Upkar Varshney, Andy Snow,
`Matt McGivern, and Christi Howard
`
`Voice
`Over IP
`
`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
`
`others as Internet telephony service providers
`(ITSPs) is likely to further increase competition
`among all phone service providers. Many commu-
`nication technology vendors are rolling out hybrid
`IP/PBX systems. Both traditional and recently
`established carriers are beginning to offer voice over
`IP (VoIP) network connectivity to both business
`and residential customers (see the sidebar “PC-to-
`Phone Providers).
`VoIP involves sending voice transmissions as data
`packets using the Internet Protocol (IP), whereby
`the 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 or
`public IP networks and reassembled and decoded
`
`on the receiving side (see Figure 1). Residential cus-
`tomers can connect to IP-based networks by using
`the local loop from the PSTN or high-speed lines,
`including ADSL/DSL and cable modems.
`Several recent industry surveys and projections
`estimate that VoIP could account for over 10% of
`all voice calls in the U.S. by 2004. It’s likely to be
`used first in places with significant IP infrastructure
`or where cost savings are significant; an example
`might be a company with multiple sites worldwide
`connected through a private or public IP network.
`However, VoIP deployment may not be possible
`everywhere, as some countries restrict the use of
`VoIP to prevent harming their monopolistic
`telecommunication markets. VoIP might also be
`suitable for highly distributed companies or for
`companies with seasonally variable voice-service
`demand.
`The idea of VoIP, or voice over the Internet or IP
`telephony, has been discussed since at least the early
`1970s [6] when the idea and technology were devel-
`oped. Despite this history, VoIP didn’t establish a
`commercial niche until the mid-1990s. This grad-
`
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`Samsung Exhibit 1027
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`

`

`ual commercial development can be attributed to a
`lack 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 poor
`quality 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
`
`Figure 1. A possible scenario for VoIP
`for business customers.
`
`A/D
`
`Compression
`
`D/A
`
`Decompression
`
`Packet
`Assembly
`
`Packet
`Disassembly
`
`Packet
`Switch
`
`LAN or Full
`Duplex Line
`
`VoIP Telephone
`
`well as Internet access. Unfortunately, it did not
`allow traditional calls through the PSTN.
`Following the rapid growth of the public mass-
`market Internet, especially the Web, during the
`early 1990s and accompanying investment in IP
`networking 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 the
`adoption of VoIP technology.
`First and foremost, the cost of a
`packet-switched network for
`VoIP could be as much as half
`that of a traditional circuit-
`switched network (such as the
`PSTN) for voice transmission
`[9]. This cost saving is a result
`of the efficient use of bandwidth
`requiring fewer long-distance
`trunks between switches. The
`traditional circuit-switched net-
`works, or the PSTN, have to
`dedicate a full-duplex 64Kbps
`
`The Internet
`
`Public IP Carrier
`
`Private IP Network
`
`Table 1. A qualitative comparison of voice over PSTN and over IP.
`
`Concept
`
`Voice over PSTN
`
`Voice over IP
`
`Switching
`
`Bit rate
`
`Latency
`
`Bandwidth
`
`Cost of
`access/billing
`
`Circuit switched (end-to-end dedicated circuit set up
`by circuit switches)
`
`Packet switched (statistical multiplexing of several
`connections over links).
`
`64Kbps pr 32Kbps
`
`14Kbps with overhead*
`
`< 100ms
`
`Dedicated
`
`200–700ms depending on the total traffic on the IP net-
`work. Lower latencies possible with private IP networks.
`
`Dynamically allocated
`
`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.
`
`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.
`
`Equipment
`
`Dumb terminal (less expensive); intelligence in the
`network
`
`Integrated smart programmable terminal
`(expensive); intelligence not in the network.
`
`Additional features
`and services
`
`Requires reprogramming or changes in the network
`design but fast enough to add if advanced intelligent
`networks (AIN) are in use.
`
`Easy to add without major changes, due to flexible protocol
`support, but standards are needed for traditional user
`services.
`
`Quality of service
`
`High (extremely low loss)
`
`Low and variable, but traffic is sensitive depending on
`packet loss and delay experienced.
`
`Authorization
`and authentication
`
`Only once when the service is installed
`
`Potentially required, per-call basis
`
`Regulations
`
`Many at federal and state levels
`
`Few yet, but regulatory uncertainty; future regulations may
`reduce the cost advantages of VoIP.
