The Public Switched Telephone Network (PSTN)
The architecture of today's PSTN is a direct descendant of the original manned switchboards of Alexander Graham Bell's day. Voice is transmitted in one way: sampled in 8-bit bytes, 8000 times a second, for an aggregate rate of 64 Kbps. The entire telephone network is designed around this rate and for voice traffic exclusively.
The fact that the entire PSTN was designed for circuit switching of voice calls has made it very difficult to add new services to the network, or to increase the efficiency of traffic handling. The PSTN has slowly evolved over the last 100 years from a mechanical switching fabric with analog circuits to a complex mixture of analog and digital circuits with a variety of signaling techniques. Moore's Law states that the power of microprocessors is doubling approximately every 18 months, but it took AT&T two years to boost just one part of the signal in the circuit-switched PSTN. This shows the slow progress of the PSTN. But now, with the ever-improving technology of Internet infrastructure, telephone service can only get better.
The PSTN uses a circuit-switched architecture in which a direct connection, or circuit, is made between two users. The circuit provides a full-duplex, or bi-directional, connection with extremely low latency, or delay, between the two end points. This connection was once a physical connection but with voice over IP it will be a logical connection through many switches and across a variety of wiring types (twisted-pair, fiber-optic cable, etc.). The users have exclusive and full use of the circuit until the connection is released.
VoIP can be defined as the ability to make telephone calls and to send facsimiles over IP-based data networks. VoIP converts standard telephone voice signals into compressed data packets that can be sent locally over ethernet or globally via an ISP's data network rather than traditional phone lines.
Packet switching is a data transmission technology in which data is assembled into distinct digital "packets" with addresses that are read by switches or routers as the packets are received. The switches/routers forward the packets on to the appropriate destination, but there is not a dedicated circuit connection between the two. In fact, packets from a particular source may take different routes to the same destination, depending on varying network traffic conditions and other factors. This type of transmission is only half-duplex, or unidirectional, which can easily lead to high delays between sending and receiving. This type of network is highly efficient for data, which can be read into memory and reassembled at the destination.
The non-deterministic nature of packet switching, however, means that some packets will arrive out of sequence and that there must be a mechanism to notify the sending device when a packet is "lost" so that it can be re-sent. This will be a challenge, discussed later in the paper, which the equipment manufactures and VoIP service providers will have to overcome in order to deliver a product as simple and reliable as the traditional telephone.
Again, voice will be transmitted over Internet Protocol. Internet protocol is a portion of the TCP/IP suite. TCP/IP was originally developed by the US Department of Defense to link dissimilar computers across many kinds of networks, both LANs and wide area networks (WANs).
Its key feature, providing multi-vendor connectivity, has made it popular among network managers and administrators. The Internet Protocol tracks Internet addresses of nodes, routes outgoing messages, and recognizes incoming messages. In other words, IP provides the addressing needed to enable routers to forward packets across multiple networks. IP provides a connectionless datagram service, which means that it attempts to deliver every packet, but has no provision for re-transmitting lost or damaged packets. IP leaves such error correction, if required, to higher-level protocols such as TCP. IP addresses are 32 bits in length and have two parts: the Network Identifier (Net ID) and the Host Identifier (Host ID). The Net ID is assigned by a central authority and specifies the IP address, unique among all entities across the Internet. The Host ID is assigned by the local network administrator and identifies a particular host, station, or node within a given network, which means that it only has to be unique within the local network.
IP's ability to run over any network medium (Ethernet, FDDI, SONET, ATM, Frame Relay, etc.) has led to its widespread adoption around the world, and it's one of the technologies that have enabled the growth of the public Internet. Its popularity, though, goes far beyond the Internet, to encompass the majority of data networks worldwide.
A key requirement for successful VoIP deployment is the availability of an underlying IP-based network that is capable of supporting real-time telephone and facsimile. Delay, jitter, and unreliable packet delivery - all of which are typical characteristics of the basic IP network service, affect voice quality.
Most of today's data network equipment - routers, LAN switches, ATM switches, network interface cards, PBXs, etc. - will need to be able to support voice traffic. Furthermore, VoIP equipment will either have to be integrated into these devices or work compatibly with them. VoIP equipment must also accommodate environments ranging from private Intranets to the less predictable Internet. Quality of service (QoS) is the key player that will determine the success of VoIP.
