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An Introduction to Voice over IP
25 January 2005 Kathleen M. Adams Kiran Bhalla Voice over IP has developed significantly during the last few years. It boasts important benefits while inherent challenges in quality are being addressed.
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| Table of Contents
|  | What Is Voice over IP?
|  | Benefits/Limitations of VoIP
|  | Technology Overview
|  | Background
|  | Circuit Switched Telephony
|  | Packet Switched Telephony
|  | Quality and the Packet Switched Network
|  | Internet Protocols and Network
|  | Implementation
|  | PSTN—Public Networks
|  | Clearinghouses
|  | Private Networks
|  | Reduced Equipment Costs
|  | Administrative Costs
|  | Reduced Call Charges
|  | Cost Savings Resulting From Regulatory Framework
|  | Reduced Leased Trunk Line Costs
|  | Increased Functionality
|  | Voice Over the Internet
|  | Standards
|  | Settlement
|  | Coder/Decoder (CODEC)
|  | Integration With the PSTN
|  | Security
|  | The Future
|  | Insight
| List of Tables
| Table 1. | Functionality of TCP/IP Suite Layers |
| Table 2. | Compression Algorithms |
| Analysis
| What Is Voice over IP? Voice over IP (VoIP) permits the movement of voice traffic over an Internet Protocol (IP)-based network. IP is a standard for data transmission based on packet-switching technology. Voice is broken into a series of packets at the transmitting end. The components are then reassembled and decoded at the receiving device. VoIP is often confused with Internet telephony and IP telephony, which are forms of VoIP where voice packets are sent over the Internet, and private IP LANs and WANs, respectively. IP telephony, in particular, includes the call processing functionality inherent with voice conversation, along with support for a range of newer, end-user applications such as personal desktop productivity, multimedia, collaboration and mobility. To make the distinction clear, VoIP relates to the network infrastructure, whereas IP telephony is a larger, more encompassing term used to describe telephony applications in an IP environment (including VoIP). IP has been designed primarily for the transmission of data where delays and occasional loss of data are less critical as packets travel throughout the network. This may not necessarily involve using the shortest possible route or the same route for each packet. However, voice communications is both real time and mission-critical. Any delay can make a call prohibitive and lead to an undesired poor quality of service (QOS). Packet loss can be caused by router congestion that may lead to a loss of portions of words or sentences. Traffic can multiply as the number of routers is increased in the network leading to longer delays. Network jitter can mean that packets don't arrive in sequence leading to unavoidable delays and poor quality of service.
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Benefits/Limitations of VoIP The increased interest in VoIP on the corporate network has been driven by the opportunity to combine data and voice networks onto a single network. The big initial gain is network efficiency. Theoretically, a single network may reduce administration costs by improving the ability to manage a network. In addition a converged network helps businesses to save procurement costs for network infrastructure. VoIP also allows the ability to improve basic telephony services by integrating it with data applications as companies strive to become more competitive. Applications such as unified messaging and virtual call centers help streamline operations and can improve a company's capability to manage its communications more effectively. One significant advantage of VoIP over the circuit switched model is that IP permits voice communications (features, telephone number) to become location-dependent and truly personalized. Critics of VoIP have argued that the benefits of converged networking are not compelling enough to justify the cost of procuring new systems. Some have questioned the virtue in adopting VoIP to bypass an incumbent operator's traditional billing mechanism when the price for long-distance services has fallen dramatically during the last several years. VoIP has not developed to a point where it has taken over circuit switched networks, which have proven prevalent in voice communications for more than a century. Most voice communication today is primarily provided over circuit switched networks—a technology that lends itself to services requiring high quality and minimum delays. However, the future of circuit voice is under threat.
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Technology Overview Background In the mid-1990s, a small Israeli company, Vocaltec, introduced software that made it possible to make telephone calls over the Internet. Provided the receiver and the recipient both used the so-called Internet Phone and were online simultaneously, it was possible to make a telephone call from one PC to another. Sound quality was poor and the conversation was often characterized by long delays. The subsequent development of IP gateways, which serve to bridge the IP and public switched telephone networks (PSTNs), made it possible to make a call from a PC to an ordinary telephone. The introduction of the IP gateway also meant there was no longer a requirement for the call to originate from a PC. It had now become possible to use the conventional telephone to call an operator and a gateway forwards the call onto the IP network.
