither the PSTN or the Internet.

The most important consideration atthe network level is to minimizeunnecessary data transfer delays.Providing sufficient node and linkcapacity and using congestionavoidance mechanisms (such asprioritization, congestion control,and access controls) can help toreduce overall delay. The ability tomanage network loading (as isfeasible with Intranets but notavailable in the Internet) andoptimize route choices will reducethe effects of jitter. Equipmentproducers should, whereverpossible, avoid proprietarymechanisms (or combinations ofmechanisms) that simply recreatesolutions that are available “off-the-shelf.”

4. Benefits of VoIP

Generally, the benefits of technologycan be divided into the followingfour categories:• Cost Reduction.Although reducing long distancetelephone costs is always a populartopic and would provide a goodreason for introducing VoIP, theactual savings over the long termare still a subject of debate in theindustry.When a telephone call is made withthe VoIP technique, there is nocharge for transfering the IP packetsaccross the Internet. And so long asthe receinving end understands theVoIP protocols, its location does notmatter. Thus, long-distance chargescan be tranformed into a flatmonthly fee.These lower prices, however, arebased on avoiding telephony accesscharges and settlement fees ratherthan being a fundamental reductionin resource costs. The sharing ofequipment and operations costsacross both data and voice userscan also improve network efficiencysince excess bandwidth on onenetwork can be used by the other,thereby creating economies of scalefor voice (especially given the rapidgrowth in data traffic). In particular,network efficency is due to the factthat a packet-switched IP networkcan handle more calls with the sametransmission infrastructure than thePSTN can with its circuit-switchedTDM approach. The IP networkdeploys resources only as needed,whereas TDM dedicates two 64-kb/stime slots to every conversation forthe duration of the call, includingsilences. In addition, VoIP typically

compresses speech quite a bit whichfurther increases network capacity.Moreover, a typical medium-sized orlarge business today employs twonetworks: one for voice (the PSTN)and one for data (the Internet).Both must be paid for andmaintained. Once a company usesits data LAN to carry voice and faxtraffic, network administrators canintegrate many features such asemail, voice mail, paging, callforwarding, and other services foreveryday use. At the same time, byeliminating the conventional voicenetwork, maintenance costs arereduced. Some consultants haveestimated that about 40 percent ofthe total budget can be cut whenvoice data networks are merged intoone.• Simplification.An integrated infrastructure thatsupports all forms of communicationallows more standardization andreduces the total equipmentcomplement. VoIP technologyrepresents an opportunity to unifynetworking technology. Voice,modem, and data-transfer servicesall have different deliveryrequirements. VoIP supports betternetwork adaptability, so thatnetwork support equipment canmore easily be changed toaccomodate changing consumerdemands and market dynamics.Finally, maintenance is generallyless for packet-based routers orswitches than it is for dedicatedcircuit switching boxes. Upgrading arouter is not a difficult as upgradinga circuit switch because the lattersuffers from its real-time deliveryconstraint. Bringing new boxes likerouters and IP switches on-line iseasier and cheaper, and their

configuration can be changedremotely.• Consolidation.Since people are among the mostsignificant cost elements in anetwork, any opportunity tocombine operations, to eliminatepoints of failure, and to consolidateaccounting systems would bebeneficial. Universal use of the IPprotocols for all applications holdsout the promise of both reducedcomplexity and more flexibility.• Advanced Applications. Eventhough basic telephony andfacsimile are the initial applicationsfor VoIP, the longer term benefitsare expected to be derived frommultimedia and multiserviceapplications. For example, Internetcommerce solutions can combineWWW access to information with avoice call button that allowsimmediate access to a call centeragent from the PC. Combining voiceand data features into newapplications will provide the greatestreturns over the longer term.

Although the use of voice overpacket networks is relatively limitedat present, there is considerableuser interest and trials arebeginning. End user demand isexpected to grow rapidly over thenext five years.

5. Problems

VoIP is not without challenges. Theymay be classified into three broadareas: quality of service (QoS),quality of Voice (QoV) and standardsan interoperability.