This subject will be mentioned various times throughout this paper. Three common techniques need to be used to ensure good QoS.
Also, a very integral factor in determining the success of VoIP will be the supporting equipment. VoIP equipment, categorized into client, access/gateway, and carrier class/infrastructure segments, should be configurable to capitalize on these different techniques but must also be sufficiently flexible to add new techniques, as they become available. Producers that make use of embedded software should focus on how to best utilize the functions instead of focusing on the problems associated with implementing and testing the objects themselves.
Naturally, real-time voice traffic capabilities will give the appearance of the traditional phone. Seamless, uninterrupted conversation will be the only excepted results. Real-time voice can be carried over IP networks in three different ways:
Future VoIP networks will include IP-based PBXs (iPBXs), which will emulate the functions of a traditional PBX.
These will allow both standard telephones and multimedia PCs to connect to either the PSTN or the Internet, providing a seamless migration path to VoIP.
An iPBX can also combine the features of today's switches and routers and could become the gateway into a variety of value-added services such as directories, message stores, firewalls and other network-based servers. Such a VoIP system would also combine real-time and non real-time communications.
Voice and facsimile messaging, for example, use functions that are very similar to a telephone call but do not need the same levels of QoS in the underlying network.
The most important consideration at the network level is to minimize unnecessary data transfer delays. Providing sufficient node and link capacity and using congestion avoidance mechanisms (such as prioritization, congestion control, and access controls) can help to reduce overall delay.
Voice and telephone calling can be viewed as one of many applications for an IP network, with software being used to support the application and interface to the network. The emergence of VoIP is a direct result of the advances that have been made in hardware and software technologies in the early 1990s.
The Voice Processing module, which prepares voice samples for transmission over the packet network. This module must perform the following functions:
Needless to say, the software used in VoIP devices must also be supported by a real-time operating environment and provided with the ability to communicate among the modules and with the external world. Implementation of protocols is another area where development time, testing, and risk can be minimized through the use of embedded software. The objective should always be to develop new ways to optimize the use of standard protocol software, not to re-invent basic functions that require extensive testing for standards compliance and product interoperability.
The deployment of a VoIP infrastructure for public use involves much more than simply adding compression functions to an IP network. Anyone must be able to regardless of location and form of network attachment (telephone, wireless phone, PC, or other device). Everyone must believe the service is as good as the traditional telephone network. Long-term costs must make the investments in the infrastructure worthwhile. Communicating via packet data networks such as IP, ATM, and Frame Relay has become a preferred strategy for both corporate and public network planners. Experts say that data traffic will soon exceed telephone traffic, if it hasn't already. At the same time, more and more companies are seeing the value of transporting voice over IP networks to reduce telephone and facsimile costs and to set the stage for advanced multimedia applications. Providing high quality telephony over IP networks is one of the key steps in the convergence of voice, fax, video, and data communications services.
The Internet and the corporate Intranet must soon be voice-enabled if they are to make the vision of "one-stop networking" a reality. By treating voice as another form of data and sending it over the same network as data, VoIP is enabling new applications that use the best characteristics of voice communications and data processing. As previously mentioned, these applications can include PC-to-PC connections, PC-to-phone connections, and phone-to-phone connections. For ISP’s, merging voice and data on one single network will allow them to expand their services beyond simple information access and into the realm of voice, fax, and virtual private networking.
Potential customers for IP telephony technology and applications include not only those directly involved in the Internet business but also companies in the PSTN carrier space, private networks or intranets, WANs/extranets, and enterprise networks. I will discuss various advantages, challenges and disadvantages, and functionality’s of VoIP throughout this document, even more so in their respective areas. Below are some basic examples of how and why many may choose to invest time and money into VoIP implementation. The idea of making "free" telephone calls over the public Internet has created a lot of excitement.
This approach is widely discussed as the replacement to the PSTN.
For many, the major factor in choosing VoIP is the idea of decreasing long-distance bills.