`
`Network availability
`
`99.999% up time
`
`Level of reliability is not known.
`
`Electrical power failure
`at customer premises
`
`Not a problem; powered by a separate source from
`phone company.
`
`Will have problems, as equipment may be down. Power
`from other sources is not easy to obtain.
`
`Security
`
`High level of security because one line is dedicated
`to one call.
`
`Possible eavesdropping at routers.
`
`Standards/status
`
`Mature (simplified interworking among equipment
`from different vendors).
`
`Emerging possible problems in interworking.
`
`*Only when speaker is talking
`
`90
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`

`

`Figure 2. Managing temporary overflow
`of calls using VoIP.
`
`Customer
`service
`center
`
`PBX
`
`Calls from
`customers
`
`Overflow calls
`
`Hybrid
`IP/PBX
`
`LAN/WAN connection
`
`Call center
`with IP phones
`
`Call center
`with IP phones
`
`Table 2. The delay factors in VoIP.
`
`efit existing telephone companies and cable
`providers by increasing the potential number of
`ADSL and cable modem subscribers nationwide.
`Long-distance carriers in the U.S. pay an average
`of $0.0171 per minute in interstate access charges to
`the regional Bell operating companies, that is, the
`local phone companies [8], a total of $9 billion a year.
`One current VoIP cost advantage is that ISPs pay no
`access charges, due to a U.S. Federal Communica-
`tions Commission exemption under enhanced-
`service-provider regulations. However, any changes in
`regulation requiring ISPs and ITSPs to pay access
`charges or treat calls to ISPs as long-distance calls may
`diminish the VoIP cost advantage.
`One VoIP application might involve managing
`temporary overload call volume
`for business users. Using a regular
`PBX, most traffic can be serviced
`with existing telephony equip-
`ment, and any excess or overload
`traffic can be routed to an IP/PBX
`system that can then be serviced
`by remote call centers with IP
`infrastructure (see Figure 2).
`
`Length of Delay
`
`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 in
`bits/sec.
`
`Fixed time for a given length of the segment.
`
`Cause of Delay
`
`Processing at a switch/router
`
`Transmission time, or time to put
`packets online.
`
`Propagation delay, or the actual time it takes
`the 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, and
`decompression.
`
`5–10msec per packet.
`
`channel for the duration of a single call. The VoIP
`networks require approximately 14Kbps, as voice
`compression is employed, and the bandwidth is used
`only when something has to be transmitted. More
`efficient use of bandwidth means more calls can be
`carried over a single link, without requiring the car-
`rier to install new lines or further augment network
`capacity; Table 1 compares voice over PSTN and
`over IP.
`Besides cost savings and improved network uti-
`lization, VoIP offers other features, including caller
`ID and call forwarding, that can be added to VoIP
`networks at little cost [5]. VoIP allows Internet
`access and voice traffic simultaneously over a single
`phone line. This function could eliminate the need
`for two phone lines in a home, one for data and one
`for voice, by using the same line to carry all traffic
`without concern for missed calls or being discon-
`nected from the ISP. Other high-speed media, such
`as ADSL and cable modems, can be used to carry
`both data and voice to IP networks while letting
`home customers use regular phone lines for voice
`calls to and from the PSTN. In this way, VoIP ser-
`vice offered by ISPs and ITSPs might indirectly ben-
`
`Variable, depending on traffic on routers
`and switches in the IP network.
`
`Technical Issues
`Among the many technical issues
`in VoIP, a major one is end-to-
`end delay, or latency. To ensure
`good 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 geosynchronous
`satellites; users deemed the 270-millisecond delay
`unacceptable. However, under certain circum-
`stances, VoIP might suffer from more latency, lead-
`ing to unacceptable quality (due to the uncertainty
`as to whether the other person is talking, possibly
`leading to interruptions). Latency is influenced by a
`number 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 and
`other factors, while increasing overhead on the sys-
`tem. Latency is also related to the number of routers
`and gateways that packets have to travel through
`before reaching their destinations. Table 2 outlines
`the four most common causes of packet delay over
`IP-based networks, public and private.
`Some VoIP systems send test messages to several
`routers 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, especially
`over public IP networks like the Internet, due to
`
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`

`SS7
`Network
`
`Signaling
`Gateway
`
`Core Packet
`Network
`
`Call
`Connection
`Agent
`
`Billing
`Agent
`
`specialized equipment, Internet
`telephony gateways can be used
`where two users communicate
`without having a computer at
`either of their locations. A gate-
`way’s basic architecture involves
`a user connection via the
`PSTN. The gateway computer
`then searches for another gate-
`way computer near the target
`location and makes a connec-
`tion using circuit switching.