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Circuit Switched Telephony Nearly all voice traffic is circuit switched and transmitted over a PSTN. The speed with which voice is transmitted is an aggregate rate of 64 Kbps. A direct connection between two connection points provides a permanent 64-Kbps link for the duration of the call. This link cannot be used for any other purpose during this time. PSTN provides low latency (delay) and is bidirectional, allowing for a two-way or full-duplex conversation to take place. The main shortfalls of circuit switching are provided by the inflexibility and inefficiency inherited in the network by requiring a dedicated connection each time.
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Packet Switched Telephony In a packet switched network, data is broken down into packets, each with a destination address. When the packets are transmitted through the network, the addresses are read at each router, or network switch, for forward routing. At the destination, the packets are reassembled and re-sequenced. Depending on congestion levels in the network, packets may take different routes on their way to the destination. Packet switching provides a virtual circuit connection and is generally half duplex. The main difference from the circuit switched network is that there is no dedicated connection. This is a connectionless network, which allows network resources to be used very efficiently as bandwidth can be shared between applications.
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Quality and the Packet Switched Network The quality issues result largely from the design of packet switched networks that were designed to carry data and do not easily carry delay-sensitive traffic such as voice. When an IP network becomes more congested, the quality suffers as the congestion causes packet loss or latency (the delay between one party speaking and the other party hearing). An acceptable voice conversation requires fast and sequential processing of voice packets. Technological developments are proceeding rapidly to address quality issues. Improvements in latency will result from the development of improved VoIP gateways and particularly the deployment of PSTN/IP gateways over private networks where bandwidth use and, therefore, latency, are better controlled. There are efforts under way to design transmission with fewer handoffs. The future development of the Internet itself should also serve to reduce latency and improve quality of service.
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Internet Protocols and Network The common standard for interconnection between technologically diverse networks is the TCP/IP suite of protocols. Protocols are essentially the rules of communication that different types of computers and operating systems need to follow in order to be capable of connecting and communicating with each other. In the TCP/IP stack of protocols, IP is the lower layer, providing transportation of information. TCP is concerned with fragmentation of the message, retransmission if and when needed, acknowledgment of delivery and flow control, and reassembly. Since these are all functions that are necessary for many different protocols, they have been combined into one separate protocol, the TCP. Above TCP are the application-specific protocols, such as Simple Mail Transfer Protocol (SMTP) for e-mail. The protocols tend to be deployed in layers, each responsible for a different function as summarized in table "Functionality of TCP/IP Suite Layers."
Table 1. Functionality of TCP/IP Suite Layers
Layer | Function
--- | ---
Network (Internet) Layer | Handles the movement of packets through the network and manages the routing of the packets from node to node. Examples include the IP, which is a best-effort principle applied without guarantees.
Transport Layer | The software segments the data message to be transmitted into small packets and adds address labels. There are two transport protocols, User Datagram Protocol (UDP) and TCP. TCP provides the added function of verifying the correct and complete delivery of data. It detects lost data and triggers retransmission.
Application Layer | Manages the details related to specific applications. The application program elects the kind of transport needed and passes it to the transport level. Examples include SMTP (for e-mail) and File Transfer Protocol (FTP).
Source: Gartner.
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Implementation The success and benefits of VoIP vary according to where in the network the applications are applied. The initial benefit, the cost factor, has remained significant in driving the technology forward. The ability to bypass established players, regulations and the traditional telecom local loops and access charges is clearly tempting. With a more prudent approach to IT spend, priorities for corporate network managers have changed to cost reduction.
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PSTN—Public Networks Public network operators have long demanded the same solutions to allow them to take advantage of the economics and openness of packet-based technologies while allowing them at the same time to replicate the robustness of the PSTN.
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Clearinghouses Operators such as TeliaSonera, iBasis and GRIC Communications have established clearinghouses allowing them to terminate traffic from other carriers for toll bypass. They have established a global presence by establishing switch sites and interconnections in as many key markets as possible. For clearinghouses, VoIP technology provides two key benefits: First, VoIP technology is highly scalable and reduces initial investments and, therefore, business risks, significantly for those markets or cities where the business potential is uncertain. Second, the use of VoIP technology enables these operators to provide fax services to those markets where international voice services have yet to be deregulated. Fax service is, strictly speaking, a data service, and data services tend not to be covered by national regulations. For some developing markets, fax traffic can represent as much as 70 percent of all international traffic and is, therefore, a highly attractive source of revenue. Carriers such as AT&T and Level 3 have been using clearinghouses to interconnect voice and data via IP backbones to points worldwide. However, there are differences between how clearinghouses use the public Internet. Some clearinghouses use the public Internet to route calls whereas others, such as Net2Phone, use a privately managed network to insulate themselves from traffic spikes that might affect voice quality. Others, however, may use the public Internet and reroute traffic at times of heavy demand over the PSTN.