5.1 Quality of ServiceQoS refers to the issue of maintainigan adequately short delivery timewhen delivering voice packets. Inthe PSTN (except for satellite links,which add about 250 ms), voice isdelivered over even transcontinentaldistances in real time – which lessthan 30 ms of delay. For theInternet, the delay is greater andhighly variable.The quality of sound reproductionover a telephone network isfundamentally subjective, althoughstandardized measures have beendeveloped by the InternationTelecommunications Union, ITU. Ithas been found that there are threefactors that can profoundly impactthe quality of the service:

Delay: A two-way phoneconversation is quite sensitive tolatency. Most callers notice round-trip delays when they exceed250mSec, so the one-way latencybudget would typically be 150mSec. 150 mSec is also specifiedin ITU-T G.114 recommendationas the maximum desired one-waylatency to achieve high-qualityvoice. When considering the one-way delay of voice traffic, onemust take into account the delayadded by the different segmentsand processes in the network. Themost important components ofthis latency are:

- Backbone (network) latency.This is the delay incurred whentraversing the VoIP

backbone. In general, tominimize this delay, try tominimize the router hops thatare traversed between end-points. Alternatively, it ispossible to negotiate or specify ahigher priority for voice trafficthan for delay-insensitive data.- CODEC latency. Eachcompression algorithm hascertain built-in delay. Choosingdifferent CODECs may reducethe latency, but reduce qualityor result in more bandwidthbeing used.- Jitter buffer depth. Tocompensate for the fluctuatingnetwork conditions, manyvendors implement a jitterbuffer in their voice gateways.This is a packet buffer that holdsincoming packets for a specifiedamount of time beforeforwarding them todecompression. However, thedownside of the jitter buffer isthat it can add significant delay.The jitter buffer size isconfigurable, and as shownbelow, can be optimized forgiven network conditions.

Two problems that result fromhigh end-to-end delay in a voicenetwork are echo and talkeroverlap. Echo becomes a problemwhen the round-trip delay is morethan 50 milliseconds.Since echo is perceived as asignificant quality problem, VoIPsystems must address the needfor echo control and implementsome means of echo cancellation.Talker overlap (the problem ofone caller stepping on the othertalker’s speech) becomessignificant if the one-way delaybecomes greater than 250milliseconds. The end-to-end

delay budget is therefore themajor constraint and drivingrequirement for reducing delaythrough a packet network.

Jitter (Delay Variability): Whilenetwork latency effects how muchtime a voice packet spends in thenetwork, jitter controls theregularity in which voice packetsarrive. Jitter is the variation ininter-packet arrival time asintroduced by the variabletransmission delay over thenetwork. Typical voice sourcesgenerate voice packets at aconstant rate. The matching voicedecompression algorithmalso expects incoming voicepackets to arrive at a constantrate. However, the packet-by-packet delay inflicted by thenetwork may be different for eachpacket. The result: packets thatare sent in equal spacing from theleft gateway arrive with irregularspacing at the right gateway, asshown in the following diagram:

Since the receivingdecompression algorithm requiresfixed spacing between thepackets, the typical solution is toimplement a jitter buffer withinthe gateway. The jitter bufferdeliberately delays incomingpackets in order to present themto the decompression algorithm atfixed spacing. The jitter buffer willalso fix any out-of-order errors bylooking at the sequence numberin the RTP frames. Hence, while

the voice decompression enginereceives packets directly on time,the individual packets are delayedfurther in transit, increasing theoverall latency.

Packet Loss: Packet loss is anormal phenomenon on packetnetworks. Loss can be caused bymany different reasons:overloaded links, excessivecollisions on a LAN, physicalmedia errors and others.Transport layers such as TCPaccount for loss and allow packetrecovery under reasonable lossconditions.Due to the time sensitivity ofvoice transmissions, however, thenormal TCP-based retransmissionschemes are not suitable.Audio CODECs also take intoaccount the possibility of packetloss, especially since RTP data istransferred over the unreliableUDP layer. The typical CODECperforms one of several functionsthat make an occasional packetloss unnoticeable to the user. Forexample, a CODEC may choose touse the packet received justbefore the lost packet instead ofthe lost one, or perform moresophisticated interpolation toeliminate any clicks orinterruptions in the audio stream.However, packet loss starts to bea real problem when thepercentage of the lost packetsexceeds a certain threshold(roughly 5% of the packets), orwhen packet losses are groupedtogether in large packet bursts. Inthose situations, even the bestCODECs will be unable to hide thepacket loss from the user,resulting in degraded voicequality.

