voice over ip

COMMUNICATIONS OF THE ACM January 2002/Vol. 45, No. 1 89

others as Internet telephony service providers(ITSPs) is likely to further increase competitionamong all phone service providers. Many commu-nication technology vendors are rolling out hybridIP/PBX systems. Both traditional and recentlyestablished carriers are beginning to offer voice overIP (VoIP) network connectivity to both businessand residential customers (see the sidebar “PC-to-Phone Providers).

VoIP involves sending voice transmissions as datapackets using the Internet Protocol (IP), wherebythe user’s voice is converted into a digital signal,compressed, and broken down into a series of pack-ets. The packets are then transported over private orpublic IP networks and reassembled and decoded

on the receiving side (see Figure 1). Residential cus-tomers can connect to IP-based networks by usingthe local loop from the PSTN or high-speed lines,including ADSL/DSL and cable modems.

Several recent industry surveys and projectionsestimate that VoIP could account for over 10% ofall voice calls in the U.S. by 2004. It’s likely to beused first in places with significant IP infrastructureor where cost savings are significant; an examplemight be a company with multiple sites worldwideconnected through a private or public IP network.However, VoIP deployment may not be possibleeverywhere, as some countries restrict the use ofVoIP to prevent harming their monopolistictelecommunication markets. VoIP might also besuitable for highly distributed companies or forcompanies with seasonally variable voice-servicedemand.

The idea of VoIP, or voice over the Internet or IPtelephony, has been discussed since at least the early1970s [6] when the idea and technology were devel-oped. Despite this history, VoIP didn’t establish acommercial niche until the mid-1990s. This grad-

How can voice over the Internet claim a greater share of the worldwide phone market from the voice infrastructure dominated for more than 100 years by the public-switched telephone network?

Voice has been transmitted over the public-switched telephone network (PSTN) since

1878 while the U.S. long-distance market has grown to about $100 billion a year in

business and residential demand. The desire of businesses and consumers alike to reduce

this cost, along with the investment over the last decade in IP-based networks, public

and private, has produced substantial

interest in transmitting voice over IP net-

works. The possible re-emergence of

Internet service providers (ISPs) and

VoiceOver IP

Upkar Varshney, Andy Snow, Matt McGivern, and Christi Howard

90 January 2002/Vol. 45, No. 1 COMMUNICATIONS OF THE ACM

ual commercial development can be attributed to alack of IP infrastructure and the fact that circuit-switched calling was and still is a much more reli-able alternative, especially in light of the poorquality of early VoIP calls. In 1995, Vocaltec(www.vocaltec.com) produced the first commer-cially available VoIP product requiring both partic-ipants in the call to have the software on a PC as

well as Internet access. Unfortunately, it did notallow traditional calls through the PSTN.

Following the rapid growth of the public mass-market Internet, especially the Web, during theearly 1990s and accompanying investment in IPnetworking infrastructure by businesses, vendors,and carriers, VoIP has finally become a viable alter-native to sending voice over the PSTN. A number

of factors are influencing theadoption of VoIP technology.First and foremost, the cost of apacket-switched network forVoIP could be as much as halfthat of a traditional circuit-switched network (such as thePSTN) for voice transmission[9]. This cost saving is a resultof the efficient use of bandwidthrequiring fewer long-distancetrunks between switches. Thetraditional circuit-switched net-works, or the PSTN, have todedicate a full-duplex 64Kbps

Circuit switched (end-to-end dedicated circuit set up by circuit switches)

64Kbps pr 32Kbps

< 100ms

Dedicated

Business customers. Monthly charge for line, plus per-minute charge for long distance, cost of PBX, and other telephony equipment. Residential customers. Monthly charge for line, plus per-minute charge for long distance, cost of simple phone.

Dumb terminal (less expensive); intelligence in the network

Requires reprogramming or changes in the network design but fast enough to add if advanced intelligent networks (AIN) are in use.

High (extremely low loss)

Only once when the service is installed

Many at federal and state levels

99.999% up time

Not a problem; powered by a separate source from phone company.

High level of security because one line is dedicated to one call.

Mature (simplified interworking among equipment from different vendors).

Switching

Bit rate

Latency

Bandwidth

Cost of access/billing

Equipment

Additional features and services

Quality of service

Authorizationand authentication

Regulations

Network availability

Electrical power failure at customer premises

Security

Standards/status

Packet switched (statistical multiplexing of several connections over links).