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Private Networks Implementation in the corporate networks has been driven by two incentives: the potential to reduce costs by transferring intra-company voice traffic to the corporate data networks and the enhanced flexibility offered by converged networks. IP telephony is being widely adopted in the WAN environment, where voice services are often paid for according to usage and distance, or dedicated private line facilities are deployed to deliver quality voice. Adding compressed voice to the WAN should only add an incremental proportion to the requirements and costs of operating the WAN as a converged network (voice and data). However, there are additional costs involved such as ensuring quality of service and reserving enough bandwidth for voice services as data traffic grows exponentially. Many businesses have adopted IP VPN that allows branch offices and remote employees to access a secure corporate network using public infrastructure rather than dedicated (and expensive) point-to-point leased circuits.
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Reduced Equipment Costs Many companies today still operate two networks: one for voice and one for data traffic, for which they own or lease separate equipment. Adding voice to a data network can potentially save costs in procurement and managing two separate networks. In addition, IP technology-related products tend to have shorter product cycles than conventional switching products, leading to a faster falling price trend.
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Administrative Costs The cost of managing one network instead of two will bring economic benefits.
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Reduced Call Charges Long-distance cost savings are most often applied to intra-company calls over the WAN. The costs associated with routing a voice call over the IP network are lower, and additional savings may be made from a reduced number of access lines.
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Cost Savings Resulting From Regulatory Framework In some regulated markets, tariffs for packet switched (data) traffic tend to be lower than for conventional circuit switched (voice) traffic. In other markets, data traffic may not be regulated at all.
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Reduced Leased Trunk Line Costs Most operators today offer integrated voice and data over one circuit such as digital subscriber line (DSL), frame relay or asynchronous transfer mode (ATM) entering into an enterprise. Voice is converted into data packets (IP) and then overlaid onto frame relay circuits. A single circuit saves money by reducing the number of leased lines installed in an enterprise. While this saves costs over a number of trunk lines, it increases the risk of a dual service disruption (voice and data).
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Increased Functionality VoIP can offer increased functionality (customized services features, unified messaging, presence management) and sophistication for applications (such as customer relationship management [CRM] solutions, collaboration and communications management) when combined with other data on the network.
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Voice Over the Internet As a result of the characteristics of the Internet (publicly available with minimum quality controls or guarantees), the quality of voice communications varies considerably when the Internet is used for voice communication. However, there has been a great deal of progress to improve service quality, especially in the telecom-advanced countries where Internet peering and network capacity have been significantly improved. There are a growing number of subscribers for VoIP services, particularly in developing countries where telecom deregulation hasn't progressed as quickly as in other developed markets. The mass-market user base often accepts the quality drawbacks, in return for significant cost reductions on long-distance and international calls; however, this is not an acceptable trade-off for enterprises. These cost reductions are available as international transmissions on the Internet and are not subject to the artificial tariff's regime that currently is in place on the international voice market.