The Internet right now is a free medium on many networks. If you can send voice over a computer network,
you can conceivably make long-distance or international calls for the cost of a local call. It also creates new
possibilities for remote workers, who can conduct voice and data calls over one phone line and retrieve voice
mail from their PCs for the cost of a local call. The typical VoIP Gateway is designed to sneak their way into the PSTN
by emulating a subscriber who happens to make a cheap long distance phone call for you. The connection is essentially free
if it is a local call. This approach may not last as the RBOCs will force IP telephony service providers to behave as long
distance companies and will move them to trunk-side circuits. The benefits of free local access will then disappear and the
need of true PSTN signaling will cause a shake out in the IP gateway industry.
With the cost of Internet access decreasing and the rapid increase of Internet users, there is less and less of a need for the telephone communication. Don’t get me wrong, nothing beats a standard phone but lets face it,
hours spent surfing the web greatly exceed the amount of hours on the telephone these days. Telephone companies aren’t worried about people not using the phone anymore, but rather less dependency on it.
With Internet companies charging very low flat rates for access, telephone lines (used to dial-up ISP’s servers) are saturated all day, even during so called off-peak hours. Telephone companies, long distance in particular,
are worried about people placing calls via the Internet vice the telephone. By using gateways, explained earlier in the paper, to make long distance calls, long distance companies are out of the loop. Which results in one thing,
profit loss! Long distance service providers aren’t the only ones sweating bullets, the local telco’s are worried, too. In countries like the United States customers pay a flat rate for local telephone access, local carries are
losing out in a sense. That’s why in most countries in the world, they charge on a per call and minute basis. Essentially, customers with flat Internet rates are getting the best of all worlds: an opportunity for continuous
web access, long distance calling, and other options offered via IP based networks.
ISP’s aren’t making their fortunes from simple web access though. In many cases they are drowning or barely staying afloat, trying to offer unlimited service for a flat rate. Web analyst say that only half of the ISP’s
today will survive in the 2000’s. It is expected that the amount of ISP’s will drop from 4000 to 500 globally. Only the reigning companies will see profits their profits through advertisements. Allowing companies to
“advertise” their home pages, ads, etc will generate the profits that offset the losses from offering the unlimited access.
How does this affect the future of VoIP? Suddenly, the long distance companies may try to make VoIP companies function as carriers to ensure they receive the full experience like the rest of them. You know benefits
like, taxation, tariffs, etc. It seems like legal action won’t be taken against VoIP providers anytime soon so they will cross that bridge when they get to it.
Voice communication via traditional mediums will probably be the basic form of interaction for a long time. The PSTN can’t be replaced or dramatically changed in a short period of time. The immediate goal for VoIP service providers is to reproduce existing telephone capabilities at a significantly lower "total cost of operation" and to offer a technically competitive alternative to the PSTN. It is the combination of VoIP with point-of-service applications that shows great promise for the longer term. I mentioned various initial measures of success for VoIP, but long distance cost savings will be the paramount as there are no additional constraints imposed on the end user. VoIP provides a competitive threat to the providers of traditional telephone services that, at the very least, will stimulate improvements in cost and function throughout the industry.
VoIP could be applied to almost any voice communications requirement, ranging from a simple inter-office intercom to complex multi-point teleconferencing/shared screen environments. The quality of voice reproduction to be provided could also be tailored according to the application. For example, customer calls may need to be of higher quality than internal corporate calls. Hence, VoIP equipment must have the flexibility to cater to a wide range of configurations and environments and the ability to blend traditional telephony with VoIP.
Some examples of VoIP applications/capabilities that are likely to be useful would be:
While there are many advantages to implementing VoIP, those industry leaders in the technology must ensure certain areas remain advantageous though ease of use, cost savings and efficient utilization of those maintaining the systems.
Although reducing long distance telephone costs is always a popular topic and would provide a good reason for introducing VoIP, the actual savings over the long term are still a subject of debate in the industry. Flat rate pricing is available with the Internet and can result in considerable savings for both voice and facsimile (at least currently). It has been estimated that up to 70% of all calls to Asia are to send faxes, most of which could be replaced by VoIP. These lower prices, however, are based on avoiding telephony access charges and settlement fees rather than being a fundamental reduction in resource costs. The sharing of equipment and operations costs across both data and voice users can also improve network efficiency since excess bandwidth on one network can be used by the other, thereby creating economies of scale for voice (especially given the rapid growth in data traffic).