14Kbps with overhead*

200–700ms depending on the total traffic on the IP net-work. Lower latencies possible with private IP networks.

Dynamically allocated

Business customers. Cost of IP infrastructure, Hybrid IP/PBX, and IP phones. Residential customers. Monthly charge for line, plus monthly charge for ISP, cost of computer, and other equipment.

Integrated smart programmable terminal(expensive); intelligence not in the network.

Easy to add without major changes, due to flexible protocol support, but standards are needed for traditional user services.

Low and variable, but traffic is sensitive depending on packet loss and delay experienced.

Potentially required, per-call basis

Few yet, but regulatory uncertainty; future regulations may reduce the cost advantages of VoIP.

Level of reliability is not known.

Will have problems, as equipment may be down. Power from other sources is not easy to obtain.

Possible eavesdropping at routers.

Emerging possible problems in interworking.

Voice over PSTNConcept Voice over IP

Table 1. A qualitative comparison of voice over PSTN and over IP.

*Only when speaker is talking

Figure 1. A possible scenario for VoIP for business customers.

A/D Compression PacketAssembly Packet

Switch

LAN or FullDuplex Line

The Internet

Public IP CarrierPacketDisassemblyD/A Decompression

VoIP Telephone

Private IP Network

channel for the duration of a single call. The VoIPnetworks require approximately 14Kbps, as voicecompression is employed, and the bandwidth is usedonly when something has to be transmitted. Moreefficient use of bandwidth means more calls can becarried over a single link, without requiring the car-rier to install new lines or further augment networkcapacity; Table 1 compares voice over PSTN andover IP.

Besides cost savings and improved network uti-lization, VoIP offers other features, including callerID and call forwarding, that can be added to VoIPnetworks at little cost [5]. VoIP allows Internetaccess and voice traffic simultaneously over a singlephone line. This function could eliminate the needfor two phone lines in a home, one for data and onefor voice, by using the same line to carry all trafficwithout concern for missed calls or being discon-nected from the ISP. Other high-speed media, suchas ADSL and cable modems, can be used to carryboth data and voice to IP networks while lettinghome customers use regular phone lines for voicecalls to and from the PSTN. In this way, VoIP ser-vice offered by ISPs and ITSPs might indirectly ben-

efit existing telephone companies and cableproviders by increasing the potential number ofADSL and cable modem subscribers nationwide.

Long-distance carriers in the U.S. pay an averageof $0.0171 per minute in interstate access charges tothe regional Bell operating companies, that is, thelocal phone companies [8], a total of $9 billion a year.One current VoIP cost advantage is that ISPs pay noaccess charges, due to a U.S. Federal Communica-tions Commission exemption under enhanced-service-provider regulations. However, any changes inregulation requiring ISPs and ITSPs to pay accesscharges or treat calls to ISPs as long-distance calls maydiminish the VoIP cost advantage.

One VoIP application might involve managingtemporary overload call volumefor business users. Using a regularPBX, most traffic can be servicedwith existing telephony equip-ment, and any excess or overloadtraffic can be routed to an IP/PBXsystem that can then be servicedby remote call centers with IPinfrastructure (see Figure 2).

Technical IssuesAmong the many technical issuesin VoIP, a major one is end-to-end delay, or latency. To ensuregood voice quality, latency for

voice communication should not exceed 200 mil-liseconds, as demonstrated in the 1980s when carri-ers tried to offer voice services over geosynchronoussatellites; users deemed the 270-millisecond delayunacceptable. However, under certain circum-stances, VoIP might suffer from more latency, lead-ing to unacceptable quality (due to the uncertaintyas to whether the other person is talking, possiblyleading to interruptions). Latency is influenced by anumber of variables. First, other traffic on IP net-works directly affects the delay for voice packets.Another is packet size, with smaller packets receiv-ing less end-to-end delay, due to faster routing andother factors, while increasing overhead on the sys-tem. Latency is also related to the number of routersand gateways that packets have to travel throughbefore reaching their destinations. Table 2 outlinesthe four most common causes of packet delay overIP-based networks, public and private.

Some VoIP systems send test messages to severalrouters over IP networks to find the paths with bet-ter quality in terms of less delay. These smart tech-niques do not always yield better quality, especiallyover public IP networks like the Internet, due to


LAN/WAN connection

Overflow calls

Calls fromcustomersPBX

HybridIP/PBX

Customerservicecenter

Call centerwith IP phones

Call centerwith IP phones

Figure 2. Managing temporary overflowof calls using VoIP.