A very important factor affectingvoice quality is the total networkload. When the network load is high,and especially for networks withstatistical access such as Ethernet,jitter and frame loss typicallyincrease. However, even incongested networks it is sometimespossible to employ packetprioritization schemes, based onport numbers or on the IPprecedence field. These methods,typically built into routers andswitches, allow giving timing-sensitive frames such as voicepriority over data frames. There isoften no perceived degradation inthe quality of data service, but voicequality significantly improves.Another alternative is to usebandwidth reservation protocolssuch as RSVP (resource reservationprotocol) to ensure that the desiredclass of service is available to thespecific stream.

5.2 Quality of VoiceMaintaining voice quality is thesecond challenge in VoIP system.Maintenance of acceptable voicequality levels despite inevitablevariations in network performance(such as congestion or link failures)is achieved using such techniques ascompression, silence suppression,and QoS-enabled transportnetworks.

Can voice quality bemeasured?With all the factors affecting voicequality, many people ask how onemeasuresvoice quality. Standards bodieslike ITU are continuouslyaddressing this issue, and havealready derived two importantrecommendations: P.800 (MOS)

and P.861 (PSQM). P.800 dealswith defining a method to derive aMean Opinion Score of voicequality. The test involvesrecording several pre-selectedvoice samples over the desiredtransmissionmedia and then playing themback to a mixed group of men andwomen under controlledconditions. The scores given bythis group are then weighed togive a single MOS score rangingbetween 1 (worst) and 5 (best). AMOS of 4 is considered “toll-quality” voice.P.861 Perceptual Speech QualityMeasurement (PSQM) tries toautomate this process by definingan algorithm through which acomputer can derive scores thathave a close correlation to theMOS scores. It seems that PSQMwas designed for the circuit-switched network and does nottake into effect importantparameters such as jitter andframe loss that are only relevantto VoIP. As a result of PSQMlimitations, researchers are tryingto come up with alternativeobjective ways to measure voicequality. One such proposal is thePerceptualAnalysis/Measurement system(PAMS) developed by BritishTelecom.

Speech coding an decodingalgorithms are good for reducing thebandwidth for a voice call. But it hasto be remembered that the qualityof voice generally degrades whenhigh compression ratios are used.Efforts are constantly under way tofind algorithms that can boostcompression ratio while maintainingvoice quality.

Software preprocessing of voiceconversations can also be used tofurther optimize voice quality. Onetechnique, called silencesuppression, detects whenever thereis a gap in the speech andsuppresses the transfer of thingslike pauses, breaths, and otherperiods of silence.Another software function thatimproves speech quality is echocancellation. In general, the greaterthe delay, the more offensiveechoes become, and the moreimportant it is to deal with themeffectively. Echo becomes a problemwhenever the end-to-end delay for acall is greater than 50 milliseconds.The ITU recommendation G.168defines the performancerequirements that are currentlyrequired for echo cancellers.

5.3 Standards andInteroperabilityStandards and interoperability arevery important issues and posequite a challenge to all serviceproviders. If one end sends packetusing one speech-coding algorithm,and the receiving end lacks theability to decode it, the service isuseless. Interoperability amongendpoints is a must. The G.711algorithm is established as a default,starting with which both endpointscan begin negotiating whichalgorithm to use in theircommunication.Packet overhead is another issue forVoIP packets. For example, a 10 msvoice frame using the G.729acompression algorithm has only 10bytes of data but a 40-byte header,for whopping 80 percent of headeroverhead. To reduce this overhead,which nullifies much of the

advantage of compression,standards must be revised and newpacket-header schemes must beadopted.

