Method for initializing and allocating bandwidth in a...

Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching

Reexamination Certificate

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C370S353000

Reexamination Certificate

active

06539011

ABSTRACT:

FIELD OF THE INVENTION
This invention relates to the field of telecommunications and data communications over a network and, more particularly, to initializing and allocating bandwidth in a permanent virtual connection suitable for integrated voice and data communications over a local area network.
Today, the typical office communications infrastructure consists of two independent networks: the telecommunications network and data communications network. The telecommunications network provides circuit switched channels with limited bandwidth (typically 64 Kbps to 128 Kbps). The circuit switched nature and limited bandwidth of this network cannot support today's high-speed data transport requirements. The office data communications network provides packet transport (Ethernet or Token Ring) via hubs and/or switches and, to a much lesser degree, cells in Asynchronous Transfer Mode (ATM). These data communications networks provide bandwidths to the desktop of 10 Mbps to 100 Mbps. However, the packet nature of these networks presents an impediment to the transport of delay sensitive data such as real-time audio or video, with the exception of ATM, which is not economically feasible to deploy to the desktop today.
Data transmission, voice and videoconferencing are converging and all will be provided over a single network fabric. The miracle that is really driving this convergence is the exponential improvement in chip technology. Products based on innovative new chip designs will soon provide for all the data, audio and video communications needs of the office using a single connection to each desktop. Office systems which unify voice, video and data communications, reducing the cost of ownership and allowing shared high speed Internet/Web access directly to the desktop are on the horizon. These products will offer a high quality alternative to existing stand-alone voice, videoconferencing and data networking equipment. These products will benefit users who want the convenience and utility of a Digital Key Telephone system or PBX along with the added advantage of a fully-integrated Local Area Network (LAN), and high-speed Wide Area Network (WAN) access, all housed within one system.
BACKGROUND OF THE INVENTION
Local Network Link Operation for a Traditional Digital Key/Hybrid Telephone System
A traditional Digital Key/Hybrid Office Telecommunications System consists of two (2) major components: 1) the Digital Key Telephone instrument; and 2) the Common Equipment Unit (i.e., the back room or wiring closet equipment) which interconnects the Digital Key Telephones and the external Central Office (C.O.) lines.
The typical office internal telecommunications network uses a “Star Wiring Topology”, consisting of “home run wiring”, where each individual telephone is connected back to the Common Equipment Unit (CEU) on a dedicated Unshielded Twisted Pair (UTP) cable.
There is an important distinction to be made here between an industry standard analog 2500 type telephone (i.e., Touch Tone® Telephone) connected to a PBX (the type of CEU) and an electronic Digital Key Telephone connected to a PBX. Like the electronic Digital Key telephone, the analog 2500 type telephone is connected to the PBX by “home run wiring”, forming a “Star Wiring Topology”, where each individual telephone is connected back to the PBX on a dedicated UTP cable. However, the analog 2500 type telephone uses “in-band” audio channel signaling to communicate to the PBX.
Analog PBX Signaling Methods
The analog 2500 type telephone is connected to the PBX over the Unshielded Twisted Pair (UTP) cable using an industry standard “Loop Interface”. The telephone loop interface port (station port) on the PBX provides a source for “DC Loop Current” and an analog signal channel bandwidth from 300 Hz to 3,400 Hz for audio signal transmission. The standard loop interface provides for two types of signaling to the Common Equipment Unit (CEU) over the UTP cable: 1) Hook Switch State and 2) In-Band DTMF (Dual Tone Multi Frequency) Signals.
When the analog 2500 type telephone is “On Hook”, it is in the idle state and no DC loop current is flowing between the associated PBX Station Port and the telephone. When the handset of the analog 2500 type telephone is lifted from its cradle (i.e., goes “Off Hook”), the “Hook Switch” contact is closed and DC loop current flows between the PBX Station Port and the telephone. The loop interface circuitry at the PBX station port monitors the status of the DC loop current (i.e., no loop current flowing; or loop current flowing within an acceptable range) to determine the state of the analog 2500 type telephone connected to the PBX station port by the UTP cable. No loop current flowing indicates that the telephone is in the “Idle On Hook State” and requires no servicing. The detection of DC loop current flowing, within an acceptable range, indicates that the telephone has gone “Off Hook” and requires servicing.
Through the “On Hook” and “Off Hook” states produced by the analog 2500 type telephone, and the detection thereof by the associated PBX station port, the telephone can communicate (i.e., signal) to the PBX that it requires service. Now that the telephone has signaled to the PBX that it needs to be serviced, it needs a means to communicate to the PBX what type of service it requires. The type of service request is communicated using “in-band” DTMF Signaling. As previously described, the loop interface provides a 300 Hz to 3,400 Hz bandwidth audio channel between the analog 2500 type telephone and the associated PBX station port. The telephone contains a DTMF signal generator and the PBX station port has access to a DTMF signal detector. The DTMF signaling scheme comprises a base of sixteen (16) unique digits, or characters. The composite spectrum of the DTMF signals fall within the 300 Hz to 3,400 Hz bandwidth audio channel allowing the DTMF digits to be transmitted over the loop interface for communicating service requests and address signaling to the PBX. Once the DTMF signal transmissions have subsided, the audio channel bandwidth is available for the transmission of voice signals. Hence the term “in-band” signaling, where the same channel bandwidth is used to transport both the DTMF signaling information and the voice signal information.
Digital Key Telephone PBX Signaling and Switching Methods
The commercially available systems today use vendor proprietary communications links to transport the digitized voice and telephone control signaling between the proprietary Digital Key Telephone and Common Equipment Unit (CEU) over the Unshielded Twisted Pair (UTP) cable. Typically, equipment vendors transport two (2) full-duplex 64 Kbps Bearer Channels and one (1) full-duplex 16Kbps Signaling D Channel (2B+D) over the communications link between the telephone and the CEU. The two 64 Kbps B Channels are used to support circuit switched digitized voice, or circuit switched data transport, channels. The 16Kbps D Channel is used to transport telephone control signaling packets and low speed data (e.g., ASCII character transmission from the CEU to the telephone LCD display).
The two (2) 64 Kbps B Channels are capable of transporting digitized voice in the form of 8 Bit PCM (Pulse Code Modulation) words, or other 8 bit digital data synchronously formatted to these Time Domain Multiplexed (TDM) channels. In both cases, the transport of information in a B Channel is on circuit switched bases. The nature of the circuit switched connection is that it is set up when there is information to transport. It provides a constant bandwidth (in this case 64Kbps per B Channel) and this constant bandwidth is available for the duration of the connection. Finally, the connection is torn down when it is no longer required. This actually describes the typical telephone call. A telephone number is dialed, the connection is made, and a conversation is held for some period of time. The connection is torn down when the conversation has been completed by the user going on hook. Therefore, the B Channels of the Digital Key Teleph

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