`When this connection is made,
`the second gateway utilizes the
`local PSTN to complete the
`collection of the call. Though
`this type of call isn’t completely
`IP, it does suggest possible
`future solutions for integrating
`the current PSTN and the VoIP
`system. Table 3 compares four
`implementations for supporting
`VoIP.
`However, data packets travel-
`ing through the Internet may
`not be secure and may require
`encryption, adding overhead by
`increasing the necessary bit rate
`beyond 14Kbps, hence reduc-
`ing the bit rate advantage of
`VoIP over PSTN. Encryption
`also increases the end-to-end
`latency caused by the processing
`delay
`for
`encryption
`and
`decryption.
`Meanwhile, technology sup-
`port for VoIP has begun to
`mature on a number of fronts.
`The newer generations of routers
`and switches are faster and better
`able to handle the added load of
`real-time data packets. Beyond the advances in com-
`pression and equipment, protocol support in the
`form of the Resource Reservation Protocol (RSVP)
`and IP version 6 (IPv6) are also starting to mature.
`These protocols offer ways to prioritize voice traffic
`over the Net, helping improve QoS, especially when
`the network is congested.
`
`Figure 3. Proposed evolution path for traditional PSTN carriers.
`
`Proposed Voice
`Over Packet Infrastructure
`
`Current
`PSTN Infrastructure
`
`800
`DB
`
`LNP
`DB
`
`Circuit
`Switch
`
`Circuit
`Switch
`
`Trunk
`Gateway
`
`Access
`Gateway
`
`PBX
`
`Traffic
`
`PBX
`
`Control
`
`Table 3. Some VoIP implementations.
`
`Approach
`
`Description
`
`Pros
`
`Cons
`
`Example
`
`PC Web
`phones
`
`VoIP
`gateway
`
`Public IP
`voice
`carriers
`
`Software that
`allows any PC with
`a sound card and a
`microphone to
`transmit voice to
`similarly equipped
`machines.
`
`Used between the
`PBX and an IP
`network/LAN,
`translating and
`routing the calls to
`other gateways.
`
`Phone companies
`that completed
`bypass the PSTN
`and provide just
`VoIP. May still need
`the local loop.
`
`Voice-
`enabled
`browsers
`
`Combining voice
`access with Web
`browsers.
`
`Only cost is
`computer and
`connection to ISP.
`
`QoS issues, along
`with the requirement
`that both users have
`similar equipment/
`software.
`
`Vocaltec
`(www.vocaltec.com)
`and Net2phone
`(www.net2phone.com)
`
`Cost savings in local
`and long-distance
`calls, better
`utilization of network
`resources.
`
`QoS issues need to
`be addressed; high
`initial coast of the
`gateway equipment.
`
`Currently immune
`from line-access
`charges, cheaper
`phone services,
`more advanced
`features; reduced
`infrastructure costs.
`
`Not as reliable as
`PSTN; QoS issues
`still need to be
`resolved; only
`available in limited
`areas; future
`regulation may
`affect.
`
`Good for services
`like live customer
`service for Web sites
`and e-commerce
`solutions.
`
`Regular dialup
`connections limit
`bandwidth available
`for the combined
`services.
`
`Quicknet
`(www.quicknet,net)
`
`Net2phone
`(www.net2phone.com)
`allowing PC-to-phone
`calls (not the other
`way round) at low
`rates
`
`Both Netscape and
`Explorer have plug-ins
`available.
`
`possible rapid fluctuations in the amount of traffic
`and resulting increase in delays experienced by the
`people speaking and listening on the line.
`VoIP systems use the User Datagram Protocol
`(UDP) as a transport layer protocol on top of IP to
`avoid acknowledgments for lost packets. Acknowl-
`edgments trigger undesirable retransmission of
`voice 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; for
`example, many voice encoders can handle up to 1%
`packet loss [2].
`For users who prefer traditional telephones, not
`
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`
`Protocol Support
`Just as in conventional telephony, VoIP needs a
`connection between users, though in the case of
`VoIP, a virtual connection. VoIP architecture
`involves many components. First, a signaling proto-
`
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`

`

`Figure 4. Possible coexistence scenario for PSTN and VoIP.