An integrated infrastructure that supports all forms of communication allows more standardization and reduces the total equipment complement. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design. The differences between the traffic patterns of voice and data offer further opportunities for significant efficiency improvements.
Since people are among the most significant cost elements in a network, any opportunity to combine operations, to eliminate points of failure, and to consolidate accounting systems would be beneficial. In the enterprise, SNMP-based management can be provided for both voice and data services using VoIP. Universal use of the IP protocols for all applications holds out the promise of both reduced complexity and more flexibility. Related facilities such as directory services and security services may be more easily shared.
End user demand is expected to grow rapidly over the next five years. Frost & Sullivan and other research firms have estimated that the compound annual growth rate for IP-enabled telephone equipment will be 132-149% over the period from 1997 to 2002. The VoIP market size in 1996 was $19.8 M, $47.3 M in 1997 and projected to be $3.16B by 2002. It is expected that VoIP will be deployed by 70% of the Fortune 1000 companies by the year 2000. Industry analysts have also estimated that the annual revenues for the IP fax gateway market will increase from less than $20M in 1996 to over $100M by the year 2000. It is clear that a market has already been established and there exists a window of opportunity for developers to bring their products to market and for investors to take their chances!
For every new development there are pros and cons. The object of the game is for the pros to greatly outweigh the cons, in order for some strategic advantage to be seen. The public telephone network and the equipment that makes it possible are taken for granted in most parts of the world. Availability of a telephone and access to a low-cost, high-quality worldwide network is considered to be essential in modern society and can even be relied on with a power outage. Anything that would jeopardize this is usually treated with suspicion. That’s why there are many challenges, some more difficult that others, that has to be mastered before “society” accepts change.
One of the greatest challenges for VoIP is to develop networks that are not only scaleable but also seamless to the subscriber and to the service provider. If the service is difficult to access by the subscriber due to complex dialing plans and special PIN numbers, or requires significant time to complete a call, or has constant call drops, then the IP gateway will only be used by a limited client base.
If it’s difficult for the service provider to install, administer, settle, and bill, it will have longer ROI and will be less likely to be deployed. Incumbent carriers have very specialized billing systems already in place and are not likely to create an entirely new billing system just for IP telephony.
One of the immediate expectations for VoIP is real-time facsimile transmission. Facsimile services normally use dial-up PSTN connections, at speeds up to 14.4 Kbps, between pairs of compatible fax machines. Transmission quality is affected by network delays, machine compatibility, and analog signal quality. To operate over packet networks, a fax interface unit must convert the data to packet form, handle the conversion of signaling and control protocols, and ensure complete delivery of the scan data in the correct order. For this application, packet loss and end-to-end delay are more critical than in voice applications.
The goal for developers is relatively simple. Add telephone calling capabilities (both voice transfer and signaling) to IP-based networks and interconnect these to the public telephone network and to private voice networks in such as way as to maintain current voice quality standards and preserve the features everyone expects from the telephone.
Several developments in the 1990s, most notably advances in digital signal processor technology, high-powered network switches, and QoS-based protocols, have combined to enable and encourage the implementation of voice over data networks. Low-cost, high-performance DSPs can process the compression and echo cancellation algorithms efficiently. Software pre-processing of voice conversations can also be used to further optimize voice quality. One technique, called silence suppression, detects whenever there is a gap in the speech and suppresses the transfer of things like pauses, breaths, and other periods of silence. This can amount to 50-60% of the time of a call, resulting in considerable bandwidth conservation. Since the lack of packets is interpreted as complete silence at the output, another function is needed at the receiving end to add "comfort noise" to the output.
Another software function that improves speech quality is echo cancellation. Echo becomes a problem whenever the end-to-end delay for a call is greater than 50 milliseconds. Sources of delay in a packet voice call include the collection of voice samples (called accumulation delay), encoding/decoding and packetizing time, jitter buffer delays, and network transit delay. The ITU recommendation G.168 defines the performance requirements that are currently required for echo cancellers.
