Multiplex communications – Communication techniques for information carried in plural... – Adaptive
Reexamination Certificate
2000-07-26
2004-05-04
Vanderpuye, Kenneth (Department: 2661)
Multiplex communications
Communication techniques for information carried in plural...
Adaptive
C370S395100, C370S395520
Reexamination Certificate
active
06731649
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of Invention
The present invention relates generally to the field of communications. More specifically, the present invention is related to a system and method for transferring TDM data over packet switched networks, such as IP networks.
2. Discussion of Prior Art
T-1 (DS1) trunks are circuit switched data networks supporting data rates of 1.544 Mbits per second. A T-1 trunk can carry 24 individual 64 Kbits per second channels, each of which may carry data or telephony quality voice. Similarly, E1 trunks are circuit switched data networks supporting data rates of 2.048 Mbps (32 channels at 64 Kbps). T-3 and E3 trunks support data rates of 44,736 and 34,368 Kbps, respectively. Together T1, E1, T3, E3 and similar circuit switched serial networks are known as TDM networks.
TDM, short for Time Division Multiplexing, is a type of multiplexing that combines data streams by assigning each stream a different time slot in a set. TDM repeatedly transmits a fixed sequence of time slots over a single transmission channel. Within T-Carrier systems, such as T-1 and T-3 (DS3), TDM combines Pulse Code Modulated (PCM) streams created for each in conversation or data stream.
ATM, short for Asynchronous Transfer Mode, represents a network technology based on transferring data in cells of a fixed size. The cell used with ATM is relatively small (53 bytes) compared to units used with older technologies. The small, constant cell size allows ATM equipment to transmit video, audio, and computer data over the same network, and assure that no single type of data hogs the line.
Current implementations of ATM support data transfer rates of from 1.544 (T1) to 622 Mbps (megabits per second). This compares to a maximum of 1000 Mbps (GbETH) for Ethernet, the current technology used for most LANs. ATM over IP pseudo OSI layers comprise an upper protocol, ATM Service Specific Convergence Sublayer (ATM-SSCS), necessary to translate between the ATM layer and RTD/UDP/IP sublayers. The User Datagram Protocol (UDP) is a connectionless protocol that, like TCP, runs on top of IP networks. Unlike TCP/IP, UDP/IP provides very few error recovery services, offering instead a direct way to send and receive data grams over an IP network. Sub-layers provide the Ethernet type, MAC header and PHY Ethernet respectively.
High-speed IP-based networks are the latest innovation in the world of communications. The capacity of these networks is increasing at a prodigious rate, fueled by the popularity of the Internet and decreasing costs associated with the technology. Worldwide data traffic volume has already surpassed that of the telephone network, and for many applications, the pricing of IP traffic has dropped below the tariffs associated with traditional TDM service. For this reason, significant effort is being expended on VoIP technologies. For users who have free, or fixed-price Internet access, Internet telephony software essentially provides free telephone calls anywhere in the world. To date, however, Internet telephony does not offer the same quality of telephone service as direct telephone connections. There are many Internet telephony applications available. Some come bundled with popular Web browsers; others are stand-alone products. Internet telephony products are sometimes called IP telephony, Voice over the Internet (VOI) or Voice over IP (VOIP) products.
Inherent in all forms of VoIP is revolutionary change, whereby much of the existing telephony infrastructure will be replaced by novel IP-based mechanisms. Despite the hype, this effort has been more protracted and less successful than initially expected. Today's telephony technology, both those portions that VoIP aims to replace and those to which VoIP must interface, is extremely complex. Revolutionary implementations of its hundreds of features and thousands of variations cannot be expected to be developed in a short time frame.
The present communications revolution has been focused on the Internet and the Internet protocol (IP). The prior art, however, has failed to teach a viable solution to handling TDM over Internet Protocol (IP). In addition, the use of TDM over IP for Voice over IP has not been heretofore possible.
Why Use IP Networks? The existing telephony infrastructure has an extremely high reliability (99.999%), supports reasonable audio quality (Mean Opinion Score, or MOS, 4.0 on a scale of 1 to 5), has almost universal market penetration, and offers a rich feature set. Accordingly, extremely potent incentives are required before one should consider supplanting it. There are two such incentives, one economic and one technological.
The part of the economic advantage of IP networks is shared by all packet networks; namely, that multiple packetized data streams can share a circuit, while a TDM timeslot occupies a dedicated circuit for the call's duration. Under “polite conversation” assumption of each party speaking only half of the time, and the “optimal engineering” assumption of minimal overhead, packet networks will, on average, double the bandwidth efficiency, thus halving operational costs. Taking overhead and peak statistics into account, the savings will be somewhat less, but a 30% reduction is attainable. However, it is doubtful whether this savings alone would be a strong enough encouragement to make the switch from TDM to IP.
The added catalyst has to do with the raw rates for data traffic as compared to voice traffic. At present, data communications are metered separately from traditional voice communications and are offered at substantial savings. These savings are partly due to tariffs and access charges that increase the cost of traditional voice services, and partly due to the attractive pricing of IP traffic. Put another way, voice service pricing is still mostly determined by incumbent carriers with high overhead costs, while IP traffic costs are much more competitive, as the provider incurs lower costs and is more focused on increasing market share.
The technological incentive has come to be called convergence. The reasoning is that technological simplification and synergy will result from consolidation of the various sources into an integrated environment. For example, a single residential information source provisioned for telephony, IP data and entertainment programming would in principle decrease end user prices, result in a single unified billing package, and eventually enable advanced services, such as video-on-demand.
The Limitations of VoIP
In principle, it would not seem difficult to carry voice over IP networks; a digitized voice signal is simply data and can be carried by a packet network just like any other data. The major technological achievement of the telephone network, that of least cost routing, has its counterpart in IP networks as well. There are, however, two fundamental problems that have to be solved before VoIP can be realistically considered to compete with TDM networks; namely, QoS and signaling.
Quality of Service
The meaning of Quality of Service is completely different for data and voice. Although most data can withstand relatively significant delay, low delay and proper time ordering of the signal are critical for voice applications, even though loss of a few milliseconds of signal is usually not noticeable. These requirements are completely at odds with the basic principles of IP networks (although not necessarily with those of other packet networks). To overcome these constraints, mechanisms such as tunneling and jitter buffers need to be employed. Additional components of voice quality such as echo cancellation and voice compression are not inherent in data-based networks at all, and need to be added ad hoc for VoIP.
Almost all of the massive R&D effort in the field of VoIP is directed towards solving these QoS problems, leaving the signaling problem largely unsolved. By signaling, we mean the exchange of information needed for a telephone call other than the speech itself Signaling consists of basic features such as the fact that the phone is off-h
Katten Muchin Zavis & Rosenman
RAD Data Communication Ltd.
Vanderpuye Kenneth
LandOfFree
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