`
`col is needed to set up individual sessions for voice
`connections between users [2]. Once a session is
`established, a transport protocol can be used to send
`the data packets. Directory access protocols are
`another important part of VoIP, providing routing
`and switching information for connecting calls.
`A signaling protocol handles user location, ses-
`sion establishment, session negotiation, call partici-
`pant management, and feature invocation. Session
`establishment is invoked when a user is located,
`allowing the call recipient to accept, reject, or for-
`ward the call [6]. Session negotiation helps manage
`different types of media, such as voice and video,
`transmitted at the same time. Call participant man-
`agement helps control which users are active on the
`call, allowing for the addition and subtraction of
`
`Hybrid
`IP/PBX
`
`VoIP carrier
`
`Private IP
`networks
`
`The Internet
`
`PSTN
`
`users. The signaling protocol also involves feature
`invocation, at which time call features, such as hold,
`transfer, and mute, are controlled.
`The Realtime Transport Protocol (RTP) can be
`used to support the transport of real-time media,
`including voice traffic, over packet networks. RTP-
`formatted packets contain media information and a
`header, providing information to the receiver that
`allows the reordering of any out-of-sequence pack-
`ets. Moreover, RTP uses payload identification to
`place an identifier in each packet to describe the
`encoding of the media so it can be changed in light
`of varying network conditions [7]. The Real Time
`Control Protocol (RTCP), a companion protocol
`for RTP, provides QoS feedback to the sending
`device, reporting on the receiver’s quality of recep-
`tion. The Real Time Streaming Protocol (RTSP)
`can be used to control stored media servers, or
`devices capable of playing and recording media
`from the server. This added RTSP-based control
`allows the integration of voice mail and prerecorded
`conference calls in VoIP environments. The ability
`to integrate these advanced services is important to
`the future growth of VoIP. The Session Initiation
`Protocol (SIP) can be used to establish, modify, and
`
`Hybrid
`IP/PBX
`
`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 the
`H.323 recommendation from the International
`Telecommunication Union and Media Gateway
`Control Protocol (MGCP) from a branch of the
`Internet Engineering Task Force. Neither is a
`standalone protocol but relies on other protocols to
`complete their jobs [1]. The H.323 architecture is
`based on four components: terminals, gateways,
`gatekeepers, and the multipoint control unit
`(MCU). Gateways are used for protocol conversion
`between IP and circuit-switched networks. Gate-
`keepers are used for bandwidth management,
`address translation, and call
`control. H.323 provides a foun-
`dation for audio, video, and
`data communications across IP-
`based networks, including the
`Internet. Complying with
`H.323 enables different multi-
`media products to interoperate.
`H.323 depends on other stan-
`dards, such as H.245, to negoti-
`ate
`channel
`usage
`and
`capabilities, modified Q.931 for
`call signaling and call setup,
`Registration Admission Status for communicating
`with a gatekeeper, and RTP/RTCP for sequencing
`audio/video packets. The MCU supports multicast
`conferences among three or more end points by
`using H.245 negotiations to determine users’ com-
`mon capabilities [1].
`The Media Gateway Control Protocol (MGCP)
`defines communications among call agents (media
`gateway controllers) and telephony gateways. Call
`agents have the intelligence for call control and
`other functions and manage telephony gateways
`used for protocol conversion. A call agent in MGCP
`is analogous to a gatekeeper in H.323 [1]. The
`MGCP can use the Session Initiation Protocol
`(SIP), which uses the HTTP format to allow a user
`to initiate a call to be initiated by clicking on a
`browser.
`Although H.323 and the MGCP have been stan-
`dardized by two different standard-setting bodies,
`some of their functions are quite similar. Both the
`gatekeeper in H.323 and the call agent in MGCP
`manage and control gateways and participate in set-
`ting up, maintaining, and terminating the VoIP’s
`telephone connection. The MGCP can also be used
`as part of H.323 for simplified interworking.
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`

`PSTN and VoIP
`The PSTN has served the needs of businesses and
`consumers worldwide for more than 100 years and
`has gone through major technological advances,
`including survivable long-distance networks based
`on 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 increasingly
`advanced features and components have increased
`the reliability of the PSTN; it is estimated that
`because of them the PSTN is today operational
`99.999% of the time. The PSTN also offers low
`latency rates and very high quality during voice
`transmission.