Engineering a VoIP network (and the equipment used to build it) involves trade-offs among the quality of the delivered speech, the reliability of the system, and the delays inherent in the system. Minimizing the end-to-end delay budget is one of the key challenges in VoIP systems. Ensuring reliability in a "best effort" environment is another. Equipment producers that offer the flexibility to configure their systems to fit the environment and thereby optimize the quality of the voice produced will have a competitive advantage.
For the most part, Challenges for the product developer arise in five specific areas:
The quality of sound reproduction over a telephone network is fundamentally subjective; although standardized measures have been developed by the ITU. It has been found that there are three factors that can profoundly impact the quality of the service.
Two problems that result from high end-to-end delay in a voice network are echo and talker overlap. Echo becomes a problem when the round-trip delay is more than 50 milliseconds. Since echo is perceived as a significant quality problem, VoIP systems must address the need for echo control and implement some means of echo cancellation. If you’ve ever used one of the IP telephony service providers like Dialpad.com, delay is very noticeable and often frustrating when trying to have a decent conversation. One almost has to talk like a “Walkie Talkie” is being used.
Then there’s the problem with talker overlap (one caller stepping on the other talker's speech). It becomes significant if the one-way delay becomes greater than 250 milliseconds. The end-to-end delay budget is therefore the major constraint and driving requirement for reducing delay through a packet network. This is frustrating because one almost has to coordinate when to talk in order for two parties hear one another.
Jitter is the variation in inter-packet arrival time as introduced by the variable transmission delay over the network. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence, which causes additional delay. The jitter buffers add delay, which is used to remove the packet delay variation that each packet is subjected to as it transits the packet network.
IP networks cannot provide a guarantee that packets will be delivered at all, much less in order. Packets will be dropped under peak loads and during periods of congestion. Due to the time sensitivity of voice transmissions, however, the normal TCP-based re-transmission schemes are not suitable. Approaches used to compensate for packet loss include interpolation of speech by re-playing the last packet, and sending of redundant information. Packet losses greater than 10% are generally not tolerable.
The race to create VoIP products that suit a wide range of user configurations has now begun. Standards must be adopted and implemented, gateways providing high-volume IP and PSTN interfaces must be deployed, existing networks need to be QoS-enabled and global public services need to be established. Adoption of VoIP must also remain economically viable even if PSTN prices decrease. Needless to say, developers often underestimate both the difficulties of adding voice to packet networks and the complexities involved in building products suitable for public networks.
Any company with an Internet connection or a corporate Intranet is a potential voice over IP customer. This includes Fortune 100 -2000 companies and countless smaller businesses as well as colleges, universities, government agencies, and non-profit organizations, all of whom have significant needs for long distance and international voice and fax service or other IP telephony applications. All the companies that have started to Implement VoIP have many things in common. First, there are those with physical boundaries. These are the organizations where little to no traditional telephone mediums exist and it would be very difficult or impossible to install those mediums. There will be a good example of this limitation (at the South Pole) in the next section.
Next, there are the companies with an excellent network infrastructure in place. These companies want to use their LAN for more than just data traffic. It has become almost normal for an organization to have huge amounts of bandwidth in their LANs and private networks. Adding voice capabilities allows them to save money and utilize what’s already in place instead of paying expensive [telephone] bills that can be easily reduced.
Also, organizations with a global presence tend to migrate toward a VoIP solution. Their motive is simply to save money. The cost to make calls and send faxes internationally is extremely expensive. These companies have realized that if they can use the “Net” to bypass distance charges and tariffs, then millions can be saved every year.
Based on my research, it appears the ideal environment would be that of an organization where the Intranet or private network can be “controlled” (from a configuration management and bandwidth standpoint). Again, the ideal environment will have an infrastructure with a significant amount of bandwidth and scalability (for potential growth). An internal solution gives administrators more visibility of their network and the ability to a suitable QoS, whereas relying on VoIP as an external or global solution is virtually uncontrollable once data/voice traffic enters the ‘Internet Cloud”.
Many organizations are taking advantage of VoIP technology, specifically those with a high-powered LAN infrastructure but no traditional telephone mediums (copper pairs) installed.
