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The convergence of voice, video, and data onto a unified IP network gives agencies new options to control communications costs while planning for future technology. Our Voice over Internet Protocol (VoIP) portfolio can consolidate your traffic and eliminate the need for managing two separate networks.
Expect quality transmissions along with significant cost savings and new voice/data applications without the fear of being locked into a given architecture. A global Multi-protocol Label Switching (MPLS) enabled backbone over an intelligent optical IP-based core network provides the interoperability for handling all your applications, old and new.
Designed for flexibility in a LAN, WAN or contact center environment, VoIP is offered as a set of building block elements that you can manage yourself or let us handle from end-to-end.
AT&T continues to invest billions to ensure superiority in its network infrastructure, support systems, and customer experience. Agencies can trust a stable provider like AT&T Government Solutions to deliver on the technologies that scale with your ever-changing workplace.
In the last few years, data networks have been growing at a much faster rate than voice networks, mainly due to the growth of the Internet. Soon the amount of data traffic will exceed that of voice traffic. As a result of this trend, more and more voice is being sent over data networks (Voice over Frame Relay, Voice over IP and Voice over ATM) than data is being sent over voice networks (via V.34 and V.90 modems).
When Frame Relay was introduced in the early 1990s, the data technology was not originally designed to carry voice. Despite valid reservations about the reliability of voice over frames, the promise of "free voice" eventually proved too alluring. Soon users were experimenting with transporting voice over their Frame Relay devices while equipment vendors worked overtime to make the promise of quality voice over Frame Relay (VoFR) a reality.
As the public Internet exploded in the mid-1990s and users began implementing IP-based networks, the call for voice over IP (VoIP) grew louder. Here, too, equipment manufacturers are developing products to enable inexpensive, universal voice over data networks.
Carriers, however, were caught in a dilemma. Could they afford to cannibalize their highly profitable public switched telephone network? Could they not afford to capitalize on the demand for digital voice? The drama is just unfolding.
Although significant progress has been made in engineering packet networks (Frame Relay, IP and ATM) to carry voice as well as data, today's market is demanding a true convergence of these technologies into a single and ubiquitous communications service without being limited by the underlying technology. The next challenge, then, is to develop interconnection and interworking standards in order to deliver voice services ubiquitously over Frame Relay, IP and ATM.
The Nature of the Data Network and its Implications for Voice
The Nature of the Data Network
Frame Relay, IP and ATM are known as packet or cell switching technologies. This is in contrast to the public telephone network, which is a circuit switching technology, designed to carry voice transmissions. Frame Relay and IP insert data into variable-sized frames or packets. ATM chops data into small cells, which facilitates fast switching of data through the network.
The packet switching and cell switching networks perform statistical multiplexing. That is, they dynamically allocate bandwidth to various links based on their transmission activity. Since bandwidth is not reserved for any specific path, the available bandwidth is allotted according to network needs at any particular time. Compare this to the traditional voice (or circuit switching) network, in which a path is dedicated to the transmission for the duration of the call, which is sent in a continuous bit stream. The line is monopolized by a call until it is terminated, even when the caller is put on hold and during periods of silence. Although this guarantees reliable and immediate transmission of voice, it results in very inefficient use of bandwidth. A line that is dedicated to the telephone cannot be utilized by other data even when there are no voice transmissions.
Originally designed to handle bursty data traffic, packet switching networks (except for ATM) are inherently less efficient than the circuit switching network in dealing with voice. To achieve good voice quality, the delay of voice packets across the network must be minimal and fixed. Due to the shared nature of the packet/cell switching network, it might take time for transmissions to travel across the network. A transmission can be delayed because of network congestion. For example, it might "get stuck" behind a long data transmission that delays other packets. Network congestion can also result in dropped packets, which also detrimentally affects the integrity of voice transmissions.
Voice-Enabling the Data Network
Unlike most data applications, voice is very sensitive to delay. Good voice quality provides a faithful recreation of the conversation, with the same tone, inflection, pauses and intonation used by the speakers. Long and variable delays between packets result in unnatural speech and interfere with the conversation. Dropped packets result in clipped speech and poor voice quality. Fax transmissions are even more sensitive to the quality of the transmission and are less tolerant of dropped packets than voice.
One way to deal with the problem of delay and congestion is to add bandwidth to the network at critical junctures. Although this is feasible in the backbone, it is a costly and ineffective solution in the access arena, defeating the "bandwidth sharing" benefits of packet networks. The best solution is to implement mechanisms at the customer premises, access node and backbone which manage congestion and delay - without increasing bandwidth - such as setting priorities for different types of traffic. Therefore, smart access equipment was developed, that could implement procedures to reduce network congestion and the delay of voice packets without adding bandwidth.
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