`With the emerging potential of IP networks to
`provide integrated voice-data communications,
`conventional PSTN carriers realize they have to
`respond to this competitive threat. For example,
`Telcordia (formally Bellcore) has developed a Voice
`over Packet (VoP) architecture and has initiated an
`industrywide effort to develop generic requirement
`documents; they will allow local and interexchange
`carriers, vendors, and other stakeholders to address
`interoperability issues associated with networks,
`services, protocols, and equipment. These initia-
`tives recognize that because bundled services cannot
`be offered cost-effectively by separate networks,
`they have to identify a migration path for PSTN
`carriers preserving their investment in circuit-
`switched technology and services. This migration
`path is supposed to allow PSTN carriers to modify
`and add only some components in existing net-
`works for offering multiple services, including VoIP.
`Telcordia’s Next Generation Network and VoP
`architecture (NGN/VOP) represents a vision for the
`coexistence 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 of
`circuit-switched calls through circuit switches and
`trunks. The SS7 network also allows other services
`to be provided, including 800-number dialing and
`local 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 number
`of elements [3]:
`
`Core packet network. Unlike the PSTN, this classi-
`cal IP network carries both control and traffic
`packets.
`Call connection agent (CCA). This software provides
`call-processing functionality. An IP network is a
`
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`
`best-effort packet delivery service; something has
`to set up, manage, and disable virtual voice con-
`nections. Moreover, packets might be lost in error
`or arrive out of sequence. The routing and man-
`agement of a virtual call across a core network is
`essential. 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 bridge
`between the circuit-switched and packet-
`switched worlds needed to manage end-to-end
`calls 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 alternative
`access for subscribers not traversing the PSTN.
`The access gateway sets up transport connections
`through the core network when directed by the
`CCA; it also provides ringing and other func-
`tions.
`Billing agent. This agent gets raw usage data from
`the CCA and generates formatted messages for
`back-end billing platforms.
`
`This architecture allows existing PSTN to evolve
`into a network supporting both traditional and IP-
`based voice communications. Though the phone
`companies 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 or
`increase their existing client base. The fate of this
`evolutionary architecture depends on carriers being
`able to forge interoperability consensus among
`themselves and with vendors.
`
`VoIP Adoption and Prospects
`Several factors regarding the adoption of VoIP make
`it difficult to forecast adoption rates. The first deals
`with how quickly existing carriers might transition
`away from their current technology. Another deals
`with demand for services from emerging carriers and
`other service providers who are unencumbered by
`sunken investment in the PSTN. Another deals with
`the regulatory environment. And yet another deals
`with users who will undoubtedly demand not only
`the same high QoS to which they are accustomed
`but cost-effective bundled services as well.
`Many users resist changing to VoIP until they are
`shown the new service’s tangible benefits, including
`reduced cost or more features; they are certainly
`unlikely to accept lower quality. In addition, many
`organizations have invested a great deal of money in
`PBX and other phone equipment. The availability
`
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`

`

`Issues
`
`Cost factor
`
`Table 4. Factors affecting VoIP adoption.
`
`Comments
`
`Cost of existing or legacy infrastructure
`Cost of upgrading
`Cost of access
`Cost of management
`
`Quality
`
`Possible improvements with provisioned bandwidth
`Better quality with private IP networks
`Possible use of IPv6
`
`Equipment
`availability
`
`Emergence of hybrid (IP/PBX) equipment (IP/PBX available in 1,000-line range, possibly scalable to10,000 lines). Also
`possible to link several IP/PBX units together, but still not comparable to many 50,000-line switches employed in PSTN.
`
`Regulations
`
`Possible change in regulations may affect access cost for users to ITSPs.
`
`Global connectivity
`
`Different countries have different views on Internet access and IP telephony; differences in quality and level of
`infrastructure.
`
`Network
`management
`
`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.
`
`Interworking with
`diverse networks
`
`Differences in control signaling and features may have to be addressed; infrastructure and interworking effect on QoS
`should be considered.
`
`Future pricing and
`revenue sharing
`
`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.
`
`Possible effect on traffic
`volume in IP networks
`
`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.
`
`Security and hacking
`threats
`
`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.
`
`Reliability and
`failure issues
`
`Methods of PSTN reliability (such as fault-tolerance, hot standby, redundancy) should be incorporated in VoIP
`networks, both public and private IP.
`
`User equipment
`requirements
`
`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.
`
`Service integration
`(voice and data)
`
`Bundled services can be provided with VoIP networks; cost savings and effect of network failure on all services should
`be considered.