One organization that surmounted their land barriers by implementing VoIP, the United States Antarctic Program (USAP) at the U.S. Amundsen-Scott South Pole station.
The personnel here relied on ham radios and ATS-3 (an ancient NASA satellite originally used to support the Apollo lunar missions) as their primary means of communication. But this was
not enough to meet the expanding requirements of the scientist and support the South Pole. The network engineer on site was challenged to deliver an affordable and reliable alternative
to the archaic devices being used for communicating. Trying to build a traditional phone system with undersea cables and land links would have cost more that $200 million dollars.
The engineer explored various options, but ultimately chose VoIP. Having to overcome the barriers of frozen terrain and inclement weather, satellite communication was used. Instrumental to
the voice over IP setup were three satellites - GOES-3, LES-9 and TDRS F1.With a capable telephone network infrastructure in place, “voice” is transmitted via the previously mentioned satellites
to one of the two ground stations at The University of Miami.
There, the networked telephone calls are connected through the campuses PBX and the South Pole Personnel are able to make calls.
Asynchronous Transfer Mode (ATM) -- (1) The CCITT standard for cell relay wherein information for multiple types of services (voice, video, data) is conveyed in small, fixed-size cells. ATM is a connection-oriented technology used in both LAN and WAN environments.
Circuit-Switched Network -- Network that establishes a temporary physical circuit until it receives a disconnect signal.
Competitive Local Exchange Carrier (CLEC) -- A company that builds and operates communication networks in metropolitan areas and provides its customers with an alternative to the local telephone company.
Compression -- Reducing the size of a data set to lower the bandwidth or space required for transmission or storage.
Computer Telephony Integration (CTI) -- The name given to the merger of traditional telecommunications (PBX) equipment with computers and computer applications. The use of Caller ID to automatically retrieve customer information from a database is an example of a CTI application.
Connectivity -- The ability of a device to connect to another: This includes not only the physical issues associated with the busses, connector topologies, and other such matters, but also the support of the protocols required to pass data successfully over the physical connection.
Dedicated Circuit -- A transmission circuit leased by one customer for exclusive use around the clock. Also called a private line, or leased line.
Dedicated Line -- (1) A communications circuit or channel provided for the exclusive use of a particular subscriber. Dedicated lines are used for computers when large amounts of data need to be moved between points. Also known as a "private line
Delay -- (1) Amount of time a call spends waiting to be processed. (2) Basically, the time the information takes to transit a network or network segment. Differential delay is the difference in transit time between data taking separate transmission paths - for example, inverse-multiplexed T1s employing different routes through T1 networks.
Dial Tone Multi-Frequency (DTMF) -- The set of standardized, superimposed tones used in telephony signaling - as generated by a touch tone pad.
Digital Signal Processor (DSP) -- A high-speed coprocessor designed to do real-time signal manipulation.
Echo Control -- The control of reflected signals in a telephone transmission path.
File Transfer Protocol (FTP) -- (1) An IP application protocol for transferring files between network nodes
Frame Relay -- High-performance interface for packet-switching networks. Considered more efficient than X.25 which it is expected to replace. Frame relay technology can handle "bursty" communications that have rapidly changing bandwidth requirements.
Interexchange Carrier (IXC) or Interexchange Common Carrier -- (1) Any individual, partnership, association, joint-stock company, trust, governmental entity, or corporation engaged for hire in interstate or foreign communication by wire or radio, between two or more exchanges
Internet Protocol (IP) -- A Layer 3 (network layer) protocol that contains addressing information and some control information that allows packets to be routed
Internet Service Provider (ISP) -- (1) Any of a number of companies that sell Internet access to individuals or organizations.
Internet Telephony -- Generic term used to describe various approaches to running voice telephony over IP.
Internetwork -- A collection of networks interconnected by routers that function (generally) as a single network. Sometimes called an internet, which is not to be confused with the Internet.
Latency -- The delay between the time a device receives a frame and the frame is forwarded out of the destination port.
Local Area Network (LAN) -- A network covering a relatively small geographic area (usually not larger than a floor or small building).
MUX -- A multiplexing device. A mux combines multiple signals for transmission over a single line. The signals are demultiplexed, or separated, at the receiving end.