Table 2. The delay factors in VoIP.

Processing at a switch/router

Transmission time, or time to put packets online.

Propagation delay, or the actual time it takesthe signal to pass between two switches.

Variable delays, or litter, introduced when packets get out of order and must be buffered and reordered before play.

Speech encoding, compression, anddecompression.

Variable, depending on the speed and traffic on the switch; usually 5–10msec per packet per hop.

Packet size in bits divided by line speed inbits/sec.

Fixed time for a given length of the segment.

Variable, depending on traffic on routersand switches in the IP network.

5–10msec per packet.

Cause of Delay Length of Delay

possible rapid fluctuations in the amount of trafficand resulting increase in delays experienced by thepeople speaking and listening on the line.

VoIP systems use the User Datagram Protocol(UDP) as a transport layer protocol on top of IP toavoid acknowledgments for lost packets. Acknowl-edgments trigger undesirable retransmission ofvoice packets and increase network traffic (and end-to-end delay) and thus affect the quality of service(QoS) for VoIP. Some packet loss is tolerable; forexample, many voice encoders can handle up to 1%packet loss [2].

For users who prefer traditional telephones, not

specialized equipment, Internettelephony gateways can be usedwhere two users communicatewithout having a computer ateither of their locations. A gate-way’s basic architecture involvesa user connection via thePSTN. The gateway computerthen searches for another gate-way computer near the targetlocation and makes a connec-tion using circuit switching.When this connection is made,the second gateway utilizes thelocal PSTN to complete thecollection of the call. Thoughthis type of call isn’t completelyIP, it does suggest possiblefuture solutions for integratingthe current PSTN and the VoIPsystem. Table 3 compares fourimplementations for supportingVoIP.

However, data packets travel-ing through the Internet maynot be secure and may requireencryption, adding overhead byincreasing the necessary bit ratebeyond 14Kbps, hence reduc-ing the bit rate advantage ofVoIP over PSTN. Encryptionalso increases the end-to-endlatency caused by the processingdelay for encryption anddecryption.

Meanwhile, technology sup-port for VoIP has begun tomature on a number of fronts.The newer generations of routersand switches are faster and betterable to handle the added load of

real-time data packets. Beyond the advances in com-pression and equipment, protocol support in theform of the Resource Reservation Protocol (RSVP)and IP version 6 (IPv6) are also starting to mature.These protocols offer ways to prioritize voice trafficover the Net, helping improve QoS, especially whenthe network is congested.

Protocol SupportJust as in conventional telephony, VoIP needs aconnection between users, though in the case ofVoIP, a virtual connection. VoIP architectureinvolves many components. First, a signaling proto-


Figure 3. Proposed evolution path for traditional PSTN carriers.

800DB

LNPDB

SS7Network

CurrentPSTN Infrastructure

Proposed VoiceOver Packet Infrastructure

PBX PBX

CircuitSwitch

CircuitSwitch

SignalingGateway

TrunkGateway

Access Gateway

BillingAgent

CallConnection

Agent

Core PacketNetwork

Traffic Control

PC Webphones

VoIPgateway

Public IPvoicecarriers

Voice-enabledbrowsers

Software that allows any PC with a sound card and a microphone to transmit voice tosimilarly equippedmachines.

Used between the PBX and an IP network/LAN, translating and routing the calls to other gateways.

Phone companies that completedbypass the PSTN and provide just VoIP. May still need the local loop.

Combining voice access with Web browsers.

QoS issues, along with the requirement that both users have similar equipment/software.

QoS issues need to be addressed; high initial coast of the gateway equipment.

Not as reliable asPSTN; QoS issues still need to be resolved; only available in limitedareas; future regulation may affect.

Regular dialupconnections limitbandwidth available for the combined services.

Vocaltec (www.vocaltec.com) and Net2phone(www.net2phone.com)

Quicknet(www.quicknet,net)

Net2phone (www.net2phone.com)allowing PC-to-phonecalls (not the otherway round) at low rates

Both Netscape andExplorer have plug-insavailable.

Description Example

Only cost is computer and connection to ISP.

Cost savings in local and long-distance calls, better utilization of network resources.