Engineering a VoIP network (andthe equipment used to build it)involves tradeoffs among the qualityof the delivered speech, thereliability of the system, and thedelays inherent in the system.Minimizing the end-to-end delaybudget is one of the key challengesin VoIP systems. Ensuring reliabilityin a “best effort” environment isanother.

6. Software Support forVoIP

Voice and telephone calling can beviewed as one of many applicationsfor an IP network, with softwarebeing used to support theapplication and interface to thenetwork. The emergence of VoIP isa direct result of the advances thathave been made in hardware andsoftware technologies in the early1990s.

VoIP services need to be able toconnect to traditional circuit-switched voice networks. The ITU-Thas addressed this goal by definingH.323, a set of standards forpacket-based multimedia networks.

The H.323 components are: H.323terminals that are endpoints on aLAN, gateways that interfacebetween the LAN and switchedcircuit network, a gatekeeper thatperforms admission controlfunctions and other chores, and theMCU (Multipoint Control Unit) thatoffers conferences between three ormore endpoints.

6.1 Audio CODECsVoice channels occupy 64 Kbpsusing PCM (pulse code modulation)coding when carried over T1 links.Over the years, compressiontechniques were developed allowinga reduction in the requiredbandwidth while preserving voicequality. Such techniques areimplemented as CODECs.Although many proprietarycompression schemes exists, mostH.323 devices today use CODECsthat were standardized by standardsbodies such as the ITU-T for thesake of interoperability acrossvendors. Applications such asNetMeeting use the H.245 protocolto negotiate which CODEC to useaccording to user preferences andthe installed CODECs. Different

compressionschemes canbe comparedusing fourparameters:· Compressedvoice rate – theCODECcompressesvoice from 64Kbps down to acertain bit rate.Some networkdesigns have abig preferencefor low-bit-rate

CODECs. Most CODECs canaccommodate different targetcompression rates such as 8, 6.4and even 5.3 Kbps. Note that thisbit rate is for audio only. Whentransmitting packetized voice overthe network, protocol overhead(such as RTP/UDP/IP/Ethernet) isadded on top of this bit rate,resulting in a higher actual datarate.

· Complexity – the higher thecomplexity of implementing theCODEC, the more CPU resources arerequired.· Voice quality – compressing voicein some CODECs results in verygood voice quality, while otherscause a significant degradation.· Digitizing delay – Each algorithmrequires that different amounts ofspeech be buffered prior to thecompression. This delay adds to theoverall end-to-end delay (seediscussion below). A network withexcessive end-to-end delay, oftencauses people to revert to a half-duplex conversation (“How are youtoday? over…”) instead of thenormal full-duplex phone call.There is no “right CODEC”. Thechoice of what compression schemeto use depends on what parametersare more important for a specificinstallation.

6.2 Voice-to-packet softwareconversionThe software functionality requiredfor voice-to-packet conversion in aVoIP terminal or gateway are:• The Voice Processing module,which prepares voice samples fortransmission over the packetnetwork (see below). This softwareis typically run on a DSP. The VoiceProcessing module must includesoftware to perform the followingfunctions:

a) The PCM Interface, whichreceives samples from thetelephony (PCM) interface andforwards them to theappropriate VoIP softwaremodule forprocessing (and vice versa). ThePCM interface performscontinuous phase resampling of

output samples to the analoginterface.b) The Echo CancellationUnit, which performs echocancellation on sampled, full-duplex voice port signals inaccordance with the ITU G.165orG.168 standard. Since round-tripdelay for VoIP is always greaterthan 50 milliseconds (the pointat which echo becomesintolerable), echo cancellationis a requirement. Operationalparameters may beprogrammable.c) The Voice Activity/IdleNoise Detector, whichsuppresses packet transmissionwhen voice signals are notpresent (and hence savesadditional bandwidth). If noactivity is detected for a periodof time, the voice encoderoutput will not be transportedacross the network. Idle noiselevels are also measured andreported to the destination sothat “comfort noise” can beinserted into the call (so that thelistener does not get dead air ontheir telephone).d) The Tone Detector, whichdetects the reception of DTMFtones and discriminates betweenvoice and facsimile signals.These can be used to invokethe appropriate voice processingfunctions (i.e., the decoding andpacketizing of facsimileinformation or the compressionof voice).e) The Tone Generator, whichgenerates DTMF tones and callprogress tones under commandof the operating system.f) The Facsimile Processingmodule, which provides a