Packet -- (1) A logical grouping of information that includes a header and (usually) user data. (2) Continuous sequence of binary digits of information is switched through the network and an integral unit
Permanent Virtual Circuit (PVC) -- Virtual circuit that is permanently established. PVC’s save bandwidth associated with circuit establishment and tear down in situations
Private Branch Exchange (PBX) -- A small telephone network for customer premises. Provides local connectivity and switching and connections to the wide area voice network.
Protocol -- (1) A formal description of a set of rules and conventions that govern how devices on a network exchange information. (2) Set of rules conducting interactions between two or more parties. These rules consist of syntax (header structure) semantics (actions and reactions that are supposed to occur) and timing (relative ordering and direction of states and events).
Public Switched Telephone Network (PSTN) -- General term referring to the variety of telephone networks and services in place worldwide.
Pulse Code Modulation (PCM) -- Transmission of analog information in digital form through sampling and encoding the samples with a fixed number of bits.
Quality of Service (QoS) -- Measure of performance for a transmission system that reflects its transmission quality and service availability.
Transmission Control Protocol/Internet Protocol (TCP/IP) -- (1) The common name for the suite of protocols developed by the U.S. Department of Defense in the 1970s to support the construction of world-wide internetworks.
Virtual Circuit (VC) -- Logical circuit created to ensure reliable communication between two network devices
Voice Over the Internet Protocol (VoIP) -- The developing standard for transmitting voice signals over the IP based Internet.
Peter Clegg, "The Road to VoIP," Ian Iamont, "Extreme Networking," http://www.cisco.com
http://networkcomputing.com
http://www.thestandard.com
http://www.Lucent.com
http://www.techguide.com
http://www.telephonyworld.com
©OCT 2000;
WEBMASTER, Reginal.Bryant@Ramstein.af.mil
Back to table of contents
Implementing VoIP
The software functionality required for voice-to-packet conversion in a VoIP terminal or gateway are:
The Call Processing (Signaling) module, which allows calls to be established across the packet network
The Facsimile Processing module that demodulates PCM data, extracts relevant information, and packs the scan data into packets.
The Network Management module, which provides management agent functionality, allowing remote fault, accounting, and configuration management to be performed from standard management.
The Packet Processing module, which processes voice and signaling packets, adding the appropriate transport headers prior to submitting the packets to the IP network (or other packet networks).
The Packet Voice Protocol module, which encapsulates the compressed voice and fax data for transmission over the data network. with a sequence number that allows the received packets to be delivered in the correct order.
The Voice Playout module at the destination, which buffers the packets that are received and forwards them to the voice codec for playout.
Some of the functions that are required for a VoIP system include:
Implementations of full-scale VoIP systems must provide all the functionality’s that are usually taken for granted in the PSTN.These functions are:
Accessibility: Telephone systems assume that any telephone can call any other telephone and allow conferencing of multiple telephones across wide areas.
Availability: Sufficient capacity must be available in the VoIP system and its gateways to minimize the likelihood of call blocking and mid-call disconnects. This will be important in environments where the network is shared with data traffic
Interoperability: In a public networking environment all products need to be interoperable if any-to-any communications is to be possible (I.e. through the usage of common software and the interconnection of VoIP to the PSTN)
Reliability: The VoIP network, whether by design or through management, should be fault tolerant with little to no likelihood of complete failure.
Scalability: There is potential for extremely high growth rates in VoIP systems, especially if they prove the equal of the PSTN at much lower cost. VoIP systems must be flexible enough to grow to very large user populations, to allow a mix of public and private services and to adapt to local regulations.
Viability: There has to be an economic advantages for the implementation of VoIP.
To table of contents
Why VoIP?
Examples include:
IXC’s (InterExchange Carriers), both within the U.S. and internationally. AT&T’s Globalnet division is currently offering IP telephony service, MCI has already begun building PC-based web servers which may support IP telephony, and Sprint has announced its Global One service that offers dedicated bandwidth on demand
CAP’s (Competitive Access Providers): there are a multitude of small national and international carriers who offer an alternative to the major carriers for long distance minutes. These CAPS now comprise over 30% of the total international long distance market share. International Callback is one example of a high profit CAP market that can improve their ROI using IP telephony.