`
`of new hybrid PBX/VoIP systems, which can be
`installed as old equipment is phased out might sig-
`nificantly influence the speed of VoIP adoption. The
`cost today of VoIP end-user equipment is much
`greater than for traditional phones. However, the
`emergence of devices that do not require a computer
`but connect to existing phones may help increase
`user acceptance (see www.phoneworld.net/aplio/).
`VoIP also has to address the issue of security for
`transmitted messages before it can become univer-
`sal. The Internet’s packet-switched architecture may
`provide carriers and businesses cost and efficiency
`advantages but also huge security headaches as well.
`Along with IPv6, many versions of VoIP software
`have built-in encryption, offering better security
`than older implementations. Table 4 lists several
`factors that could affect VoIP adoption.
`These issues make it evident that VoIP will not
`completely eliminate but rather integrate with and
`work in parallel with the traditional established
`PSTN. Even though the two systems reflect quite
`different 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 the
`end user. Additionally, VoIP relies on the PSTN to
`enable its users to reach their ISPs and Internet gate-
`way servers. The two systems are likely to coexist for
`the foreseeable future, each one serving a particular
`market or purpose. This competitive coexistence
`should continue until VoIP quality and reliability
`finally catches up to PSTN, and some of the older
`PSTN architecture becomes outdated and needs to
`be replaced. Figure 4 shows one possible scenario for
`PSTN and VoIP coexistence for customers.
`The Cahners In-Stat group estimates that VoIP
`gateway sales will reach $4 billion in sales in the
`U.S. in 2003. As a harbinger of VoIP deployment,
`Cisco Systems has many business customers with
`more than 2,000 IP phones [4]. Moreover, many
`other small but technologically advanced companies
`are likely to install IP/PBX systems; Gartner Group
`predicts that 50% of all small companies will have
`IP/PBXs by 2004. One major factor influencing
`would-be commercial customers is the ability of
`vendors to offer large IP/PBX systems that match
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`large PSTN switches in terms of cost, size, number
`of lines, reliability, and configurability. The last-
`mile issue can be resolved if carriers offer high-
`bandwidth service aggregation points at business
`customers’ premises. Due to the perceived unman-
`ageability of public IP networks, it’s unlikely that
`most VoIP traffic will be carried by public IP net-
`works in 2004. According to some estimates, it’s
`likely to be less than 20% even by 2004 [1].
`As VoIP gains a commercial foothold, wireless
`VoIP might emerge as a way to transmit voice over
`the Internet from cell and personal communica-
`tions services (PCS) phones. This advance could
`affect cellular and PCS providers as their customers
`gain the option of connecting to IP networks for
`long-distance calls. Since the number of wireless
`customers is increasing exponentially and the cost
`of wireless long-distance service remains high, the
`effect of WVoIP on wireless carriers may be signifi-
`cant, despite the hurdles of QoS and reliability. Bet-
`
`PC-to-Phone Providers
`
`Early adopters of VoIP include newly established
`
`providers seeking to exploit specific markets,
`such as the international market for long-distance
`calling where traditional calling is expensive and
`highly profitable. To take advantage of emerging
`VoIP markets, several major PC-to-phone commu-
`nication providers have emerged. The adoption of
`VoIP 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 service
`is not commonly available today, most countries
`can be reached through such services. PC-to-phone
`providers include:
`DialPad (www.dialpad.com), offering several
`different types of VoIP services; one of them allows
`400 minutes of VoIP calls for $9.99 (approximately
`2.5 cents/minute).
`Net2Phone (www.net2phone.com), offering sev-
`eral options for VoIP service; one of which allows
`the first five minutes of calling for free, then
`charges 2 cents/minute.
`Go2Call (www.go2call.com), offering up to 15
`minutes of free calls from PC-to-phones in
`Canada, the U.K., the U.S., and several other
`countries.
`Delta three (www.iconnecthere.com), offering
`PC-to-phone calls within and to the U.S.; one plan
`allows unlimited VoIP calling for $9.95 and another
`c
`400 minutes for $1.99.
`
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`
`ter loss algorithms and transmission equipment are
`also needed before WVoIP becomes an engineering
`and commercial reality.
`
`Conclusion
`Our aim here has been to provide background infor-
`mation, major concepts, and issues concerning the
`technology, deployment scenarios, and approaches to
`protocol support for VoIP. We’ve also addressed a
`number of unresolved en