Currently immune from line-access charges, cheaper phone services, more advanced features; reduced infrastructure costs.

Good for services like live customer service for Web sites and e-commerce solutions.

ProsApproach Cons

Table 3. Some VoIP implementations.

col is needed to set up individual sessions for voiceconnections between users [2]. Once a session isestablished, a transport protocol can be used to sendthe data packets. Directory access protocols areanother important part of VoIP, providing routingand switching information for connecting calls.

A signaling protocol handles user location, ses-sion establishment, session negotiation, call partici-pant management, and feature invocation. Sessionestablishment is invoked when a user is located,allowing the call recipient to accept, reject, or for-ward the call [6]. Session negotiation helps managedifferent types of media, such as voice and video,transmitted at the same time. Call participant man-agement helps control which users are active on thecall, allowing for the addition and subtraction of

users. The signaling protocol also involves featureinvocation, at which time call features, such as hold,transfer, and mute, are controlled.

The Realtime Transport Protocol (RTP) can beused to support the transport of real-time media,including voice traffic, over packet networks. RTP-formatted packets contain media information and aheader, providing information to the receiver thatallows the reordering of any out-of-sequence pack-ets. Moreover, RTP uses payload identification toplace an identifier in each packet to describe theencoding of the media so it can be changed in lightof varying network conditions [7]. The Real TimeControl Protocol (RTCP), a companion protocolfor RTP, provides QoS feedback to the sendingdevice, reporting on the receiver’s quality of recep-tion. The Real Time Streaming Protocol (RTSP)can be used to control stored media servers, ordevices capable of playing and recording mediafrom the server. This added RTSP-based controlallows the integration of voice mail and prerecordedconference calls in VoIP environments. The abilityto integrate these advanced services is important tothe future growth of VoIP. The Session InitiationProtocol (SIP) can be used to establish, modify, and

terminate multimedia calls.To encourage rapid, widespread deployment of

VoIP services, several standards bodies have gener-ated agreements based on groups of existing proto-cols and standards. The two most important are theH.323 recommendation from the InternationalTelecommunication Union and Media GatewayControl Protocol (MGCP) from a branch of theInternet Engineering Task Force. Neither is astandalone protocol but relies on other protocols tocomplete their jobs [1]. The H.323 architecture isbased on four components: terminals, gateways,gatekeepers, and the multipoint control unit(MCU). Gateways are used for protocol conversionbetween IP and circuit-switched networks. Gate-keepers are used for bandwidth management,

address translation, and callcontrol. H.323 provides a foun-dation for audio, video, anddata communications across IP-based networks, including theInternet. Complying withH.323 enables different multi-media products to interoperate.H.323 depends on other stan-dards, such as H.245, to negoti-ate channel usage andcapabilities, modified Q.931 forcall signaling and call setup,

Registration Admission Status for communicatingwith a gatekeeper, and RTP/RTCP for sequencingaudio/video packets. The MCU supports multicastconferences among three or more end points byusing H.245 negotiations to determine users’ com-mon capabilities [1].

The Media Gateway Control Protocol (MGCP)defines communications among call agents (mediagateway controllers) and telephony gateways. Callagents have the intelligence for call control andother functions and manage telephony gatewaysused for protocol conversion. A call agent in MGCPis analogous to a gatekeeper in H.323 [1]. TheMGCP can use the Session Initiation Protocol(SIP), which uses the HTTP format to allow a userto initiate a call to be initiated by clicking on abrowser.

Although H.323 and the MGCP have been stan-dardized by two different standard-setting bodies,some of their functions are quite similar. Both thegatekeeper in H.323 and the call agent in MGCPmanage and control gateways and participate in set-ting up, maintaining, and terminating the VoIP’stelephone connection. The MGCP can also be usedas part of H.323 for simplified interworking.


Figure 4. Possible coexistence scenario for PSTN and VoIP.

VoIP carrier

Private IPnetworks

The Internet

PSTN

HybridIP/PBX

HybridIP/PBX

PSTN and VoIPThe PSTN has served the needs of businesses andconsumers worldwide for more than 100 years andhas gone through major technological advances,including survivable long-distance networks basedon synchronous optical network (SONET) rings,intelligent networking, Signaling System No. 7(SS7)-based signaling, and a high degree of redun-dancy in telephone switches. All these increasinglyadvanced features and components have increasedthe reliability of the PSTN; it is estimated thatbecause of them the PSTN is today operational99.999% of the time. The PSTN also offers lowlatency rates and very high quality during voicetransmission.