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Standards The process of developing standards for the VoIP segment has been ongoing since the mid-1990s. Early gateways were based on proprietary protocols and were not capable of communicating with gateways on different networks. Gateways also needed to be as robustly scalable as those which controlled the PSTN. The International Telecommunication Union (ITU) recommended the H.323 standard to bridge IP and PSTN networks, enabling interoperability between IP and PSTN gateway vendors. H.323 is an umbrella standard that defines how delay-sensitive traffic, such as voice, receives priority on a network such as a LAN or WAN, thereby providing real-time applications with the QOS through bandwidth management. However, the standard is subject to continuous enhancements. The second version of H.323 set in early 1998 included elements to set QOS through Resource Reservation Protocol (RSVP): increased security, faster call setup and advanced services such as call forwarding and transfer. The standard provides a common protocol, which includes coding and compressing algorithms and call-switching functions. Applications function seamlessly and are network-, platform- and application-independent. While H.323 is adopted to handle "packetization" and handshakes between the PSTN and IP networks, it does not manage large user groups well and is more suited to multimedia applications. One of the major criticisms of the standard is that call setup is relatively slow. Incredibly, even products from different vendors that are so-called "H.323-compliant" do not interoperate because of the different H.323 versions available and the inherent flexibility within the H.323 standard itself. H.323 also lacks network-to-network interface and congestion control mechanisms. Subsequent versions of H.323 were released to address concerns over reliability, scalability and flexibility. H.323 is still the most prevalent standard used in IP telephony; however, the standard is expected to have a short life due to its complexity. The Internet Engineering Task Force (IETF) has developed the Session Initiation Protocol (SIP) specifically designed for the IP environment, as an alternative to H.323. Work on the standard began in 1996, but it wasn't until 1998 that a protocol was first approved by the IETF. SIP has several advantages over H.323 such as faster call setup since calls can be completed between two clients without the need of a call gateway, and SIP is also a multimedia protocol that simplifies the integration of voice with data and video. SIP intelligence for call setup is obtained from the end device such as a SIP handset phone. Call control signaling is run directly at the SIP end devices. The end device sends requests to a caller's client with a simple text command that can be accepted or rejected. SIP is designed to be lightweight with some its design philosophy and architecture inherited from Hypertext Transfer Protocol (HTTP) and Simple Mail Transfer Protocol (SMTP). It is currently suitable for small, portable devices since it is less complex and uses fewer resources than H.232. SIP does have the edge as the long-term VoIP standard of choice because it is easier to implement than H.323. During the last few years, SIP has won endorsements from major U.S. carriers such as Level 3, Qwest and MCI, as well as from software vendors such as Microsoft. MCI Advantage offers a managed VoIP service using primarily SIP handsets. Media Gateway Control Protocol (MGCP) is an alternative standard from the IETF, designed to overcome the shortfalls of H.323, such as scalability. MGCP/Media Gateway Control (Megaco) enables the external control and management of gateways as opposed to embedding control functions within the protocol itself. In August 2000, an agreement was reached by the ITU to supplement the H.323 family of protocols and integrate MGCP into a protocol called Megaco. Megaco, also known as H.248 protocol, separates the call control layer from the call processing function so that gateways are more scalable. Centralized gateways are capable of controlling and performing signaling across a larger number of ports. At present, the market is dominated by gateways and other equipment that support H.323 protocols. However, Megaco is likely to be adopted in some form since it is designed to accommodate connected-oriented media such as ATM and IP. Products have been launched incorporating the global standard; however, the standard is likely to face competition from SIP proponents. Nevertheless, H.323 expertise is more widespread, and interoperability of the protocol has progressed considerably through collaborative forums between manufacturers and operators. While proponents of these three main protocols continue to tout their individual advantages, SIP is being adopted by all of the mainstream IP telephony vendors, a sure sign that is not likely to go away as a successor to H.323.
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Settlement Traditionally, settlements of telephone charges are through bilateral agreements between operators. It is a simple procedure to monitor over which networks a telephone call is transmitted and charge accordingly. Internet telephony does not make it as easy to trace the route of a telephone call. For a long time, Internet telephony had to originate and terminate within the network of one operator. The Open Settlement Protocol (OSP) should provide a solution in the absence of intercarrier agreements. OSP is used by clearinghouses and can route intercarrier connections and provide billing data. OSP can also manage authentication and authorization of callers and provide call detail reports. OSP will institute per-minute settlement for VoIP calls. Benefits include confidentiality of information provided by the Secure Sockets Layer (SSL) encryption. The providers will be capable of sharing traffic and allocating costs in a cooperative standards environment.
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Coder/Decoder (CODEC) A voice coder is the device that converts an analog voice signal into a digital signal. The digital signal is also compressed to reduce bandwidth requirements. Using a hybrid coding technique with complex algorithms, the voice waveform is sampled and the speech parameters are extracted. Thus, in any pre-defined time period, the waveform is assembled by a synthesis technique to closely assemble the original waveform. The best way to reduce latency is to change the voice coding method; however, the trade-off is voice quality vs. bandwidth required. While there is a delay in the voice compression methods used, there is little further delay with decompression regardless of the algorithm used.
Table 2. Compression Algorithms
Algorithm | Rates
--- | ---
G.711 | Pulse code modulation (PCM); specifies the initial analog-to-digital conversion of speech. Speech is transmitted at 64 Kbps—which is considered to be toll quality.
G.723 | ITU algorithm; offers voice transmission with quality at rates of 6.3 Kbps and 5.3 Kbps with 30-ms delay.
G.726 | Adaptive Differential PCM (ADPCM); coding at 40 Kbps, 32 Kbps, 24 Kbps and 16 Kbps, which yields as much as a 4-1 compression ratio.