facsimile relay function bydemodulating the PCM data,extracting the relevantinformation, and packing thescan data into packets.g) The Packet Voice Protocolmodule, which encapsulates thecompressed voice and fax datafor transmission over the datanetwork. Each packet includes asequence number that allowsthe received packets to bedelivered in the correct order.This also allows silence intervalsto be reproduced properly andlost packets to be detected.h) The Voice Playout moduleat the destination, which buffersthe packets that are receivedand forwards them to the voicecodec for playout. This moduleprovides an adaptive jitter bufferand a measurement mechanismthat allows buffer sizes to beadapted to the performance ofthe network.

• The Call Processing (Signaling)module, which serves as a signalinggateway allowing calls to beestablished across the packetnetwork. This software supportsE&M (wink, delay and immediate),loop, or ground start FXS and FXO.The Call Processing (signaling)subsystem detects the presence of anew call and collects addressinginformation. Various telephonysignaling standards must besupported. A number of functionsmust be performed if full telephonecalling is to be supported:

· The interface to the telephonenetwork must be monitored tocollect incoming commands andresponses.· The signaling protocols (e.g.,E&M) must be terminated and

the information must beextracted.· The signaling information mustbe mapped into a format thatcan be used to establish asession across the packetnetwork.· Telephone numbers (E.164 dialaddresses) must be convertedinto IP addresses (with thepotential need for an externalreference to a directory service).Two approaches to dialing arebeing used: single stage (dialthe destination number and useautomatic route selectionfunctions), and two stage (dialthe VoIP gateway number, thendial the real destination).

• The Packet Processing module,which processes voice and signalingpackets, adding the appropriatetransport headers prior tosubmitting the packets to the IPnetwork (or other packet networks).Signaling information is convertedfrom telephony protocols to thepacket signaling protocol.

• The Network Managementmodule, which providesmanagement agent functionality,allowing remote fault, accounting,and configuration management tobe performed from standardmanagement systems (see the nextsection). The Network Managementmodule could include services suchas support for security features,access to dialing directories, andremote access support.

VoIP equipment should comply withthe H.323 standard which has beendefined by the ITU to describeterminals, equipment, and servicesfor multimedia communication over

networks (such as LANs or theInternet) that do not provide aguaranteed QoS. In H.323 standard,control messages (Q.931 signalling,H.245 capability exchange and theRAS protocol) are carried over thereliable TCP layer. This ensures thatimportant messages getretransmitted if necessary so theycan make it to the other side. Mediatraffic is transported over theunreliable UDP layer and includestwo protocols as defined in IETF RFC1889: RTP (Real-Time Protocol) thatcarries the actual media and RTCP(Real-Time Control Protocol) thatincludes periodic status and controlmessages.

Although H.323 is the recognizedstandard for VoIP terminals, thereare additional standards that aremore appropriately suited for clientapplications, such as IP phones.

6.3 Tunable factors in VoIPequipment

6.3.1 Jitter buffer settingsThe jitter buffer can be configured inmost VoIP gear. The jitter buffersize must strike a delicate balancebetween delay and quality. If thejitter buffer is too small, networkperturbations such as loss and jitterwill cause audible effects in thereceived voice. If the jitter buffer istoo large, voice quality will be fine,but the two-way conversation mightturn into a half-duplex one.One can decide on a jitter bufferpolicy that specifies that a certainpercentage of packets should fit inthe jitter buffer, say 95%.