With the emerging potential of IP networks toprovide integrated voice-data communications,conventional PSTN carriers realize they have torespond to this competitive threat. For example,Telcordia (formally Bellcore) has developed a Voiceover Packet (VoP) architecture and has initiated anindustrywide effort to develop generic requirementdocuments; they will allow local and interexchangecarriers, vendors, and other stakeholders to addressinteroperability issues associated with networks,services, protocols, and equipment. These initia-tives recognize that because bundled services cannotbe offered cost-effectively by separate networks,they have to identify a migration path for PSTNcarriers preserving their investment in circuit-switched technology and services. This migrationpath is supposed to allow PSTN carriers to modifyand add only some components in existing net-works for offering multiple services, including VoIP.

Telcordia’s Next Generation Network and VoParchitecture (NGN/VOP) represents a vision for thecoexistence of these two technologies (see Figure 3)[3]. The current PSTN is controlled by SS7, an out-of-band packet-switched network used to coordi-nate the establishment, use, and termination ofcircuit-switched calls through circuit switches andtrunks. The SS7 network also allows other servicesto be provided, including 800-number dialing andlocal number portability, as required by the U.S.Telecommunications Act of 1996 to foster competi-tion in local telephone markets.

The NGN/VOP architecture involves a numberof elements [3]:

Core packet network. Unlike the PSTN, this classi-cal IP network carries both control and trafficpackets.

Call connection agent (CCA). This software providescall-processing functionality. An IP network is a

best-effort packet delivery service; something hasto set up, manage, and disable virtual voice con-nections. Moreover, packets might be lost in erroror arrive out of sequence. The routing and man-agement of a virtual call across a core network isessential. The CCA also has to generate SS7 mes-sages if 800 toll-free dialing, local number porta-bility, and other services are desired.

Signaling gateway. This device is the control bridgebetween the circuit-switched and packet-switched worlds needed to manage end-to-endcalls through both infrastructures.

Trunk gateway. This traffic bridge terminates cir-cuit-switched trunks on the PSTN side and vir-tual connections on the packet-switched side.

Access gateway. This device provides alternativeaccess for subscribers not traversing the PSTN.The access gateway sets up transport connectionsthrough the core network when directed by theCCA; it also provides ringing and other func-tions.

Billing agent. This agent gets raw usage data fromthe CCA and generates formatted messages forback-end billing platforms.

This architecture allows existing PSTN to evolveinto a network supporting both traditional and IP-based voice communications. Though the phonecompanies serve more than 100 million U.S. sub-scribers today, they have to provide bundled ser-vices in the future if they hope to maintain orincrease their existing client base. The fate of thisevolutionary architecture depends on carriers beingable to forge interoperability consensus amongthemselves and with vendors.

VoIP Adoption and ProspectsSeveral factors regarding the adoption of VoIP makeit difficult to forecast adoption rates. The first dealswith how quickly existing carriers might transitionaway from their current technology. Another dealswith demand for services from emerging carriers andother service providers who are unencumbered bysunken investment in the PSTN. Another deals withthe regulatory environment. And yet another dealswith users who will undoubtedly demand not onlythe same high QoS to which they are accustomedbut cost-effective bundled services as well.

Many users resist changing to VoIP until they areshown the new service’s tangible benefits, includingreduced cost or more features; they are certainlyunlikely to accept lower quality. In addition, manyorganizations have invested a great deal of money inPBX and other phone equipment. The availability


of new hybrid PBX/VoIP systems, which can beinstalled as old equipment is phased out might sig-nificantly influence the speed of VoIP adoption. Thecost today of VoIP end-user equipment is muchgreater than for traditional phones. However, theemergence of devices that do not require a computerbut connect to existing phones may help increaseuser acceptance (see www.phoneworld.net/aplio/).

VoIP also has to address the issue of security fortransmitted messages before it can become univer-sal. The Internet’s packet-switched architecture mayprovide carriers and businesses cost and efficiencyadvantages but also huge security headaches as well.Along with IPv6, many versions of VoIP softwarehave built-in encryption, offering better securitythan older implementations. Table 4 lists severalfactors that could affect VoIP adoption.