G.728 | Digitized voice data runs at 16 Kbps with 2.5-ms delay.
G.729 | Algorithm runs at 8.4 Kbps with 10-ms delay and a compression ratio of 8-to-1.
Source: Gartner.
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Integration With the PSTN Despite some similarities, there are fundamental differences in the way signaling takes place in a PSTN and in packet networks based on IP. Signaling is essential to ensure call-related control information necessary to establish, bill and terminate connections. To allow full convergence and seamless integration of functionality between PSTN and IP networks, signaling has to be developed to avoid degradation of services. The signaling used in a PSTN is carried over a different physical network known as Signaling System 7 (SS7). SS7 messages are exchanged in the form of data packets similar to the way IP networks transmit data, on a dedicated overlay network used exclusively for signaling. By using this separate network for "out-of-band" signaling, this ensures that lines are clear and are free prior to setting up calls. IP networks do not use the same type of "out-of-band" signaling. Instead, they use the protocols such as H.323 or SIPs that are not compatible with the PSTN. A new signaling architecture has been required to integrate both networks seamlessly; however, this does not yet exist. A variety of protocols has been proposed to meet PSTN-IP integration, and these include H.323, SIP IPS7 and Megaco to migrate interoperability and signaling issues. Nevertheless, these standards are being improved to protect against packet loss and to allow for the transmission of toll-quality voice services. Until these are resolved, it is likely that Internet telephony traffic will be carried over dedicated IP networks.
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Security With the convergence of the voice and data networks, security of network infrastructure is a key issue. Security has often been cited as an impediment by network managers for implementing VoIP. Network managers have cited fears that their voice servers or IP-PBX could be the target of a Distributed Denial of Service (DDOS) attack. However, VoIP processing servers, gateway and messaging systems can be protected behind firewalls to enforce access policies and protect them from attack. Rigorous firewall policies should be implemented between VoIP end devices and processing servers.
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The Future By its very nature, VoIP is disruptive to legacy voice services. Regulators and legislators find themselves in a difficult situation because as more voice traffic is delivered over IP networks they are losing regulatory control over services/operators and losing tax revenue. Most regulators around the world have viewed IP and the Internet as an enabling technology for cheaper next-generation data and information services. Until recently, regulators have adopted a laissez-faire approach to IP telephony in fear of stifling development of the technology. So far, VoIP has been deemed an "information service" as opposed to regulated voice services. The challenge for regulators is by no means easy given that IP services erase national borders and thereby intrinsically complicating regulation. Issues such as universal service and a funding pool for emergency calls have yet to be resolved. The cost benefit that sparked the initial incentive will not be long lasting with PSTN charges falling rapidly. Service providers will, therefore, need to differentiate their services from those offered on the PSTN relatively soon and make better use of what IP networks are good at: a more flexible environment for voice and data. VoIP offerings are increasingly prevalent; however, enterprises are often adopting the service with careful considerations for risk management. Some of the benefits for VoIP for the enterprise include increased flexibility, infrastructure consolidation, reduced management and administration, as well support for advanced features. However, there are some drawbacks which may involve upfront infrastructure investment, and there may be interoperability issues between vendors. Nonetheless, there is commitment to the VoIP industry from all major vendors and operators. This commitment will result in the resolution of many of the issues discussed. In the short term, the industry still lacks a compelling story regarding total cost of ownership and feature benefits to entice enterprises to move more rapidly toward IP telephony. Beyond the short term, the small user shall continue to benefit from inexpensive long-distance call charges and calling cards. However, VoIP, in conjunction with the advancement of integrated services, will particularly benefit the corporate segment.
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Insight VoIP has developed significantly in the last few years. Offerings from telecom operators have become more prevalent. The key advantages of VoIP for enterprises have been the promise of lower costs, infrastructure consolidation and increased flexibility. Carriers are also positioning VoIP as a cost-saving business solution. However, deploying VoIP requires careful consideration as the industry is not without its challenges on standards and interoperability as the technology continues to develop. Businesses should not approach VoIP in the same way as traditional telephony due to the former's shared use of infrastructure and the inherent problem this brings with variances in load and usage. Carriers and equipment vendors often neglect to highlight that while VoIP may exhibit cost savings, this frequently requires businesses to upgrade their infrastructure. Businesses need to choose protocols carefully, including the choice of various protocols for their data networks. They should look for vendors that promote open standards and support multiple protocols with their products so that systems are not prone to become proprietary as conferred by legacy voice systems.
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