6.3.2 Packet sizePacket size selection is also aboutbalance. Larger packet sizessignificantly reduce the overallbandwidth but add to thepacketization delay as the senderneeds to wait more time to fill upthe payload.Overhead in VoIP communications isquite high. Consider a scenariowhere you are compressing down to8 Kbps and sending frames every 20mSec. This results in voice payloadsof 20 bytes for each packet.However, to transfer these voicepayloads over RTP, the followingmust be added: an Ethernet headerof 14 bytes, IP header of 20 bytes,UDP header of 8 bytes and anadditional 12 bytes for RTP. This is awhopping total of 54 bytesoverhead to transmit a 20-bytepayload. In some cases, such anoverhead is fine. In others, thereare two solutions to the problem:

· Increase packet size. Bydeciding to send packets every 40mSec, it is possible to increasethe payload efficiency. Before theinter-arrival time is increased, itshould be verified that the delaybudget can support this.· Employ header compression.Header compression is popularwith some vendor’s equipment,especially on slow links such asPPP, FR or ISDN. This iscommonly called CRTP orCompressed RTP. It compressesthe header down to a few byteson a hop-by-hop basis.

6.3.3 Silence suppressionSilence suppression takes advantageof prolonged periods of silence inconversations to reduce the numberof packets. In a normal interactiveconversation, each speaker typically

listens for about half the time, so itis not necessary to transmit packetscarrying the speaker’s silence.

7. Implementing VoIP inSystemsThe deployment of a VoIPinfrastructure for public use involvesmuch more than simply addingcompression functions to an IPnetwork.

Some of the functions that arerequired for a VoIP system include:a) Fault Management: One of themost critical tasks of anytelecommunications managementsystem is to assist with theidentification and resolution ofproblems and failures. Full SNMPmanagement capabilities using MIBsshould be provided for enterprise-level equipment. Integrating themanagement facilities of thetelephone and data systems usingTMN-based standards is essential forcarrier-class systems.b) Accounting/Billing: VoIPgateways must keep track ofsuccessful and unsuccessful calls.Call detail records that include suchinformation as call start/stop times,dialed number, source/destinationIP address, packets sent andreceived, etc. should be produced.This information would preferablybe processed by the externalaccounting packages that are alsoused for the PSTN calls. The enduser should not need to receivemultiple bills.c) Configuration: An easy-to-usemanagement interface is needed toconfigure the equipment (even whilethe service is running). A variety ofparameters and options areinvolved. Examples include:telephony protocols, compressionalgorithm selection, dialing plans,access controls, PSTN fallbackfeatures, port arrangements,Internet timers, etc.

d) Addressing/Directories:Telephone numbers and IPaddresses need to be managed in away that is transparent to the user.PCs that are used for voice callsmay need telephone numbers, IP-enabled telephones will need IPaddresses (or at least access to onevia DHCP protocols) and InternetDirectory services will need to beextended to include mappingsbetween the two types of address.e) Authentication/Encryption:VoIP offers the potential for securetelephony by making use of thesecurity services available in TCP/IPenvironments. Access controls canbe implemented usingauthentication and calls can bemade private using encryption ofthe links. Implementations of full-scale VoIP systems must provide allthe abilities that are usually takenfor granted in open systems(including the PSTN). These include:

• Interoperability: In a publicnetworking environment differentproducts will need to interwork ifany-to-any communications is tobe possible. Using commonsoftware that has been tested forconformance to all applicablestandards (such as forcompression) can significantlyreduce the cost of productdevelopment. Interconnection ofVoIP to the PSTN also involvesmeeting the specific standards fortelephone network access.• Reliability: The VoIP network,whether by design or throughmanagement, should be faulttolerant with only a very smalllikelihood of complete failure. Inparticular, the gateway betweenthe Telephone and VoIP systemsneeds to be highly reliable.

