Multiplex communications – Communication techniques for information carried in plural... – Combining or distributing information via frequency channels
Reexamination Certificate
1999-09-20
2004-06-08
Nguyen, Hanh (Department: 2662)
Multiplex communications
Communication techniques for information carried in plural...
Combining or distributing information via frequency channels
C370S350000, C370S494000, C370S352000, C370S419000, C379S093010, C379S093070
Reexamination Certificate
active
06747995
ABSTRACT:
FIELD OF THE INVENTION
The present invention relates to digital subscriber loop applications, and more particularly to multiple voice lines with data over a single shared subscriber loop.
BACKGROUND OF THE INVENTION
With the popularity of the Internet and the increasing trend of small businesses locating to the home, telephone service providers are experiencing a large and increasing demand for additional voice line service to businesses and homes.
Most central offices (COs) have excess switching capacity for providing additional voice lines to subscribers. Once an additional access line is extended to a subscriber, there is little expense involved in providing voice services and the added line can provide the telephone service provider with incremental revenue generating services.
The conventional approach for providing additional access voice lines to the subscriber is to add analog subscriber loops by laying additional copper lines to, and changing or adding lightning protection devices at, the subscriber premise. The subscriber loop is the two-wire copper transmission and signaling path between a telephone subscriber's terminal equipment and the serving central office or another piece of terminal equipment. However, the time and expense involved in this approach can be considerable, greatly increasing the time to recoup a return on investment.
A problem with any analog subscriber loop based signaling system, from a transmission perspective, is loss and impairment of the signal. This can be caused by physical conditions, such as bridge taps, gauge changes, line length, insulation, age, and environmental cable damage, or due to interference from external sources such as impulse noise and cross talk. Signal degradation typically manifests as noise, loss, distortion, and interference.
Another problem with the conventional approach is that analog loops are typically used with standard modems which use baseband POTS (Plain Old Telephone Service) voice frequency spectrum (0-4 kHz) to transmit information, and cannot exceed transmission power levels as dictated by the FCC due to cable pair crosstalk effects. The effect of the current FCC rules is to restrict the output of service providers' modems to download speeds of 53 kbps and upload speeds of 31.2 kbps. Actual speeds may vary depending on line conditions, but cannot exceed these maximums.
Frequency Division Multiplexing (FDM) is one technique for providing additional voice lines over a subscriber loop that does not require laying additional copper lines. This approach uses a frequency spectrum that is spectrally isolated from that used by baseband POTS, thus allowing additional 4 kHz analog POTS channels on higher frequency carrier signals to use the same two-wire subscriber loop. Such passband analog carrier techniques tend to amplify the loss and impairments analog loops typically suffer.
A technique that uses FDM is Digital Added Main Line (DAML). At the CO, a DAML modem is presented with two or more subscriber loop analog voice signals. These analog voice signals are converted by the modem to a digital line code format and transmitted over a single subscriber loop to another DAML modem located at or near the customer premise. The customer premise DAML modem decodes the line and presents the subscriber with two or more two-wire connections corresponding to the subscriber loop connections to the DAML modem at the CO. The digital line codes can take a number of forms, the most common of which are Amplitude, Phase and Frequency Shift Keying, 2-Binary-1-Quaternary, Carrierless Amplitude Phase Modulation, and Quadrature Amplitude Phase Modulation. A problem with this approach is that the D/A/D conversion at the CO of the pulse code modulation (PCM) digital signal to an analog loop signal back to the DAML digital signal can cause degradation of the signal through such effects as quantization errors and phase distortion.
Another technique used to transport multiple voice lines in a digital fashion over the subscriber loop is Integrated Services Digital Network (ISDN). This is a direct digital, multiple voice/data channel system that also includes a signaling channel. However, ISDN requires changes in equipment, administration and maintenance at the switching system.
Another approach involves transmitting voice packets over a data network which can include subscriber loops. The better known implementations of this approach are Voice Over IP (VOIP), Voice Over ATM (VOATM), and Voice Over Frame Relay (VOFR).
VOIP applications are typically deployed throughout a campus environment, using CAT 5 wiring or fiber as described in standards publication EIA/TIA-570-91, “Residential and Light Commercial Telecommunications Wiring,” Electronic Industries Alliance/Telecommunications Industry Association, June 1991, to each terminal and connected through a common switching fabric such as Ethernet, ATM or a hybrid system. In addition, calls can bridge to the Internet from the campus environment, or Intranet, via gateways such as routers or Layer 3 switching systems.
In some applications, a desktop computer or other device acts as the VOIP enabled terminal used to support remote communications consistent with ITU-T standards publication H.323, “Packet Based Multimedia Communications Systems,” International Telecommunications Union (ITU), Feb 1998. Such systems, typically employ Digital Signal Processors (DSPs) to provide compression of voice IP packets at the desktop which are then forwarded to other stations on the local Intranet or on through the Internet to remote stations. VOATM and VOFR are other packet techniques used to transport voice and interwork3 with the Public Switched Telephone Network (PSTN).
Subscriber loops can extend the reach of a WAN network for VOIP applications using xDSL signaling and transmission techniques. xDSL technologies enable bandwidth to the premise that may co-exist with baseband POTS service. ISDN can also provide bandwidth to the home that connects to a packet network through which it provides voice services. IP packets, ATM Cells, or other frame formats can be transported over subscriber loops using ISDN or xDSL technologies such as ADSL and HDSL.
However, voice and data have different requirements for network services. Voice transmission requires only a small amount of bandwidth, but that bandwidth must be available on a dedicated or continuous basis with very little delay, delay variation, or loss. Even delays in the millisecond range can give rise to noticeable echoes or gaps in the conversation. For example, delays introduced by routers and gateways can have adverse affects on voice.
Packetized speech belongs to the category of realtime data traffic, and as such has stringent delivery requirements with respect to loss and error. In packetized speech, the end-to-end average network delivery time must be small, and the end-to-end variation of the delivery time, including losses, must be small.
In voice transmission, the overall delay should not exceed 200 ms, which is the delay that has been accepted as commercially acceptable. 100-200 ms is the typical goal. At around 800 ms, the delay impedes normal telephonic conversation. Normally, a delay of 200-800 ms is conditionally acceptable for a short portion of the conversation when such occurrences are rare and far apart.
In traditional voice networks, the round trip delay is about 20-30 ms. Voice delays in frame relay networks, can be around 125-200 ms. In Ethernet networks carrying TCP/IP packets, the delay can vary widely depending on traffic loads. Due to the inherent realtime deficiencies of shared data networking technologies, the above issues represent serious challenges for the transmission of voice over typical campus networking environments extended to the premise.
In addition, ATM as a standard still lacks support for voice compression, silence suppression, idle channel cell suppression and signaling support including translation of voice signaling to switched virtual connection ATM signaling.
Further, ATM trunking for narrowband services, s
Brown Randall L.
Buege Thomas J.
Hoff James
McDonald Keith M.
Pawlowski Christopher A.
Lucent Technologies - Inc.
Nguyen Hanh
Santema Steven R.
Zwick David J.
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