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Voice Over IP   
Voice Over Internet Protocol (VOIP) may seem too advanced for your library right now, however this technology is growing in popularity for Internet and telephone users.
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Voice over IP (VOIP) communication first received wide attention in the late 1990s because of its potential to save money over traditional voice communication, and its ability to simplify IT management by merging voice and data systems. While VOIP is still not widely used in libraries right now, there is a distinct buzz surrounding this technology that your more advanced patrons may be interested in. So here are the basic principles behind VOIP.

Traditional telephone calls use dedicated bandwidth. While this preserves the quality of the call, it uses up more bandwidth than is needed -- bandwidth that could potentially be combined with data traffic over IP (Internet Protocol) lines.

While the first VOIP products and services demonstrated potential, the quality of service was a major factor discouraging widespread adoption, as IP was not built for voice traffic and broadband Internet access was not widespread. However, with advances in the technology and increased broadband Internet access by ordinary users, VOIP technology has come back into focus.

What is VOIP?

VOIP refers to the digital transmission of voice as discrete packets of voice information over Internet Protocol (IP) networks, as opposed to the circuit protocols used in traditional telephone calls. IP is a standard for data transmission for closed groups (an office for example), Wide Area Networks (WANs) or intranets, and the Internet. IP is an efficient way for transmitting data, allowing data heading to various destinations to share bandwidth.

Internet telephony is VOIP technology using the public Internet for transmission. VOIP is often used incorrectly to refer to Internet telephony.

How does VOIP work?

Traditional telephone communication is connected through a Public Switched Telephone Network (PSTN), also referred to as POTS (Plain Old Telephone Service) and “Circuit Switching.” The PTSN creates a direct connection (64k) for each voice call. Only one call can travel on that connection.

VOIP technology (also called “packet switching”) does not use direct connections. Rather, voice data is sent through an IP gateway and is broken down into small packets that contain the destination address. Packets are then sent via IP to their destination, where they are reassembled by the recipient IP gateway and delivered to the recipient of the call. As there is no direct connection, packets with different destinations may follow the same path, and those with the same destinations may follow divergent paths depending on network conditions.

Packet switching transmission allows more data to be sent using less bandwidth than when using direct connections. The same 64k line used to transmit one conversation in the POTS system can carry six or seven conversations using packet switching.

Applications for VOIP

Internet telephony: This is the most popular application of VOIP, with many vendors offering free or inexpensive voice communications via the public Internet. Internet telephony can be implemented as an ASP (Application Service Provider) service or in-house system; configured as a Computer-to-Computer (Microsoft NetMeeting), Computer-to-Phone (Net2phone.com), or Phone-to-Phone system. You can read Online Communication via Fax and Telephone, to find out more about online phone communication.

LAN/WAN voice communication

This is used to replace traditional PSTN systems for inter- or intra-office communications. VOIP implemented over an existing WAN connection or even over a Virtual Private Network (VPN) can save money and require less maintenance than traditional calls or direct ISDN (Integrated Services Digital Network) connections. LAN/WAN VOIP solutions are emerging as the most viable applications for many businesses because they use excessive LAN/WAN bandwidth toward VOIP, making the transition less costly and the result more scalable.

These systems minimally consist of a VOIP gateway, an Internet server, and compatible head/handsets. Some systems require an external computer to serve the Internet and product software, while others build these functions into the gateway and handset hardware. Most of these systems can plug right into an existing network using standard RJ45 ports and network cabling. Examples of vendors who provide these systems include Multi Tech and Cisco.

Fax over IP: This is similar to Internet telephony, without the quality of service issues that hamper voice communications over IP. Example services include Efax and J2 Communications.

Unified messaging: This entails one Web-based application for retrieving and managing e-mail, fax, and voice communications. These services most commonly provide an online service where members can log into their account and read/listen/respond to their messages from the computer. Some services provide a telephone number to listen and respond to all messages (fax and e-mail included). Examples include Onebox, MailboxCity, and MessagePoint.

Conferencing: Tele- and video-conferencing over IP is less expensive than ISDN connections, saving travel time and money.

Benefits of VOIP

Lower hardware and support costs: Instead of implementing and managing separate voice and data networks, libraries can combine these services with an IP network, using the same in-house or outsourced technical skills for maintenance. It also requires less specialized hardware.

Reduced long distance call costs: With Internet telephony, local loop and long distance carrier fees are largely avoided, significantly reducing call costs. This can be especially advantageous in Wide Area Network setups, replacing more costly dedicated ISDN setups between multiple offices with IP systems.

Increased flexibility for application additions: Fully IP based data and voice communications enable you to consider time and cost management applications such as IP conferencing, IP faxing and unified messaging, with reduced overhead cost for implementing each.

Drawbacks of VOIP

Quality of service: A big concern and industry challenge is overcoming quality of service issues with VOIP implementation. Unlike normal data transfers, voice data must be received continuously, at a constant speed. Voice traffic requires “low latency,” i.e., no interruptions in the communication stream, but IP communication was not designed with low latency in mind. With IP, it does not matter if the data comes in non-sequentially or at different speeds. For example, you don't need to open your e-mail one letter at a time. Rather, you wait for the whole e-mail to arrive before opening and reading the message. Along with latency, intermittent packet loss and “jitter” (packets sent at different speeds) are characteristics of IP networks that adversely affect the quality of voice transmission.

While these problems can be managed in a closed environment (such as a Local Area Network), open environments such as the public Internet create problems for voice communication. Because the traffic and the hardware used to route communications cannot be controlled, the Internet is often congested and voice communications degrade due to latency, packet loss, and jitter. Internet telephony calls are frequently subject to fading, static, and complete loss of transmission depending on Internet congestion and speed of the caller's Internet connection.

Standards and interoperability: Although there is convergence around the H.323 standard, a standard for IP gateways developed by the International Telecommunications Union (ITU), other protocols are used as well. Interoperability between different vendors' products and among product lines is not well developed, reducing the potential for scalability of features and services.

For more information on VOIP, please check out these resources:

IPxStream - News and market research on VOIP

Protocols.com - Various articles and references about VOIP

VOIP Watch - News, links, and a host of information on VOIP


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