• Availability: Sufficient capacitymust be available in the VoIPsystem and its gateways tominimize the likelihood of callblocking and mid-call disconnects.This will be especially importantwhen the network is shared withdata traffic that may causecongestion. Mechanisms foradmission control should beavailable for both the voice anddata traffic, with prioritizationpolicies set.• Scalability: There is potentialfor extremely high growth rates inVoIP systems, especially if theyprove the equal of the PSTN atmuch lower cost. VoIP systemsmust be flexible enough to growto very large user populations, toallow a mix of public and privateservices and to adapt to localregulations. The need for largenumbers of addressable pointsmay force the use of improvedInternet protocols such as IPv6.• Accessibility: Telephonesystems assume that anytelephone to call any othertelephone and to allowconferencing of multipletelephones across wide areas.This will be driven by functionsthat map between telephonenumbers and other types ofpacket network address,specifically IP addresses. Theremust, of course, exist gatewaysthat allow every device to bereachable.• Viability: Many are claimingsignificant economic advantagesto the implementation of VoIP.These are often based on flat rateprices for Internet service, thefact that services such as the“Internet 911” are not requiredand that there is no regulatory

prohibition againstinterconnection of telephonesystems with IP systems. Alsoassumed is that higherperformance compression will notbe used in the telephone networkto reduce costs. If circumstanceschange, the motivation for VoIPpurely for cost avoidance reasonsmay change also.

8. Market products

In this chapter some importantmarket products as IP telephonesand PC based software Phones arediscussed.

8.1 IP telephones8.1.1 Cisco’s IP PhonesThe Cisco 30 VIP voice instrument ismarketed as a full featured IPtelephone for executives andmanagers. It provides 30programmable line and featurebuttons, an internal, high-quality,two-way speakerphone withmicrophone mute, and a transferfeature button. The 30 VIP alsoprovides a large 40-character LCDdisplay consisting of two lines of 20characters each. The displayprovides features such as date andtime, calling party name, callingparty number, and digits dialed. AnLED associated with each of the 30feature and line buttons providesfeature and line status.8.1.2 Selsius IP phonesThe Selsius IP Ethernet telephone isa device, which connects to thestandard Ethernet LAN jack. It givesaudio quality comparable to that ofa PBX telephone and it is easy touse with single button access to lineappearances and features. The IPtelephone has many characteristicsof a PC in that it can operate as astandard IP device and has its ownIP address. Because the IP phone iscompatible with H.323, it can talk toother H.323 devices like MicrosoftNetMeeting.8.1.3 Nokia Systems’ IPCourierIPCourier is an Ethernet businesstelephone that delivers PBXfunctionality to the desktop withoutthe PBX. Its features are multiple

line appearances, speakerphonecapability, programmable buttonsfor memory dialing and LCD display.IPCourier also supports advancedcall features such as call waiting,caller ID, forward, transfer, muteand do not disturb. 8.2 PC based software Phones8.2.1 VocalTec Iphone

• PC-to-Phone Communication:requires signing up with anInternet Telephone ServiceProvider (ITSP) and thenregular telephones around theworld can be called.Community Browser serves asa virtual neighborhood inCyberspace. Direct Callingmakes calling as simple asentering an e-mail address. Caller ID, Call Waiting, Muting,Blocking and DirectoryAssistance offer the amenitiesof a full featured phone.

• Live Motion Video: One canactually see the person withwhom he is speaking. (Noadditional hardware isrequired to receive video).

• Audio Conferencing Support:Can talk with up to 100 peopleusing the VocalTecConferencing Server.

• White boarding lets one shareand edit documents, photos,and drawings with otherusers. Text Chat lets fingersdo the talking.

• Multitasking and Auto AcceptCalls let Internet Phone run inthe background while working.

8.2.2 Netscape’s CoolTalkCoolTalk is a real time desktopaudio conferencing and datacollaboration tool specificallydesigned for the Internet. Not onlydoes CoolTalk provide real-timeaudio conferencing at either 9600

baud, 18.4k or 28.8k modemspeeds, but also includes a fullfunction White board, text basedchat tool, and answering machine.8.2.3 White Pine’s CU-SeeMEPro

• Directory Service lets one seea list of all of the userspublished on a particular ILSserver, whether they are usingCU-SeeMe Pro, CU-SeeMeVersion 3.1.2, or MicrosoftNetMeeting.