These issues make it evident that VoIP will notcompletely eliminate but rather integrate with andwork in parallel with the traditional establishedPSTN. Even though the two systems reflect quitedifferent design philosophies and commercial histo-ries as to their switching mechanisms, they also share

some of the same technologies and links. For exam-ple, each system utilizes the local loop to reach theend user. Additionally, VoIP relies on the PSTN toenable its users to reach their ISPs and Internet gate-way servers. The two systems are likely to coexist forthe foreseeable future, each one serving a particularmarket or purpose. This competitive coexistenceshould continue until VoIP quality and reliabilityfinally catches up to PSTN, and some of the olderPSTN architecture becomes outdated and needs tobe replaced. Figure 4 shows one possible scenario forPSTN and VoIP coexistence for customers.

The Cahners In-Stat group estimates that VoIPgateway sales will reach $4 billion in sales in theU.S. in 2003. As a harbinger of VoIP deployment,Cisco Systems has many business customers withmore than 2,000 IP phones [4]. Moreover, manyother small but technologically advanced companiesare likely to install IP/PBX systems; Gartner Grouppredicts that 50% of all small companies will haveIP/PBXs by 2004. One major factor influencingwould-be commercial customers is the ability ofvendors to offer large IP/PBX systems that match


Cost of existing or legacy infrastructureCost of upgradingCost of accessCost of management

Possible improvements with provisioned bandwidthBetter quality with private IP networksPossible use of IPv6

Emergence of hybrid (IP/PBX) equipment (IP/PBX available in 1,000-line range, possibly scalable to10,000 lines). Alsopossible to link several IP/PBX units together, but still not comparable to many 50,000-line switches employed in PSTN.

Possible change in regulations may affect access cost for users to ITSPs.

Different countries have different views on Internet access and IP telephony; differences in quality and level of infrastructure.

Perceived unmanageability of public IP networks may hurt use of the Internet for business VoIP; carriers should consider adding sophisticated network management and monitoring features in their VoIP offerings.

Differences in control signaling and features may have to be addressed; infrastructure and interworking effect on QoS should be considered.

User cost of VoIP likely to shift from per-minute to fixed or usage-sensitive class-based pricing; new business models are necessary for revenue sharing among multiple ISPs and ITSPs.

VoIP traffic may affect the delay (and QoS) of other important data on IP networks; VoIP traffic growth should be monitored and attempts made for allowing sufficient bandwidth for VoIP for required voice quality.

Effect of security threats and possible security weaknesses in VoIP features and implementation should be considered; user authentication and authorization, along with billing software, should be carefully implemented and monitored.

Methods of PSTN reliability (such as fault-tolerance, hot standby, redundancy) should be incorporated in VoIP networks, both public and private IP.

New IP phones and IP adapters for existing phones should ease the transition to VoIP; the cost of the equipment may be a factor for some users; good cognitive interfaces.

Bundled services can be provided with VoIP networks; cost savings and effect of network failure on all services should be considered.

Cost factor

Quality

Equipmentavailability

Regulations

Global connectivity

Networkmanagement

Interworking withdiverse networks

Future pricing andrevenue sharing

Possible effect on trafficvolume in IP networks

Security and hackingthreats

Reliability andfailure issues

User equipmentrequirements

Service integration(voice and data)

CommentsIssues

Table 4. Factors affecting VoIP adoption.

large PSTN switches in terms of cost, size, numberof lines, reliability, and configurability. The last-mile issue can be resolved if carriers offer high-bandwidth service aggregation points at businesscustomers’ premises. Due to the perceived unman-ageability of public IP networks, it’s unlikely thatmost VoIP traffic will be carried by public IP net-works in 2004. According to some estimates, it’slikely to be less than 20% even by 2004 [1].

As VoIP gains a commercial foothold, wirelessVoIP might emerge as a way to transmit voice overthe Internet from cell and personal communica-tions services (PCS) phones. This advance couldaffect cellular and PCS providers as their customersgain the option of connecting to IP networks forlong-distance calls. Since the number of wirelesscustomers is increasing exponentially and the costof wireless long-distance service remains high, theeffect of WVoIP on wireless carriers may be signifi-cant, despite the hurdles of QoS and reliability. Bet-

ter loss algorithms and transmission equipment arealso needed before WVoIP becomes an engineeringand commercial reality.