RBOC’s (Regional Bell Operating Companies): such as Bell Atlantic, Pacific Bell and US West. While the recent ruling allowing RBOCs to offer long distance services will be in dispute for quite some time, it is natural to assume that RBOCs will look at packet-based technology as they begin to roll out their inevitable long distance applications. Additionally, through their alliances with ISP’s, RBOCs are a natural convergence point for traditional voice communications and data networks.
CLEC’s (Competitive Local Exchange Carriers) Have a strong reason to adopt VoIP technologies into their networks. Most CLECs are focusing on specific client types such as commercial businesses instead of home subscribers. As such, CLECs are growing rapidly in metropolitan regions while not maintaining any network infrastructure between regions. With IP telephony, the CLEC can easily connect their regions together over a private IP network allowing low cost intranet calling as well as fax services. IP telephony gateways can become the glue to merge their metro PSTN networks together instead of relying on other PSTN service provider who charge an exorbitant rate for access. CLECs are a fast growing market with over 2,500 applications for CLEC status with the FCC.
ISP's (Internet Service Providers) There are also hundreds to thousands of single-region ISP’s such, as well as tens of thousands of bulletin board services (BBSs). Many international ISP’s are also offering IP telephony services. Many ISP’s are moving towards CLEC status.
ITSP’s (Internet Telephony Service Providers): Representing a new class of service providers, these companies are building global IP networks specifically designed for low-latency traffic, including voice and fax.
Call centers and larger service bureaus: These organizations will benefit from new business processes that will capture revenue from web-related sources. This is one of the key applications for IP telephony and is discussed in a later section.
Free long distance!
What are telephone companies (Telcos) to do?
To table of contents
Advantages
Interconnection of the Internet to the PSTN can be accomplished using a gateway, enabling a PC-based telephone to access the public network by calling a gateway at a point close to the destination (thereby minimizing long distance charges).
Ordinary telephones (wired or wireless) can be enhanced to serve as an Internet access device as well as providing normal telephony
A small office (or a home office) could gain access to corporate voice, data, and facsimile services using the company's PBX. This may be useful for home-based agents working in a call center.
Calls to the office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from places where the use of the facilities telephone is very expensive (I.e. Hotels).
Internet call center access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online.
One advantage of IP telephony is that it dramatically improves efficiency of bandwidth use for real-time voice transmission, in many cases by a factor of 8 or more. This increase in efficiency is a real long-term driver for the evolution from circuit-switched technology to packet-switched.
Combination of real-time voice communications and data processing, such as web-enabled call centers, collaborative white-boarding, multimedia, telecommuting, and distance learning. Basic telephony and facsimile are the initial applications for VoIP but the longer term benefits are expected to be derived from multimedia and multi-service applications (I.e. Internet commerce solutions combing call centers).
Needless to say, voice is an integral part of conferencing systems that may also include shared screens, whiteboarding, etc. Combining voice and data features into new applications will provide the greatest returns over the longer term.
The final advantage of VoIP is that it is additive to today's communications networks.
IP telephony can be used in conjunction with existing PSTN switches, leased and dial-up lines,
PBXs and other customer premise equipment (CPE), enterprise LANs, and Internet connections.
IP telephony applications can be implemented through dedicated gateways, which in turn can be based on open standards platforms for reliability and scalability.
To table of contents
Disadvantages/Challenges
Within those five areas, product developers need also to focus on areas such a Speech Quality and Characteristics. Providing a level of quality that at least equals the PSTN is viewed as a basic requirement, although some experts argue that a cost versus function quality trade-off should be applied. QoS usually refers to the fidelity of the transmitted voice and facsimile documents, it can also be applied to network availability, telephone feature, and scalability.To table of contents
Ideal environments for Implementation
Who's using VoIP
To table of contents
Glossary
H.323 -- A standard approved by the International Telecommunication Union (ITU) that defines how audiovisual conferencing data is transmitted across networks
Intranet -- A private network inside a company or organization that uses the same kinds of software that you would find on the public Internet, but that is only for internal use. As
To table of contents
References
Network World, March 2000:17
Network World, March 2000:17
To table of contents