• One can view up to 12 videoimages simultaneously

• Integrated T.120 datacollaboration for sharingapplications, white board, andfile transfer for multi-usercollaboration duringconferences

• A choice of video and audiocodecs for best performanceover a variety of networkspeeds

• It is H.323 compatible so onecan make point-to-point callsto users of MicrosoftNetMeeting, Intel ProShareand other H.323 clients

• It is available for Windows andWindows NT

8.2.4 Microsoft Net MeetingNetMeeting for Windows is an awardwinning product that provides themost complete conferencing solutionfor the Internet and corporateIntranet. Its features let onecommunicate with both audio andvideo, collaborate on virtually anyWindows based application,exchange graphics on an electronicwhite board, transfer files or use thetext based chat program. Using thePC and the Internet, one can nowhold face-to-face conversations withfriends and family around the world.NetMeeting works with any video

capture card or camera thatsupports Video for Windows.The main benefits of this softwareare:

• Multipoint Data Conferencing.Allows sharing of any Windowsbased application or folder withseveral other participants usingstandards based T.120 dataconferencing. There is also anelectronic white board, textbased chat as well as file transfercapabilities.• Internet Audio/Video

Conferencing.With a sound card, microphone,and speakers, NetMeeting letsone place standards based H.323audio calls over the Internet or acorporate Intranet. Addition of avideo camera permits face-to-face communication.

9. Future

VoIP is leading-edge technology,being developed as the markettrends and market requirementsget clearer. This evolution of theVoIP market can be expected togo through three phases. Themarket for real-time services overIP will evolve from the currentarbitrage market, to the verticalbusiness applications andeventually widely deployedmultimedia applications. This doesnot mean that the technology istoo immature to provide benefitsunder the right circumstances. Alarge and increasing number ofthe so-called next generationTelcos are offering telephoneservices over IP. Traditionaltelecom operators such as AT&T,Telenor and MCI have startedutilizing VoIP for both telephoneand fax traffic. Consulting andmarket research firm Frost &Sullivan expects voice-over-Internet Protocol (VoIP)technology to be the mostsignificant development in thetelecommunications industry sincewireless technology. Companiessuch as Nortel, Lucent and Ciscohave staked their futures onInternet telephony.Although the advantage thatsending information over an IPnetwork is more efficient andcosts less than traditional circuit-switched technologies lead someindustry analysts to predictInternet-based voice services willsoon be widely adopted, thetechnology still faces concerns,including voice quality of VoIPcalls. In order for widespreadacceptance, technical concernsover call quality must be

overcome. In addition, a lack oftechnical standards, which wouldallow multiple systems to worktogether, could hold the marketback.Despite these challenges, analystspredict widespread adoption ofVoIP around the year 2010.

10. References

• Marcus Gonclaves, "Voice over IP Networks", 1998• Uyless Black, "Voice over IP", 1999• Jonathan Davidson, Jim Peters, "Voice Over IP Fundamentals," Macmillan,

November 1999• Matthew Kolon, Walter J. Goralski, "IP Telephony," McGraw Hill, September

1999• D. Minoli and E. Minoli, "Delivering Voice over IP Networks," John Wiley,

1998• Bill Douskalis, , "IP Telephony: The Integration of Robust VolP Services,"

Prentice Hall, 1999• Tom Mercer , "An Overview of the Internet Telephony Market", Compaq

White papers• Richard jones, Jesus Cruz, Sridhar Solur, "Carrier Class Voice over IP",

Compaq White Papers• Jerry Ryan, "Voice Over IP", Techguide• Henning Schulzrine, Jonathan Rsenburg, " The IETF Internet Telephony

Architecture and Protocols ", IEEE Networks (June 99), pp 18-23• Christian Huitema, Jane Cameron, Petros Mouchtaris, Darek Smyk, " An

Architecture for Residential Telephony Service ", IEEE Networks (June99),pp 50-55

• "AVVID", Cisco White Paper ,30 pages,• Daniele Rizetto, Claudio Catania, " A Voice over IP Service Architecture for

Integrated Communications ", IEEE Networks (June 99), pp 34-39• Bjarne Munch, "IP Telephony - Today/Tomorrow Ever? ", Ericsson White

Paper