ConclusionOur aim here has been to provide background infor-mation, major concepts, and issues concerning thetechnology, deployment scenarios, and approaches toprotocol support for VoIP. We’ve also addressed anumber of unresolved engineering and marketingquestions and how VoIP might coexist with the tradi-tional voice infrastructure. Other areas where the tech-nology of VoIP must be developed further before fullor even substantial adoption is possible include billingand customer service; VoIP implementers must stilldetermine the most effective billing structure for callsplaced using VoIP systems and develop proceduresand systems for implementing it (for more on VoIP,please see www.cis.ohio-state.edu/~jain/refs/ref_voip.htm). Ultimately, services, QoS, and cost-effec-tiveness will determine the speed of VoIP adoptionand evolution.

References1. Black, U. Voice over IP. Prentice Hall, Upper Saddle River, NJ, 2000.2. Goyal, P., Greenberg, A., Kalmanek, C., Marshall, W., Mishra, P., and

Nortz, D. Integration of call signaling and resource management for IPtelephony. IEEE Internet Comput. 3, 3 (May/June 1999), 44–52.

3. Katzenberger, G., Ed. Telcordia Digest Tech. Info. SR-104 17, 2 (Feb.2000); see www.telcordia.com/resources/genericredigest/downloads/feb2000digest.pdf.

4. News@Cisco; see newsroom.cisco.com/dlls/innovators/VoIP/.5. Polyzois, K., Purdy, H., Yang, P., Shrader, D., Shinnreick, H., and

Schulzrinne, H. From pots to pans: A commentary on the evolution toInternet telephony. IEEE Internet Comput. 3, 3 (May/June 1999), 83–91.

6. Schulzrinne, H. Service for telecom, Version II. IEEE Internet Comput.3, 3 (May/June 1999), 40–43.

7. Schulzrinne, H. and Rosenburg, J. Tutorial: The IETF Internet tele-phony architecture and protocols. IEEE Internet Comput. Online(1999); see www.computer.org/ internet/ telephony/w3schrosen.htm.

8. U.S. Federal Communications Commission, Industry Analysis Divi-sion, Common Carrier Bureau. Monitoring Report and Access Tariff Fil-ings. Statistical Trends in Telephony. Washington, DC, Aug. 2001); see www.fcc.gov/Bureaus/Common_Carrier/Reports/FCC-State_Link/trends.html.

9. Weiss, M. and Hwang, J. Internet Telephony or Circuit-Switched Tele-phony: Which is Cheaper? School of Information Science, University ofPittsburgh, Pittsburgh, PA, Sept. 1999; see www2.sis.pitt.edu/~mweiss/papers/itelv3b.pdf.

Upkar Varshney ([email protected]) is an assistant professorin the Department of Computer Information Systems at GeorgiaState University, Atlanta, GA.Andy Snow ([email protected]) is an assistant professor in theDepartment of Computer Information Systems at Georgia State University, Atlanta, GA.Matt McGivern ([email protected]) is atArthur Anderson, Atlanta, GA.Christi Howard ([email protected]) is an undergraduate student in the Department of Computer Information Systems atGeorgia State University, Atlanta, GA.

© 2002 ACM 0002-0782/02/0100 $5.00

c


Early adopters of VoIP include newly establishedproviders seeking to exploit specific markets,

such as the international market for long-distancecalling where traditional calling is expensive andhighly profitable. To take advantage of emergingVoIP markets, several major PC-to-phone commu-nication providers have emerged. The adoption ofVoIP has faced a number of hurdles, including tech-nical differences in telephone systems and con-flicting regulatory paradigms in different countries.Although a completely global PC-to-phone serviceis not commonly available today, most countriescan be reached through such services. PC-to-phoneproviders include:

DialPad (www.dialpad.com), offering severaldifferent types of VoIP services; one of them allows400 minutes of VoIP calls for $9.99 (approximately2.5 cents/minute).

Net2Phone (www.net2phone.com), offering sev-eral options for VoIP service; one of which allowsthe first five minutes of calling for free, thencharges 2 cents/minute.

Go2Call (www.go2call.com), offering up to 15minutes of free calls from PC-to-phones inCanada, the U.K., the U.S., and several other countries.

Delta three (www.iconnecthere.com), offeringPC-to-phone calls within and to the U.S.; one planallows unlimited VoIP calling for $9.95 and another400 minutes for $1.99. c

PC-to-Phone Providers

voice over ip

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